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Cisco ATA 186 and Cisco ATA 188
Analog Telephone Adaptor
Administrator’s Guide (SIP)
Corporate Headquarters
Cisco Systems, Inc.
170 West Tasman Drive
San Jose, CA 95134-1706
USA
http://www.cisco.com
Tel: 408 526-4000
800 553-NETS (6387)
Fax: 408 526-4100
Text Part Number: OL-3410-01
THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL
STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT
WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.
THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THAT
SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE
OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY.
The following information is for FCC compliance of Class A devices: This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant
to part 15 of the FCC rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial
environment. This equipment generates, uses, and can radiate radio-frequency energy and, if not installed and used in accordance with the instruction manual, may cause
harmful interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful interference, in which case users will be required
to correct the interference at their own expense.
The following information is for FCC compliance of Class B devices: The equipment described in this manual generates and may radiate radio-frequency energy. If it is not
installed in accordance with Cisco’s installation instructions, it may cause interference with radio and television reception. This equipment has been tested and found to
comply with the limits for a Class B digital device in accordance with the specifications in part 15 of the FCC rules. These specifications are designed to provide reasonable
protection against such interference in a residential installation. However, there is no guarantee that interference will not occur in a particular installation.
Modifying the equipment without Cisco’s written authorization may result in the equipment no longer complying with FCC requirements for Class A or Class B digital
devices. In that event, your right to use the equipment may be limited by FCC regulations, and you may be required to correct any interference to radio or television
communications at your own expense.
You can determine whether your equipment is causing interference by turning it off. If the interference stops, it was probably caused by the Cisco equipment or one of its
peripheral devices. If the equipment causes interference to radio or television reception, try to correct the interference by using one or more of the following measures:
• Turn the television or radio antenna until the interference stops.
• Move the equipment to one side or the other of the television or radio.
• Move the equipment farther away from the television or radio.
• Plug the equipment into an outlet that is on a different circuit from the television or radio. (That is, make certain the equipment and the television or radio are on circuits
controlled by different circuit breakers or fuses.)
Modifications to this product not authorized by Cisco Systems, Inc. could void the FCC approval and negate your authority to operate the product.
NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED “AS IS” WITH
ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT
LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF
DEALING, USAGE, OR TRADE PRACTICE.
IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING,
WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO
OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.
CCIP, CCSP, the Cisco Arrow logo, the Cisco Powered Network mark, the Cisco Systems Verified logo, Cisco Unity, Follow Me Browsing, FormShare, iQ Breakthrough,
iQ FastTrack, the iQ Logo, iQ Net Readiness Scorecard, Networking Academy, ScriptShare, SMARTnet, TransPath, and Voice LAN are trademarks of Cisco Systems, Inc.;
Changing the Way We Work, Live, Play, and Learn, The Fastest Way to Increase Your Internet Quotient, and iQuick Study are service marks of Cisco Systems, Inc.; and
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Systems, Cisco Systems Capital, the Cisco Systems logo, Empowering the Internet Generation, Enterprise/Solver, EtherChannel, EtherSwitch, Fast Step, GigaStack, Internet
Quotient, IOS, IP/TV, iQ Expertise, LightStream, MGX, MICA, the Networkers logo, Network Registrar, Packet, PIX, Post-Routing, Pre-Routing, RateMUX, Registrar,
SlideCast, StrataView Plus, Stratm, SwitchProbe, TeleRouter, and VCO are registered trademarks of Cisco Systems, Inc. and/or its affiliates in the U.S. and certain other
countries.
All other trademarks mentioned in this document or Web site are the property of their respective owners. The use of the word partner does not imply a partnership relationship
between Cisco and any other company. (0301R)
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
Copyright © 2003, Cisco Systems, Inc.
All rights reserved.
C ON T E N T S
Preface
xi
Overview
xi
Audience
xi
Organization
xii
Conventions
xii
Related Documentation
xiv
Obtaining Documentation xv
World Wide Web xv
Documentation CD-ROM xv
Ordering Documentation xv
Documentation Feedback xv
Obtaining Technical Assistance xvi
Cisco.com xvi
Technical Assistance Center xvi
Cisco TAC Web Site xvii
Cisco TAC Escalation Center xvii
Cisco Analog Telephone Adaptor Overview
Session Initiation Protocol (SIP) Overview
SIP Capabilities 1-3
Components of SIP 1-3
SIP Clients 1-4
SIP Servers 1-4
Hardware Overview
1-1
1-2
1-5
Software Features 1-7
Voice Codecs Supported 1-7
Additional Supported Signaling Protocols
Other Supported Protocols 1-8
Cisco ATA SIP Services 1-8
Fax Services 1-9
Methods Supported 1-9
Supplementary Services 1-10
Installation and Configuration Overview
1-8
1-10
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Contents
Installing the Cisco ATA
2-1
Network Requirements
2-2
Safety Recommendations
2-2
What the Cisco ATA Package Includes
What You Need
2-2
2-3
Installation Procedure
Power-Down Procedure
2-3
2-6
Configuring the Cisco ATA for SIP
Default Boot Load Behavior
3-1
3-2
Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation
3-3
Steps Needed to Configure the Cisco ATA 3-5
Basic Configuration Steps in a TFTP Server Environment 3-5
Basic Configuration Steps in a Non-TFTP Server Environment 3-6
Configuring the Cisco ATA Using a TFTP Server 3-7
Setting Up the TFTP Server with Cisco ATA Software 3-7
Configurable Features and Related Parameters 3-7
Creating Unique and Common Cisco ATA Configuration Files 3-8
Using atapname.exe Tool to Obtain MAC Address 3-10
Using the EncryptKey Parameter and cfgfmt Tool 3-11
atadefault.cfg Configuration File 3-12
Configuring the Cisco ATA to Obtain its Configuration File from the TFTP Server
Using a DHCP Server 3-13
Without Using a DHCP Server 3-15
Voice Configuration Menu 3-15
Using the Voice Configuration Menu 3-16
Entering Alphanumeric Values 3-17
Resetting the Cisco ATA to Factory Default Values
Cisco ATA Web Configuration Page
3-13
3-18
3-18
Refreshing or Resetting the Cisco ATA 3-21
Procedure to Refresh the Cisco ATA 3-22
Procedure to Reset the Cisco ATA 3-22
Upgrading the SIP Signaling Image
Basic and Additional SIP Services
3-22
4-1
Important Basic SIP Services 4-1
Required Parameters 4-1
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Establishing Authentication 4-2
Setting the Codec 4-3
Configuring Refresh Interval 4-3
Additional SIP Services 4-3
Advanced Audio Configuration 4-4
Billable Features 4-4
Comfort Noise During Silence Period When Using G.711 4-5
Configurable Hook Flash Timing 4-5
Configurable Mixing of Call Waiting Tone and Audio 4-5
Configurable On-hook delay 4-5
Debugging Diagnostics 4-5
Dial Plan 4-6
Disabling Access To The Web Interface 4-6
Distinctive Ringing 4-6
DNS SRV Support 4-6
Hardware Information Display 4-7
NAT Gateway 4-7
NAT/PAT Translation 4-7
Network Timing 4-8
OutBoundProxy Support 4-8
Progress Tones 4-8
Receiver-tagged VIA header 4-9
Repeat Dialing on Busy Signal 4-9
SIP Proxy Server Redundancy 4-10
Stuttering Dial Tone on Unconditional Call Forward 4-10
User Configurable Call Waiting Permanent Default Setting 4-10
User Configurable Timeout On No Answer for Call Forwarding 4-10
Setting Up and Placing a Call Without Using a SIP Proxy 4-11
Configuration 4-11
Placing an IP Call 4-12
Complete Reference Table of all Cisco ATA SIP Services
Parameters and Defaults
5-1
Configuration Text File Template
User Interface (UI) Parameter
UIPassword 5-3
Configuration Parameter
ToConfig 5-4
4-12
5-2
5-3
5-4
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Parameters for Configuration Method
UseTFTP 5-4
TftpURL 5-5
CfgInterval 5-5
EncryptKey 5-6
5-4
Network Parameters 5-6
DHCP 5-7
StaticIp 5-7
StaticRoute 5-7
StaticNetMask 5-8
Account Information Parameters
UID0 5-9
PWD0 5-9
UID1 5-9
PWD1 5-10
GkOrProxy 5-10
Gateway 5-11
Gateway2 5-11
UseLoginID 5-11
LoginID0 5-12
LoginID1 5-12
Backup Server Parameters
AltGk 5-13
AltGkTimeOut 5-13
GkTimeToLive 5-14
GkId 5-14
5-8
5-13
SIP Parameters 5-14
UseSIP 5-14
SIPRegInterval 5-15
MAXRedirect 5-15
SIPRegOn 5-16
NATIP 5-16
SIPPort 5-17
MediaPort 5-17
OutBoundProxy 5-17
NatServer 5-18
NatTimer 5-19
Operating Parameters
5-19
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LBRCodec 5-20
AudioMode 5-20
RxCodec 5-21
TxCodec 5-22
NumTxFrames 5-22
CallFeatures 5-23
PaidFeatures 5-24
CallerIdMethod 5-25
FeatureTimer 5-26
Polarity 5-27
ConnectMode 5-28
AutMethod 5-30
TimeZone 5-30
NTPIP 5-30
AltNTPIP 5-31
DNS1IP 5-31
DNS2IP 5-31
UDPTOS 5-32
SigTimer 5-32
OpFlags 5-34
VLANSetting 5-35
Optional Feature Parameters 5-35
NPrintf 5-36
TraceFlags 5-36
RingOnOffTime 5-37
IPDialPlan 5-38
DialPlan 5-38
About Dial Plan Commands 5-39
Dial Plan Blocking (In Rule) 5-41
'H' Rule to Support Hot/Warm Line 5-41
'P' Rule to Support Dial Prefix 5-42
Call-Progress Tone Parameters 5-42
List of Call-Progress Tone Parameters 5-42
Tone Parameter Syntax 5-42
How to Calculate Scaling Factors 5-43
Recommended Values 5-44
Specific Call-Progress Tone Parameter Information
CallCmd 5-47
5-44
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Contents
Call Commands
6-1
Call Command Structure
6-1
Syntax 6-2
Context-Identifiers 6-3
Input Sequence Identifiers
Action Identifiers 6-4
Call Command Example
6-5
Call Command Behavior
6-7
6-4
Configuring and Debugging Fax Services
7-1
Using Fax Pass-through Mode 7-1
Configuring the Cisco ATA for Fax Pass-through mode 7-2
AudioMode 7-2
ConnectMode 7-3
Configuring Cisco IOS Gateways to Enable Fax Pass-through
Enable Fax Pass-through Mode 7-4
Disable Fax Relay Feature 7-5
Using FAX Mode 7-6
Configuring the Cisco ATA for Fax Mode 7-6
Configuring the Cisco ATA for Fax Mode on a Per-Call Basis
Configuring the Cisco IOS Gateway for Fax Mode 7-7
Debugging the Cisco ATA 186/188 Fax Services 7-7
Common Problems When Using IOS Gateways 7-7
Using prserv for Diagnosing Fax Problems 7-9
prserv Overview 7-9
Analyzing prserv Output for Fax Sessions 7-9
Using rtpcatch for Diagnosing Fax Problems 7-12
rtpcatch Overview 7-12
Example of rtpcatch 7-13
Analyzing rtpcatch Output for Fax Sessions 7-16
Using rtpcatch to Analyze Common Causes of Failure
rtpcatch Limitations 7-19
Upgrading the Cisco ATA Signaling Image
7-7
7-17
8-1
Upgrading the Signaling Image from a TFTP Server
Upgrading the Signaling Image Manually
Preliminary Steps 8-3
7-3
8-1
8-2
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Running the Executable File 8-3
Upgrade Requirements 8-3
Syntax 8-3
Upgrade Procedure 8-4
Confirming a Successful Signaling Image Upgrade
Using a Web Browser 8-5
Using the Voice Configuration Menu 8-5
Troubleshooting
9-1
General Troubleshooting Tips
Symptoms and Actions
9-1
9-2
Installation and Upgrade Issues
Debugging
8-5
9-3
9-4
Frequently Asked Questions
Contacting TAC
9-5
9-6
Using SIP Supplementary Services
Changing Call Commands
A-1
A-1
Cancelling a Supplementary Service
A-1
Common Supplementary Services A-1
Caller ID A-2
Call-Waiting Caller ID A-2
Voice Mail Indication A-2
Unattended Transfer A-3
Semi-unattended Transfer A-3
Fully Unattended Transfer A-3
Attended Transfer A-4
Making a Conference Call in the United States
Making a Conference Call in Sweden A-4
Call Waiting in the United States A-5
Call Waiting in Sweden A-5
About Call Forwarding A-5
Call Forwarding in the United States A-5
Call Forwarding in Sweden A-6
Call Return in the United States A-6
Call Return in Sweden A-6
A-4
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Calling Line Identification Presentation A-6
About Calling Line Identification Restriction A-6
Calling Line Identification Restriction in the United States
Calling Line Identification Restriction in Sweden A-7
Voice Menu Codes
B-1
Cisco ATA Specifications
Physical Specifications
Electrical Specifications
C-1
C-1
C-2
Environmental Specifications
Immunity Specifications
Physical Interfaces
C-2
C-2
C-3
Ringing Characteristics
Software Specifications
C-3
C-3
SIP Compliance Reference Information
SIP Call Flows
A-7
C-5
D-1
Supported SIP Request Methods
D-1
Call Flow Scenarios for Successful Calls D-2
Cisco ATA-to-SIP Server—Registration without Authentication D-2
Cisco ATA-to-SIP Server—Registration with Authentication D-3
Cisco ATA-to-Cisco ATA—Basic SIP to SIP Call without Authentication D-6
Cisco ATA-to-Cisco ATA—Basic SIP to SIP Call with Authentication D-12
GLOSSARY
INDEX
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Preface
This preface includes the following sections:
•
Overview, page xi
•
Audience, page xi
•
Organization, page xii
•
Conventions, page xii
•
Related Documentation, page xiv
•
Obtaining Documentation, page xv
•
Obtaining Technical Assistance, page xvi
Overview
The Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP) provides
the information you need to install, configure and manage the Cisco ATA 186 and Cisco ATA 188 on a
Session Initiation Protocol (SIP) network.
Note
The term Cisco ATA is used throughout this manual to refer to both the Cisco ATA 186 and the
Cisco ATA 188, unless differences between the Cisco ATA 186 and Cisco ATA 188 are explicitly stated.
Audience
This guide is intended for service providers and network administrators who administer Voice over IP
(VoIP) services using the Cisco ATA. Most of the tasks described in this guide are not intended for end
users of the Cisco ATA. Many of these tasks impact the ability of the Cisco ATA to function on the
network, and require an understanding of IP networking and telephony concepts.
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
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Organization
Organization
Table 1 provides an overview of the organization of this guide.
Table 1
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP) Organization
Chapter
Description
Chapter 1, “Cisco Analog Telephone Adaptor Overview”
Provides descriptions of hardware and software features of
the Cisco ATA Analog Telephone Adaptor along with a brief
overview of the Session Initiation Protocol (SIP).
Chapter 2, “Installing the Cisco ATA”
Provides information about installing the Cisco ATA.
Chapter 3, “Configuring the Cisco ATA for SIP”
Provides information about configuring the Cisco ATA and
the various methods for configuration.
Chapter 4, “Basic and Additional SIP Services”
Provides information about SIP services that the Cisco ATA
supports.
Chapter 5, “Parameters and Defaults,”
Provides information on all parameters and defaults that you
can use to configure the Cisco ATA.
Chapter 6, “Call Commands”
Provides the Cisco ATA call commands for SIP.
Chapter 7, “Configuring and Debugging Fax Services”
Provides instructions for configuring both ports of the
Cisco ATA to support fax transmission.
Chapter 8, “Upgrading the Cisco ATA Signaling Image”
Provides instructions for remotely upgrading Cisco ATA
software.
Chapter 9, “Troubleshooting”
Provides basic testing and troubleshooting procedures for the
Cisco ATA.
Appendix A, “Using SIP Supplementary Services”
Provides end-user information about pre-call and mid-call
services.
Appendix B, “Voice Menu Codes”
Provides a quick-reference list of the voice configuration
menu options for the Cisco ATA.
Appendix C, “Cisco ATA Specifications”
Provides physical specifications for the Cisco ATA.
Appendix D, “SIP Call Flows”
Provides Cisco ATA call flows for SIP scenarios.
Glossary
Provides definitions of commonly used terms.
Index
Provides reference information.
Conventions
This document uses the following conventions:
•
Alternative keywords are grouped in braces and separated by vertical bars (for example, {x | y | z}).
•
Arguments for which you supply values are in italic font.
•
Commands and keywords are in boldface font.
•
Elements in square brackets ([ ]) are optional.
•
Information you must enter is in boldface
•
Optional alternative keywords are grouped in brackets and separated by vertical bars (for example,
[x | y | z]).
screen
font.
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
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Conventions
•
Note
Timesaver
Tip
Caution
Terminal sessions and information the system displays are in screen font.
Means reader take note. Notes contain helpful suggestions or references to material not covered in the
publication.
Means the described action saves time. You can save time by performing the action described in the
paragraph.
Means the following information will help you solve a problem. The tips information might not be
troubleshooting or even an action, but could be useful information, similar to a Timesaver.
Means reader be careful. In this situation, you might do something that could result in equipment
damage or loss of data.
Warning
This warning symbol means danger. You are in a situation that could cause bodily injury. Before you
work on any equipment, be aware of the hazards involved with electrical circuitry and be familiar
with standard practices for preventing accidents. (To see translations of the warnings that appear
in this publication, refer to the appendix “Translated Safety Warnings.”)
Waarschuwing
Dit waarschuwingssymbool betekent gevaar. U verkeert in een situatie die lichamelijk letsel kan
veroorzaken. Voordat u aan enige apparatuur gaat werken, dient u zich bewust te zijn van de bij
elektrische schakelingen betrokken risico’s en dient u op de hoogte te zijn van standaard
maatregelen om ongelukken te voorkomen. (Voor vertalingen van de waarschuwingen die in deze
publicatie verschijnen, kunt u het aanhangsel “Translated Safety Warnings” (Vertalingen van
veiligheidsvoorschriften) raadplegen.)
Varoitus
Tämä varoitusmerkki merkitsee vaaraa. Olet tilanteessa, joka voi johtaa ruumiinvammaan. Ennen
kuin työskentelet minkään laitteiston parissa, ota selvää sähkökytkentöihin liittyvistä vaaroista ja
tavanomaisista onnettomuuksien ehkäisykeinoista. (Tässä julkaisussa esiintyvien varoitusten
käännökset löydät liitteestä "Translated Safety Warnings" (käännetyt turvallisuutta koskevat
varoitukset).)
Attention
Ce symbole d’avertissement indique un danger. Vous vous trouvez dans une situation pouvant
entraîner des blessures. Avant d’accéder à cet équipement, soyez conscient des dangers posés par
les circuits électriques et familiarisez-vous avec les procédures courantes de prévention des
accidents. Pour obtenir les traductions des mises en garde figurant dans cette publication, veuillez
consulter l’annexe intitulée « Translated Safety Warnings » (Traduction des avis de sécurité).
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Related Documentation
Warnung
Dieses Warnsymbol bedeutet Gefahr. Sie befinden sich in einer Situation, die zu einer
Körperverletzung führen könnte. Bevor Sie mit der Arbeit an irgendeinem Gerät beginnen, seien Sie
sich der mit elektrischen Stromkreisen verbundenen Gefahren und der Standardpraktiken zur
Vermeidung von Unfällen bewußt. (Übersetzungen der in dieser Veröffentlichung enthaltenen
Warnhinweise finden Sie im Anhang mit dem Titel “Translated Safety Warnings” (Übersetzung der
Warnhinweise).)
Avvertenza
Questo simbolo di avvertenza indica un pericolo. Si è in una situazione che può causare infortuni.
Prima di lavorare su qualsiasi apparecchiatura, occorre conoscere i pericoli relativi ai circuiti
elettrici ed essere al corrente delle pratiche standard per la prevenzione di incidenti. La traduzione
delle avvertenze riportate in questa pubblicazione si trova nell’appendice, “Translated Safety
Warnings” (Traduzione delle avvertenze di sicurezza).
Advarsel
Dette varselsymbolet betyr fare. Du befinner deg i en situasjon som kan føre til personskade. Før du
utfører arbeid på utstyr, må du være oppmerksom på de faremomentene som elektriske kretser
innebærer, samt gjøre deg kjent med vanlig praksis når det gjelder å unngå ulykker. (Hvis du vil se
oversettelser av de advarslene som finnes i denne publikasjonen, kan du se i vedlegget "Translated
Safety Warnings" [Oversatte sikkerhetsadvarsler].)
Aviso
Este símbolo de aviso indica perigo. Encontra-se numa situação que lhe poderá causar danos
fisicos. Antes de começar a trabalhar com qualquer equipamento, familiarize-se com os perigos
relacionados com circuitos eléctricos, e com quaisquer práticas comuns que possam prevenir
possíveis acidentes. (Para ver as traduções dos avisos que constam desta publicação, consulte o
apêndice “Translated Safety Warnings” - “Traduções dos Avisos de Segurança”).
¡Advertencia!
Este símbolo de aviso significa peligro. Existe riesgo para su integridad física. Antes de manipular
cualquier equipo, considerar los riesgos que entraña la corriente eléctrica y familiarizarse con los
procedimientos estándar de prevención de accidentes. (Para ver traducciones de las advertencias
que aparecen en esta publicación, consultar el apéndice titulado “Translated Safety Warnings.”)
Varning!
Denna varningssymbol signalerar fara. Du befinner dig i en situation som kan leda till personskada.
Innan du utför arbete på någon utrustning måste du vara medveten om farorna med elkretsar och
känna till vanligt förfarande för att förebygga skador. (Se förklaringar av de varningar som
förekommer i denna publikation i appendix "Translated Safety Warnings" [Översatta
säkerhetsvarningar].)
Related Documentation
•
RFC3261 (SIP: Session Initiation Protocol)
•
RFC2543 (SIP: Session Initiation Protocol)
•
Cisco ATA SIP Compliance Reference Information
http://www-vnt.cisco.com/SPUniv/SIP/documents/CiscoATASIPComplianceRef.pdf
•
RFC768 (User Datagram Protocol)
•
RFC2198 (RTP Payload for Redundant Audio Data)
•
RFC2833 (RTP Payload for DTMF Digits, Telephony Phones and Telephony Signals)
•
RFC2327 (SDP: Session Description Protocol)
•
RFC3266 (Support for IPv6 in Session Description Protocol (SDP))
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
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Obtaining Documentation
•
Read Me First - ATA Boot Load Information
•
Cisco ATA 186 and Cisco 188 Analog Telephone Adaptor At a Glance
•
Regulatory Compliance and Safety Information for the Cisco ATA 186 and Cisco 188
•
Cisco ATA Release Notes
Obtaining Documentation
These sections explain how to obtain documentation from Cisco Systems.
World Wide Web
You can access the most current Cisco documentation on the World Wide Web at this URL:
http://www.cisco.com
Translated documentation is available at this URL:
http://www.cisco.com/public/countries_languages.shtml
Documentation CD-ROM
Cisco documentation and additional literature are available in a Cisco Documentation CD-ROM
package, which is shipped with your product. The Documentation CD-ROM is updated monthly and may
be more current than printed documentation. The CD-ROM package is available as a single unit or
through an annual subscription.
Ordering Documentation
You can order Cisco documentation in these ways:
•
Registered Cisco.com users (Cisco direct customers) can order Cisco product documentation from
the Networking Products MarketPlace:
http://www.cisco.com/cgi-bin/order/order_root.pl
•
Registered Cisco.com users can order the Documentation CD-ROM through the online Subscription
Store:
http://www.cisco.com/go/subscription
•
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Obtaining Technical Assistance
Cisco provides Cisco.com as a starting point for all technical assistance. Customers and partners can
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Cisco.com
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The Cisco Technical Assistance Center (TAC) is available to all customers who need technical assistance
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Cisco TAC inquiries are categorized according to the urgency of the issue:
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Priority level 4 (P4)—You need information or assistance concerning Cisco product capabilities,
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Priority level 1 (P1)—Your production network is down, and a critical impact to business operations
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The Cisco TAC resource that you choose is based on the priority of the problem and the conditions of
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Cisco TAC Escalation Center
The Cisco TAC Escalation Center addresses priority level 1 or priority level 2 issues. These
classifications are assigned when severe network degradation significantly impacts business operations.
When you contact the TAC Escalation Center with a P1 or P2 problem, a Cisco TAC engineer
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To obtain a directory of toll-free Cisco TAC telephone numbers for your country, go to this URL:
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Before calling, please check with your network operations center to determine the level of Cisco support
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Supported Accounts (NSA). When you call the center, please have available your service agreement
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1
Cisco Analog Telephone Adaptor Overview
This section describes the hardware and software features of the Cisco Analog Telephone Adaptor
(Cisco ATA) and includes a brief overview of the Session Initiation Protocol (SIP).
The Cisco ATA analog telephone adaptors are handset-to-Ethernet adaptors that allow regular analog
telephones to operate on IP-based telephony networks. Cisco ATAs support two voice ports, each with
an independent telephone number. The Cisco ATA 188 also has an RJ-45 10/100 BASE-T data port.
This section covers the following topics:
•
Session Initiation Protocol (SIP) Overview, page 1-2
•
Hardware Overview, page 1-5
•
Software Features, page 1-7
•
Installation and Configuration Overview, page 1-10
Figure 1-1
Cisco ATA Analog Telephone Adaptor
CISCO A
TA 186
TELEPH
ONE AD
APTOR
72209
ANALOG
The Cisco ATA, which operates with Cisco voice-packet gateways, makes use of broadband pipes that
are deployed through a digital subscriber line (DSL), fixed wireless-cable modem, and other Ethernet
connections.
Note
The term Cisco ATA refers to both the Cisco ATA 186 and the Cisco ATA 188, unless otherwise stated.
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Session Initiation Protocol (SIP) Overview
Figure 1-2
Cisco ATA 186 as Endpoint in SIP Network
Voice
gateway
Layer 3
Broadband
Ethernet
Cisco ATA 186
Broadband CPE
(DSL, cable,
fixed wireless)
V
IP infrastructure
V
PSTN
72088
Telephone/fax
SIP proxy
Cisco ATA 188 as Endpoint in SIP Network
Voice
gateway
Layer 3
Telephone/fax
Broadband
Ethernet
Cisco ATA 188
Broadband CPE
(DSL, cable,
fixed wireless)
V
IP infrastructure
V
PSTN
72444
Figure 1-3
SIP proxy
Session Initiation Protocol (SIP) Overview
Session Initiation Protocol (SIP) is the Internet Engineering Task Force (IETF) standard for real-time
calls and conferencing over Internet Protocol (IP). SIP is an ASCII-based, application-layer control
protocol (defined in RFC3261) that can be used to establish, maintain, and terminate multimedia
sessions or calls between two or more endpoints.
Like other Voice over IP (VoIP) protocols, SIP is designed to address the functions of signaling and
session management within a packet telephony network. Signaling allows call information to be carried
across network boundaries. Session management provides the ability to control the attributes of an
end-to-end call.
Note
SIP for the Cisco ATA is compliant with RFC2543.
This section contains the following topics:
•
SIP Capabilities, page 1-3
•
Components of SIP, page 1-3
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Session Initiation Protocol (SIP) Overview
SIP Capabilities
SIP provides the following capabilities:
•
Determines the availability of the target endpoint. If a call cannot be completed because the target
endpoint is unavailable, SIP determines whether the called party is already on the phone or did not
answer in the allotted number of rings. SIP then returns a message indicating why the target endpoint
was unavailable.
•
Determines the location of the target endpoint. SIP supports address resolution, name mapping, and
call redirection.
•
Determines the media capabilities of the target endpoint. Using the Session Description Protocol
(SDP), SIP determines the lowest level of common services between endpoints. Conferences are
established using only the media capabilities that are supported by all endpoints.
•
Establishes a session between the originating and target endpoint. If the call can be completed, SIP
establishes a session between the endpoints. SIP also supports mid-call changes, such as adding
another endpoint to the conference or changing the media characteristic or codec.
•
Handles the transfer and termination of calls. SIP supports the transfer of calls from one endpoint
to another. During a call transfer, SIP establishes a session between the transferee and a new
endpoint (specified by the transferring party) and terminates the session between the transferee and
the transferring party. At the end of a call, SIP terminates the sessions between all parties.
Conferences can consist of two or more users and can be established using multicast or multiple
unicast sessions.
Components of SIP
SIP is a peer-to-peer protocol. The peers in a session are called User Agents (UAs). A user agent can
function in one of the following roles:
•
User agent client (UAC)—A client application that initiates the SIP request.
•
User agent server (UAS)—A server application that contacts the user when a SIP request is received
and returns a response on behalf of the user.
Typically, a SIP endpoint is capable of functioning as both a UAC and a UAS, but functions only as one
or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UA that
initiated the request.
From an architectural standpoint, the physical components of a SIP network can also be grouped into
two categories—Clients and servers. Figure 1-4 illustrates the architecture of a SIP network.
Note
SIP servers can interact with other application services, such as Lightweight Directory Access Protocol
(LDAP) servers, a database application, or an extensible markup language (XML) application. These
application services provide back-end services such as directory, authentication, and billable services.
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Session Initiation Protocol (SIP) Overview
Figure 1-4
SIP Architecture
SIP proxy and
redirect servers
SIP
SIP
SIP
SIP user agents
SIP gateway
PSTN
Legacy PBX
72342
RTP
SIP Clients
SIP clients include:
•
Gateways—Provide call control. Gateways provide many services, the most common being a
translation function between SIP conferencing endpoints and other terminal types. This function
includes translation between transmission formats and between communications procedures. In
addition, the gateway also translates between audio and video codecs and performs call setup and
clearing on both the LAN side and the switched-circuit network side.
•
Telephones—Can act as either a UAS or UAC. The Cisco ATA can initiate SIP requests and respond
to requests.
SIP Servers
SIP servers include:
•
Proxy server—The proxy server is an intermediate device that receives SIP requests from a client
and then forwards the requests on the client’s behalf. Proxy servers receive SIP messages and
forward them to the next SIP server in the network. Proxy servers can provide functions such as
authentication, authorization, network access control, routing, reliable request retransmission, and
security.
•
Redirect server—Receives SIP requests, strips out the address in the request, checks its address
tables for any other addresses that may be mapped to the address in the request, and then returns the
results of the address mapping to the client. Redirect servers provide the client with information
about the next hop or hops that a message should take, then the client contacts the next hop server
or UAS directly.
•
Registrar server—Processes requests from UACs for registration of their current location. Registrar
servers are often co-located with a redirect or proxy server.
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Hardware Overview
Hardware Overview
The Cisco ATA 186 and Cisco ATA 188 are compact, easy to install devices. Figure 1-5 shows the rear
panel of the Cisco ATA 186. Figure 1-6 shows the rear panel of the Cisco ATA 188.
PHONE 2
10BaseT
ACT
PHONE 1
5V
Power
connector
RJ-11 FXS ports
RJ-45 10BaseT
Figure 1-6
72210
PHONE 1
Cisco ATA 186—Rear View
ACT LED
Cisco ATA 188—Rear View
PHONE 2
LINK
RJ-11 FXS ports
10/100 PC
10/100 UPLINK LINK
72211
Figure 1-5
5V
Power
connector
LINK LED
LINK LED
RJ-45 10/100BaseT ports
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Hardware Overview
The unit provides the following connectors and indicators:
Note
•
5V power connector.
•
Two RJ-11 FXS (Foreign Exchange Station) ports—The Cisco ATA supports two independent RJ-11
telephone ports that can connect to any standard analog telephone device. Each port supports either
voice calls or fax sessions, and both ports can be used simultaneously.
The Cisco ATA186-I1 and Cisco ATA188-I1 provide 600-ohm resistive impedance. The Cisco
ATA186-I2 and Cisco ATA188-I2 provide 270 ohm + 750 ohm // 150-nF complex impedance. The
impedance option is requested when you place your order and should match your specific application.
If you are not sure of the applicable configuration, check your country or regionaltelephone impedance
requirements.
•
Ethernet ports
– The Cisco ATA 186 has one RJ-45 10BASE-T uplink Ethernet port to connect the
Cisco ATA 186 to a 10/100BASE-T hub or another Ethernet device.
– The Cisco ATA 188 has two Ethernet ports: an RJ-45 10/100BASE-T uplink port to connect the
Cisco ATA 188 to a 10/100BASE-T hub or another Ethernet device and an RJ-45
10/100BASE-T data port to connect an Ethernet-capable device, such as a computer, to the
network.
Note
The Cisco ATA 188 performs auto-negotiation for duplexity and speed and is capable of 10/100 Mbps,
full-duplex operation. The Cisco ATA 186 is fixed at 10 Mbps, half-duplex operation.
•
The Cisco ATA 188 RJ-45 LED shows network link and activity. The LED blinks twice when the
Cisco ATA is first powered on, then turns off if there is no link or activity. The LED blinks to show
network activity and is solid when there is a link.
•
The Cisco ATA 186 RJ-45 LED is solid when the Cisco ATA is powered on and blinks to show
network activity.
•
Function button—The function button is located on the top panel of the unit (see Figure 1-7).
Figure 1-7
Function Button
Function
button
CISCO
72214
ATA 186
ANALOG
TELEPH
ONE AD
APTOR
The function button lights when you pick up the handset of a telephone attached to the Cisco ATA.
The button blinks quickly when the Cisco ATA is upgrading its configuration.
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Software Features
Note
If the function button blinks slowly, the Cisco ATA cannot find the DHCP server. Check your
Ethernet connections and make sure the DHCP server is available.
Pressing the function button allows you to access to the voice configuration menu. For additional
information about the voice configuration menu, see the “Voice Configuration Menu” section on
page 3-15.
Caution
Never press the function button during an upgrade process. Doing so may interfere with the process and
may permanently disable the Cisco ATA.
Software Features
The Cisco ATA supports the following protocols, services and methods:
•
Voice Codecs Supported, page 1-7
•
Additional Supported Signaling Protocols, page 1-8
•
Other Supported Protocols, page 1-8
•
Cisco ATA SIP Services, page 1-8
•
Fax Services, page 1-9
•
Methods Supported, page 1-9
•
Supplementary Services, page 1-10
Voice Codecs Supported
The Cisco ATA supports the following voice codecs (check your other network devices for the codecs
they support):
•
G.711µ-law
•
G.711A-law
•
G.723.1
•
G.729
•
G.729A
•
G.729B
•
G.729AB
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Software Features
Additional Supported Signaling Protocols
In addition to SIP, the Cisco ATA supports the following signaling protocols:
•
H.323
•
Skinny Client Control Protocol (SCCP)
•
Media Gateway Control Protocol (MGCP)
SIP and H.323 share the same software image. SCCP and MGCP also share a software image, which is
separate from the SIP/H.323 image. If you wish to perform a cross-protocol upgrade from SIP to another
signaling image, see the “Upgrading the Signaling Image from a TFTP Server” section on page 8-1.
Other Supported Protocols
Other protocols that the Cisco ATA supports include the following:
•
802.1Q VLAN tagging
•
Cisco Discovery Protocol (CDP)
•
Domain Name System (DNS)
•
Dynamic Host Configuration Protocol (DHCP)
•
Internet Control Message Protocol (ICMP)
•
Internet Protocol (IP)
•
Real-Time Transport Protocol (RTP)
•
Transmission Control Protocol (TCP)
•
Trivial File Transfer Protocol (TFTP)
•
User Datagram Protocol (UDP)
Cisco ATA SIP Services
For a list of required SIP parameters as well as descriptions of all supported Cisco ATA SIP services and
cross references to the parameters for configuring these services, see Chapter 4, “Basic and Additional
SIP Services.”
These services include the following features:
•
IP address assignment—DHCP-provided or statically configured
•
Cisco ATA configuration by means of a TFTP server, web browser, or voice configuration menu.
•
VLAN configuration
•
Cisco Discovery Protocol (CDP)
•
Low-bit-rate codec selection
•
User authentication
•
Configurable tones (dial tone, busy tone, alert tone, reorder tone, call waiting tone)
•
Dial plans
•
Network Address Translation (NAT) Gateway
•
NAT/Port Address Translation (PAT) translation
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Software Features
•
SIP proxy server redundancy
•
Outbound-proxy support
•
DNS SRV support
•
User-configurable, call-waiting, permanent default setting
•
Comfort noise during silence period when using G.711
•
Advanced audio
•
Billable features
•
Caller ID format
•
Ring cadence format
•
Silence suppression
•
Hook-flash detection timing configuration
•
Configurable on-hook delay
•
UDP Type of Service (ToS) configuration
•
Debugging and diagnostic tools
Fax Services
The Cisco ATA supports two modes of fax services, in which fax signals are transmitted using the G.711
codec:
•
Fax pass-through mode—Receiver-side Called Station Identification (CED) tone detection with
automatic G.711A-law or G.711µ-law switching.
•
Fax mode—The Cisco ATA is configured as a G.711-only device.
How you set Cisco ATA fax parameters depends on what network gateways are being used. You may
need to modify the default fax parameter values (see Chapter 7, “Configuring and Debugging Fax
Services”).
Note
Success of fax transmission depends on network conditions and fax modem response to these conditions.
The network must have reasonably low network jitter, network delay, and packet loss rate.
Methods Supported
The Cisco ATA supports the methods listed below. For more information, refer to RFC3261 (SIP:
Session Initiation Protocol).
•
REGISTER
•
INVITE
•
BYE
•
CANCEL
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Installation and Configuration Overview
•
NOTIFY
•
OPTIONS
•
ACK
Supplementary Services
SIP supplementary services are services that you can use to enhance your telephone service. For
information on how to enable and subscribe to these services, see the “CallFeatures” section on
page 5-23 and the “PaidFeatures” section on page 5-24.
For information on how to use these services, see Appendix A, “Using SIP Supplementary Services.”
The following list contains the SIP supplementary services that the Cisco ATA supports:
•
Caller ID
•
Call-waiting caller ID
•
Voice mail indication
•
Making a conference call
•
Call waiting
•
Call forwarding
•
Call return
•
Calling-line identification
•
Unattended transfer
•
Attended transfer
Installation and Configuration Overview
Table 1-1 provides the basic steps required to install and configure the Cisco ATA to make it operational
in a typical SIP environment where a large number of Cisco ATAs must be deployed.
Table 1-1
Overview of the Steps Required to Install and Configure the Cisco ATA and Make it Operational
Action
Reference
1.
Plan the network and Cisco ATA configuration.
2.
Install the Ethernet connection.
3.
Install and configure the other network devices.
4.
Install the Cisco ATA but do not power up the Cisco ATA yet.
5.
Download the desired Cisco ATA release software zip file from Chapter 3, “Configuring the Cisco ATA for SIP”
the Cisco web site, then configure the Cisco ATA.
6.
Power up the Cisco ATA.
7.
Periodically, you can upgrade the Cisco ATA to a new signaling Chapter 8, “Upgrading the Cisco ATA Signaling
image by using the TFTP server-upgrade method or the
Image”
manual-upgrade method.
What the Cisco ATA Package Includes, page 2-2
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2
Installing the Cisco ATA
This section provides instructions for installing the Cisco ATA 186 and Cisco ATA 188. Before you
perform the installation, be sure you have met the following prerequisites:
•
Planned the network and Cisco ATA configuration.
•
Installed the Ethernet connection.
•
Installed and configured the other network devices.
This section contains the following topics:
Note
•
Network Requirements, page 2-2
•
Safety Recommendations, page 2-2
•
What the Cisco ATA Package Includes, page 2-2
•
What You Need, page 2-3
•
Installation Procedure, page 2-3
•
Power-Down Procedure, page 2-6
The term Cisco ATA is used throughout this manual to refer to both the Cisco ATA 186 and the
Cisco ATA 188, unless differences between the Cisco ATA 186 and Cisco ATA 188 are explicitly stated.
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Network Requirements
Network Requirements
The Cisco ATA acts as an endpoint on an IP telephony network. The following equipment is required:
•
Call Control system
•
Voice packet gateway—Required if you are connecting to the Public Switched Telephone Network
(PSTN). A gateway is not required if an analog key system is in effect.
•
Ethernet connection
Safety Recommendations
To ensure general safety, follow these guidelines:
•
Do not get this product wet or pour liquids into this device.
•
Do not open or disassemble this product.
•
Do not perform any action that creates a potential hazard to people or makes the equipment unsafe.
•
Use only the power cube that comes with the Cisco ATA.
Warning
Ultimate disposal of this product should be handled according to all national laws and regulations.
Warning
Read the installation instructions before you connect the system to its power source.
Warning
The plug-socket combination must be accessible at all times because it serves as the main
disconnecting device.
Warning
Do not work on the system or connect or disconnect cables during periods of lightning activity.
For translated warnings, see the Regulatory Compliance and Safety Information for the Cisco ATA 186
and Cisco ATA 188 manual.
What the Cisco ATA Package Includes
The Cisco ATA package contains the following items:
•
Cisco ATA 186 or Cisco ATA 188 Analog Telephone Adaptor
•
Read Me First - ATA Boot Load Information
•
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor at a Glance
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What You Need
•
Regulatory Compliance and Safety Information for the Cisco ATA 186 and Cisco ATA 188
•
5V power adaptor
•
Power cord
Note
The Cisco ATA is intended for use only with the 5V DC power adaptor that comes with the unit.
What You Need
You also need the following items:
•
Category-3 10BASE-T or 100BASE-T or better Ethernet cable. One cable is needed for each
Ethernet connection.
A Category-3 Ethernet cable supports 10BASE-T for up to 100 meters without quality degradation,
and a Category-3 Ethernet cable supports 100BASE-T for up to 10 meters without quality
degradation.
For uplink connections, use a crossover Ethernet cable to connect the Cisco ATA to another Ethernet
device (such as a router or PC) without using a hub. Otherwise, use straight-through Ethernet cables
for both uplink and data port connections.
•
Access to an IP network
•
One or two analog Touch-Tone telephones or fax machines, or one of each
Installation Procedure
After the equipment is in place, see Figure 2-1 (for Cisco ATA 186) or Figure 2-2 (for Cisco ATA 188)
and follow the next procedure to install the Cisco ATA.
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Installation Procedure
PHONE 1
Cisco ATA 186 Rear Panel Connections
PHONE 2
10BaseT
ACT
72212
Figure 2-1
5V
Power outlet
IP network
Analog telephones
(or fax)
5V power
adaptor
Power cord
PHONE 1
Cisco ATA 188 Rear Panel Connections
PHONE 2
LINK
10/100 PC
10/100 UPLINK LINK
72213
Figure 2-2
5V
Power outlet
IP network
Analog telephones
(or fax)
PC
5V power
adaptor
Power cord
Procedure
Step 1
Place the Cisco ATA near an electrical power outlet.
Step 2
Connect one end of a telephone line cord to the Phone 1 input on the rear panel of the Cisco ATA.
Connect the other end to an analog telephone set.
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Installation Procedure
If you are connecting a telephone set that was previously connected to an active telephone line, unplug
the telephone line cord from the wall jack and plug it into the Phone 1 input.
Warning
Caution
Note
Step 3
Note
To reduce the risk of fire, use only No. 26 AWG or larger telecommunication line cord.
Do not connect the Phone input ports to a telephone wall jack. To avoid damaging the Cisco ATA or
telephone wiring in the building, do not connect the Cisco ATA to the telecommunications network.
Connect the Phone port to a telephone only, never to a telephone wall jack.
The telephone must be switched to tone setting (not pulse) for the Cisco ATA to operate properly.
(Optional) Connect the telephone line cord of a second telephone to the Phone 2 input port.
If you are connecting only one telephone to the Cisco ATA, you must use the Phone 1 input port.
Step 4
Connect an Ethernet cable to the uplink RJ-45 connector on the Cisco ATA. For the Cisco ATA 186, this
is the 10BASE-T connector; for the Cisco ATA 188, this is the 10/100UPLINK connector.
Note
Use a crossover Ethernet cable to connect the Cisco ATA to another Ethernet device (such as a router or
PC) without using a hub. Otherwise, use a straight-through Ethernet cable.
Step 5
(Cisco ATA 188 only—optional) Connect a straight-through Ethernet cable from your PC to the 10/100
PC RJ-45 connector on the Cisco ATA.
Step 6
Connect the socket end of the power cord to the 5V DC power adaptor.
Step 7
Insert the power adaptor cable into the power connector on the Cisco ATA.
Warning
Step 8
This product relies on the building’s installation for short-circuit (overcurrent) protection. Ensure that
a fuse or circuit breaker no larger than 120 VAC, 15A U.S. (240VAC, 10A international) is used on the
phase conductors (all current-carrying conductors).
Connect the plug end of the 5V DC power adaptor cord into an electrical power outlet.
When the Cisco ATA is properly connected and powered up, the green activity LED flashes to indicate
network activity. This LED is labeled ACT on the rear panel of the Cisco ATA 186 and is labeled LINK
on the rear panel of the Cisco ATA 188.
Caution
Do not cover or block the air vents on either the top or the bottom surface of the Cisco ATA. Overheating
can cause permanent damage to the unit.
For more information about LEDs and the function button, see the “Hardware Overview” section on
page 1-5.
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Power-Down Procedure
Power-Down Procedure
Caution
If you need to power down Cisco ATA 186 or Cisco 188 at any time, use the following power-down
procedure to prevent damage to the unit.
Procedure
Step 1
Unplug the RJ45 Ethernet cable
Step 2
Wait for 20 seconds.
Step 3
Unplug the power cable.
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3
Configuring the Cisco ATA for SIP
This section describes how to configure the Cisco ATA to operate with the Session Initiation Protocol
(SIP) signaling image and how the Cisco ATA obtains the latest signaling image.
You can configure the Cisco ATA for use with SIP with any of the following methods:
•
By using a TFTP server—This is the Cisco-recommended method for deploying a large number of
Cisco ATAs. This method allows you to set up a unique Cisco ATA configuration file or a
configuration file that is common to all Cisco ATAs. The Cisco ATA can automatically download its
latest configuration file from the TFTP server when the Cisco ATA powers up, is refreshed or reset,
or when the specified TFTP query interval expires.
•
By using manual configuration:
– Voice configuration menu—This is the method you must use if the process of establishing IP
connectivity for the Cisco ATA requires changing the default network configuration settings. These
settings are CDP, VLAN, and DHCP. You also can use the voice configuration menu to review all IP
connectivity settings. The voice configuration menu can also be used when Web access is not
available.
– Web-based configuration—This method is convenient if you plan to deploy a small number of
Cisco ATAs in your network. To use this method, the Cisco ATA must first obtain IP connectivity,
either through the use of a DHCP server or by using the voice configuration menu to statically
configure IP addresses.
This section contains the following topics:
•
Default Boot Load Behavior, page 3-2—This section describes the process that the Cisco ATA
follows by default when it boots up. It is very important to understand this process because, if your
network environment is not set up to follow this default behavior, you need to make the applicable
configuration changes. For example, by default, the Cisco ATA attempts to contact a DHCP server
for the necessary IP addresses to achieve network connectivity. However, if your network does not
use a DHCP server, you must manually configure various IP settings as described in this section.
•
Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation, page 3-3—This
section includes a table of the parameters you can configure for VLAN and CDP settings.
•
Steps Needed to Configure the Cisco ATA, page 3-5—This section provides tables that summarize
the general configuration steps you must follow to configure the Cisco ATA.
•
Configuring the Cisco ATA Using a TFTP Server, page 3-7—This section describes procedures for
configuring the Cisco ATA by using a TFTP server, which is the recommended configuration
method for the deployment of a large number of Cisco ATAs.
•
Voice Configuration Menu, page 3-15—This section includes information on how to obtain basic
network connectivity for the Cisco ATA and how to perform a factory reset if necessary.
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Default Boot Load Behavior
Note
•
Cisco ATA Web Configuration Page, page 3-18—This section shows the Cisco ATA Web
configuration page and contains a procedure for how to configure Cisco ATA parameters using this
interface.
•
Refreshing or Resetting the Cisco ATA, page 3-21—This section gives the procedure (via the Web
configuration page) for refreshing or resetting the Cisco ATA so that your most recent configuration
changes take effect immediately.
•
Upgrading the SIP Signaling Image, page 3-22—This section provides references to the various
means of upgrading your Cisco ATA signaling image.
The term Cisco ATA is used throughout this manual to refer to both the Cisco ATA 186 and the
Cisco ATA 188, unless differences between the Cisco ATA 186 and Cisco ATA 188 are explicitly
stated.
Default Boot Load Behavior
Before configuring the Cisco ATA, you need to know how the default Cisco ATA boot load process
works. Once you understand this process, you will be able to configure the Cisco ATA by following the
instructions provided in this section and in the sections that follow.
All Cisco ATAs are shipped with a bootload signaling-protocol image. However, because this image is
not a fully functional signaling image, the image must be upgraded. The image is designed to be
automatically upgraded by a properly configured TFTP server. To configure the Cisco ATA to
automatically upgrade to the latest signaling image, see the “Upgrading the Signaling Image from a
TFTP Server” section on page 8-1.
In addition, the Cisco ATA obtains its configuration file during the bootload process.
The following list summarizes the default Cisco ATA behavior during its boot-up process:
1.
Note
2.
Note
3.
The Cisco ATA uses the Cisco Discovery Protocol (CDP) to discover which VLAN to enter. If the
Cisco ATA receives a VLAN ID response from the network switch, the Cisco ATA enters that
VLAN and adds 802.1Q VLAN tags to its IP packets. If the Cisco ATA does not receive a response
with a VLAN ID from the network switch, then the Cisco ATA assumes it is not operating in a VLAN
environment and does not perform VLAN tagging on its packets.
If your network environment is not set up to handle this default behavior, make the necessary
configuration changes by referring to the “Specifying a Preconfigured VLAN ID or Disabling
VLAN IP Encapsulation” section on page 3-3.
The Cisco ATA contacts the DHCP server to request its own IP address.
If your network environment does not contain a DHCP server, you need to statically configure
various IP addresses so that the Cisco ATA can obtain network connectivity. For a list of
parameters that you must configure to obtain network connectivity, see Table 3-5 on page 3-16.
For instructions on how to use the voice configuration menu, which you must use to perform this
configuration, see the “Voice Configuration Menu” section on page 3-15.
Also from the DHCP server, the Cisco ATA requests the IP address of the TFTP server.
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Specifying a Preconfigured VLAN ID or Disabling VLAN IP Encapsulation
4.
Note
Note
The Cisco ATA contacts the TFTP server and downloads the Cisco ATA release software that
contains the correct signaling image for the Cisco ATA to function properly.
If you are not using a TFTP server, you need to manually upgrade the Cisco ATA to the correct
signaling image. For information on this procedure, see the “Upgrading the Signaling Image
Manually” section on page 8-2.
5.
The Cisco ATA looks for a Cisco ATA-specific configuration file (designated by the MAC address
of the Cisco ATA and named ata) on the TFTP server and downloads this file if it
exists.
6.
If the Cisco ATA does not find the ata configuration file, it looks for the
atadefault.cfg configuration file and downloads this file if it exists. This file can contain default
values for the Cisco ATA to use.
When the Cisco ATA is downloading its DHCP configuration, the function button on the top panel
blinks.
Specifying a Preconfigured VLAN ID or Disabling VLAN IP
Encapsulation
If you want the Cisco ATA to use a preconfigured VLAN ID instead of using the Cisco Discovery
Protocol to locate a VLAN, or if you want to disable VLAN IP encapsulation, refer to Table 3-1 for a
reference to the parameters and bits you may need to configure. Use the voice configuration menu to
configure these parameters. (See the “Voice Configuration Menu” section on page 3-15 for instructions
on using this menu.) Also, refer to Table 3-2 for a matrix that indicates which VLAN-related parameters
and bits to configure depending on your network environment.
Table 3-1
Parameters and Bits for Preconfiguring a VLAN ID
Parameter and Bits
Reference
OpFlags:
OpFlags, page 5-34
•
Bit 4—Enable the use of user-specified voice VLAN ID.
•
Bit 5—Disable VLAN encapsulation
•
Bit 6—Disable CDP discovery.
VLANSetting:
VLANSetting, page 5-35
•
Bits 0-2—Specify VLAN CoS bit value (802.1P priority) for TCP
packets.
•
Bits 3-5—Specify VLAN CoS bit value (802.1P priority) for UDP
packets
•
Bits 18-29—User-specified 802.1Q VLAN ID
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Table 3-2
VLAN-Related Features and Corresponding Configuration Parameters
VLANSetting
OpFlags Bit 4 OpFlags Bit 5 OpFlags Bit 6 Bits 18-29
Feature
Static VLAN
1
0
1
VLAN ID
CDP-acquired
VLAN
0
0
0
N/A
No VLAN
N/A
1
N/A
N/A
No CDP
N/A
N/A
1
N/A
No CDP and no
VLAN
0
1
1
N/A
N/A indicates that the variable is not applicable to the feature and the setting of this varaible does not affect the feature.
Example
The following procedure shows you how to configure the OpFlags and VLANSetting parameters to
allow the Cisco ATA to use a user-specified VLAN ID. In this example, the voice VLAN ID is 115 (in
decimal format).
Step 1
Set bits 4-6 of the OpFlags parameter to 1, 0, and 1, respectively. This setting translates to the following
bitmap:
xxxx xxxx xxxx xxxx xxxx xxxx x101 xxxx
The remaining bits of the OpFlags parameter, using all default values, make up the following bitmap
representation:
0000 0000 0000 0000 0000 0000 0xxx 0010
Therefore, the resulting value of the OpFlags parameter becomes the following bitmap representation:
0000 0000 0000 0000 0000 0000 0101 0010
In hexadecimal format, this value is 0x00000052.
Step 2
Set bits 18-29 of the VLANSetting parameter to to voice VLAN ID 115. This setting translates to the
following bitmap
xx00 0001 1100 11xx xxxx xxxx xxxx xxxx
where 000001110011 is the binary representation of the demical value 115.
The remaining bits of the VLANSetting parameter, using all default values, make up the following
representation:
00xx xxxx xxxx xx00 0000 0000 0010 1011
Therefore, the resulting value of the VLANSetting parameter becomes the following bitmap
representation:
0000 0001 1100 1100 0000 0000 0010 1011
In hexadecimal format, this value is 0x01cc002b.
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Note
If you are using the voice configuration menu to set the parameters, you must convert hexadecimal
values to decimal values. For example, the OpFlags setting of 0x00000052 is equivalent to 82 in decimal
format, and the VLANSetting of 0x01cc002b is equivalent to 30146603 in decimal format.
Steps Needed to Configure the Cisco ATA
This section contains the following topics:
•
Basic Configuration Steps in a TFTP Server Environment, page 3-5
•
Basic Configuration Steps in a Non-TFTP Server Environment, page 3-6
Basic Configuration Steps in a TFTP Server Environment
Table 3-3 shows the basic steps for configuring the Cisco ATA and making it operational in a typical
SIP environment, which includes a TFTP server.
Table 3-3
Basic Steps to Configure the Cisco ATA in a TFTP Environment
Action
Reference
1.
Download the desired Cisco ATA release software zip file from
the Cisco web site and store it on the TFTP server.
Setting Up the TFTP Server with Cisco ATA
Software, page 3-7
2.
Follow these basic steps to create a unique Cisco ATA
configuration file, which actually entails creating two files:
Creating Unique and Common Cisco ATA
Configuration Files, page 3-8
a. Create a Cisco ATA configuration text file that contains
parameters that are common to all Cisco ATAs in your
network.
b. Create a unique Cisco ATA configuration text file that
contains parameters that are specific to a Cisco ATA.
Make sure to use an include command in the unique
configuration file to pull in values from the common
configuration file.
c. Convert the unique configuration file to binary format.
d. Place the unique binary configuration file on the TFTP
server.
atadefault.cfg Configuration File, page 3-12
3.
Optionally, create a default configuration file called
atadefault.cfg, which the Cisco ATA will download from the
TFTP server only if the unique Cisco ATA file called
ata does not exist on the TFTP server.
4.
Configure the upgradecode parameter so that the Cisco ATA will “Upgrading the Signaling Image from a TFTP
obtain the correct signaling image from the TFTP server when the Server” section on page 8-1
Cisco ATA powers up.
5.
Configure the desired interval for the Cisco ATA to contact the
TFTP server to check for a configuration-file update or an
upgrade of the signaling image file.
Configuring Refresh Interval, page 4-3
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Table 3-3
Basic Steps to Configure the Cisco ATA in a TFTP Environment (continued)
Action
Reference
6.
Configure the method with which the Cisco ATA will locate the
TFTP server at boot up time.
7.
Power up the Cisco ATA.
8.
If you make configuration changes to the Cisco ATA or upgrade
the signaling image on the TFTP server, you can refresh the
Cisco ATA so that these changes take effect immediately.
Otherwise, these changes will take effect when the specified
interval (CfgInterval parameter value) for the TFTP query
expires.
Configuring the Cisco ATA to Obtain its
Configuration File from the TFTP Server, page 3-13
Refreshing or Resetting the Cisco ATA, page 3-21
Basic Configuration Steps in a Non-TFTP Server Environment
Table 3-4 shows the basic steps for configuring the Cisco ATA without using the TFTP server method.
Table 3-4
Basic Steps to Configure the Cisco ATA Without Using the TFTP Server Method
Action
1.
Reference
Download the desired Cisco ATA release software zip file from the Cisco web site:
a. If you are a registered CCO user. go to the following URL:
http://www.cisco.com/cgi-bin/tablebuild.pl/ata186
b. Download the zip file that contains the software for the applicable release and signaling
image you are using. The contents of each file are described next to the file name.
c. Extract the files to the desired location on your PC.
Note
The file that contains the protocol signaling image has an extension of .zup.
2.
Manually upgrade the Cisco ATA to the correct signaling image.
3.
Configure the Cisco ATA by using either one of the manual-configuration methods.
4.
Upgrading the Signaling
Image Manually, page 8-2
•
Voice Configuration
Menu, page 3-15
•
Cisco ATA Web
Configuration Page,
page 3-18
Power up the Cisco ATA.
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Configuring the Cisco ATA Using a TFTP Server
Configuring the Cisco ATA Using a TFTP Server
The TFTP method of configuration is useful when you have many Cisco ATA because you can use a
TFTP server for remote, batch configuration of Cisco ATAs. A TFTP server can host one unique
configuration file for each Cisco ATA.
This section contains the following topics:
•
Setting Up the TFTP Server with Cisco ATA Software, page 3-7
•
Configurable Features and Related Parameters, page 3-7
•
Creating Unique and Common Cisco ATA Configuration Files, page 3-8
•
atadefault.cfg Configuration File, page 3-12
•
Configuring the Cisco ATA to Obtain its Configuration File from the TFTP Server, page 3-13
Setting Up the TFTP Server with Cisco ATA Software
This section provides the procedure for the Cisco ATA administrator to obtain the correct Cisco ATA
software and set up the TFTP server with this software.
Procedure
Step 1
If you are a registered CCO user. go to the following URL:
http://www.cisco.com/cgi-bin/tablebuild.pl/ata186
Step 2
Download the zip file that contains the software for the applicable release and signaling image you are
using. The contents of each file are described next to the file name. Save the zip file onto a floppy disc.
Note
Step 3
The file that contains the protocol signaling image has an extension of .zup.
Extract the signaling files onto the TFTP server. This should be the same TFTP server that will contain
the binary Cisco ATA configuration file that you create (either ata or atadefault.cfg).
Configurable Features and Related Parameters
Table 4-1 on page 4-2 contains a list of all required SIP parameters. These parameters must be properly
configured for the Cisco ATA to work.
For descriptions of important Cisco ATA SIP services that you can configure, and references to their
configuration parameters, see the “Important Basic SIP Services” section on page 4-1 and the
“Additional SIP Services” section on page 4-3.
Table 4-3 on page 4-12 lists, in alphabetical order, various features that you can configure for the
Cisco ATA. Table 4-3 on page 4-12 also includes links to the related parameter that allows you to
configure each of these features. Each link takes you to a detailed description of the parameter that
includes its default values.
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For an example of how to configure parameters for the TFTP Server configuration method, see the
“Creating Unique and Common Cisco ATA Configuration Files” section on page 3-8.
Note
Be sure to configure the UseSIP parameter to the value of 1 to enable the SIP protocol. This parameter
is 0 (for H.323) by default.
Creating Unique and Common Cisco ATA Configuration Files
If you have many Cisco ATAs to configure, a good approach is to create two configuration files:
•
One file that will contain only parameter values unique to a specific Cisco ATA.
•
One file for parameters that will be configured with values common to a group of Cisco ATAs. If
this file is updated, all Cisco ATA devices in this common group can obtain the new configuration
data in a batch-mode environment.
The following procedure demonstrates the steps needed to create these configuration files.
Note
The parameters used in this section help illustrate the process of creating a unique Cisco ATA
configuration file, and do not include all required SIP parameters in the examples. See Chapter 4, “Basic
and Additional SIP Services,” for complete listings and descriptions of required parameters and
additional configurable features. Also, refer back to Table 3-3 on page 3-5 for all main configuration
steps.
Procedure
Step 1
Use the example_uprofile.txt file as a template for creating a text file of values that are common to one
group of Cisco ATAs. The example_uprofile.txt file is included in the software-release zip file and
contains all default values. This file is shown without its annotations in the “Configuration Text File
Template” section on page 5-2.
Copy the example_uprofile.txt file and save it with a meaningful name, such as common.txt.
Step 2
Configure all common parameters by editing the text file as desired. For example, you might configure
some parameters as follows:
ToConfig:0
UseTftp:1
DHCP:1
TFtpURL:10.10.10.1
UseSIP:1
Tip
It is helpful to always include the parameter/value of ToConfig:0 in the Cisco ATA
configuration file so that every time this file is downloaded to the Cisco ATA, it will set
ToConfig to 0, which the appropriate value for this parameter once the Cisco ATA has been
configured. If ToConfig is 1, the Cisco ATA will continue to unnecessarily contact the TFTP
server.
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The settings in this example indicate that a group of Cisco ATAs is using the TFTP server with an IP
address of 10.10.10.1 to obtain their configuration files. These Cisco ATAs will use a DHCP server to
obtain their own IP addresses but not to obtain the TFTP server IP address (because the TftpURL
parameter has a configured value).
Step 3
Save your changes.
Step 4
Use the example_uprofile.txt file again, this time as a template for creating a text file of values that are
specific to one Cisco ATA. For example, you might configure the following parameters:
UserID:8530709
GkorProxy:192.168.1.1
Save this file of Cisco ATA-specific parameters as:
ata.txt
where macaddress is the non-dotted hexadecimal version of the MAC address of the Cisco ATA you are
configuring. This non-dotted hexadecimal MAC address is labeled on the bottom of most Cisco ATAs
next to the word “MAC.” The file name must be exactly 15 characters long. (However, if this filename
is supplied by the DHCP server, the name can be as long as 31 characters and can be any name with
printable ASCII characters.)
If necessary, you can obtain the non-dotted hexadecimal MAC address by using the atapname.exe
command. For information on using the atapname.exe command, see the “Using atapname.exe Tool to
Obtain MAC Address” section on page 3-10. That section includes an example of a dotted decimal MAC
address and its corresponding non-dotted hexadecimal address.
The ata.txt file should contain only those parameters whose values are different
from the file of common parameters. Parameter values in the ata configuration
file will overwrite any manually configured values (values configured through the web or voice
configuration menu) when the Cisco ATA powers up or refreshes.
Note
Step 5
On the top line of the ata.txt file, add an include command to include the name of the
common-parameters file, and save the file.
include:common.txt
UserID:8530709
GkorProxy:192.168.1.1
Step 6
Run the cfgfmt.exe tool, which is bundled with the Cisco ATA software, on the ata.txt
text file to generate the binary configuration file. If you wish to encrypt the binary file for security
reasons, see the “Using the EncryptKey Parameter and cfgfmt Tool” section on page 3-11.
The syntax of the cfgfmt program follows:
Syntax
cfgfmt [-eRC4Password] -tpTagFile input-text-file output-binary-file
– -eRC4Password is the optional RC 4key to encrypt the binary TFTP file provided by the
cfgfmt program (up to eight alphanumeric characters).
– pTagFile is the command used to specify the ptag.dat file that is provided with the
Cisco ATA software version you are running. Search on the keyword ptag to find the complete
name of the ptag file that is included with the Cisco ATA software for the signaling protocol
you are using. Be sure this file resides in the same directory from which you are running the
cfgfmt program. The ptag.dat file is used by cfgfmt.exe to format a text input representation of
the parameter/value pairs to its output binary representation.
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– input-text-file is the input text file representation of the Cisco ATA configuration file.
– output-binary-file is the final output binary file that Cisco ATA uses as the TFTP
configuration file.
Example
cfgfmt -tptag.dat ata0a141e28323c.txt ata0a141e28323c
This example is based on a Cisco ATA MAC address of 10.20.30.40.50.60, which converts to the
two-digit, lower-case hexadecimal representation of each integer as 0a141e28323c. This example also
uses a ptag file name of ptag.dat.
When you convert the ata.txt file to a binary file, the binary file will merge the two text
files to form one Cisco ATA-specific binary configuration file for your Cisco ATA.
If the same parameter is configured with different values in these two files, the value in the
ata.txt file takes precedence over the value in the common.txt file.
Step 7
Store the binary configuration file in the TFTP server root directory.
When the Cisco ATA powers up, it will retrieve its unique configuration file from the TFTP server.
Step 8
If you want to make configuration changes after boot up, repeat the process of creating or editing the
text files containing the desired parameters, then converting the ata.txt text file to the
binary file and storing the binary file on the TFTP server. For the configuration changes to take effect
immediately, refresh the Cisco ATA. (See the “Refreshing or Resetting the Cisco ATA” section on
page 3-21.)
After being refreshed, the Cisco ATA will download the updated ata configuration file.
Note
If you do not perform a refresh procedure, the Cisco ATA will update its configuration the next
time it contacts the TFTP server, which is based on the configured value of the CfgInterval
parameter.
Using atapname.exe Tool to Obtain MAC Address
This bundled tool is useful for converting the dotted decimal version of the Cisco ATA MAC address
(available on the Cisco ATA Web configuration page or from the voice configuration menu code 24#)
to its default Cisco ATA profile name. This name has the following format:
ataxxxxxxxxxxxx
where each xx is the two-digit, lower-case hexadecimal representation of each integer in the dotted,
decimal version of the Cisco ATA MAC address. This is the name you use for the unique Cisco ATA
binary configuration file.
The following command and output show an example of this command.
Command Example
atapname.exe 10.20.30.40.50.60
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Command Output
ata0a141e28323c
Note
The same functionality is available from the voice configuration menu (voice menu code 84#), which
will announce the Cisco ATA profile name.
Using the EncryptKey Parameter and cfgfmt Tool
The EncryptKey parameter encrypts binary files being transferred over TFTP. You can change this key
for each Cisco ATA, so that only one specific Cisco ATA can decode the information.
By default, the Cisco ATA-specific ata configuration file is not encrypted. If encryption
is required, however, you must manually configure the EncryptKey parameter before you boot up the
Cisco ATA so that the TFTP method is secure. Use either the voice configuration menu (see the “Voice
Configuration Menu” section on page 3-15) or the Cisco ATA web configuration page (see the
“Cisco ATA Web Configuration Page” section on page 3-18) to configure the EncryptKey parameter.
Note
Because the factory-fresh ATA cannot accept encrypted configuration files, the first unencrypted file, if
intercepted, can easily be read. (You would still have to know the data structure format in order to
decode the binary information from the unencrypted file.) Therefore, the new encryption key in the
unencrypted file can be compromised.
Set the EncryptKey parameter to a nonzero value. When this value is nonzero, the Cisco ATA assumes
that the binary configuration file on the TFTP server is to be encrypted with this key by means of the
RC4 cipher algorithm. The Cisco ATA will use this key to decrypt the configuration file.
The Cisco ATA EncryptKey parameter and the encryption key used in the cfgfmt tool command syntax
must match.
Note
For security reasons, Cisco recommends that you set the UIPassword parameter (if desired) in the
configuration file and not by using one of the manual configuration methods.
The cfgfmt.exe syntax affects how the EncryptKey parameter is used, as shown in the following
examples. In these examples, input_text is the ata.txt file that you will convert to binary
to create the ata configuration file for the Cisco ATA; output_binary is that binary
ata file, and Secret is the encryption key.
Syntax examples
•
cfgfmt -tpTagFile input-text-file output-binary-file
If input-text-file sets the Cisco ATA EncryptKey parameter to 0, then output-binary-file is not
encrypted. If the input-text-file sets EncryptKey to a non-zero value, then output-binary-file is
encrypted with that value.
•
cfgfmt -eSecret -tpTagFile input-text-file output-binary-file
If the Cisco ATA EncryptKey parameter has the value of 0 or is not included in input-text-file, the
Secret is used to encrypt the output-binary-file. If input-text-file sets the Cisco ATA EncryptKey
parameter to a nonzero value and the -e option is used, then output-binary-file is encrypted with the
EncryptKey parameter set in input-text-file and Secret is ignored.
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•
cfgfmt -E -tpTagFile input-text-file output-binary-file
The -E (uppercase) option means that any value specified for the Cisco ATA EncryptKey parameter
in input-text-file is ignored. However, because Secret is not specified in this example,
output-binary-file is not encrypted. Nevertheless, the EncryptKey parameter and its value, if
specified in input-file-text, will be included in output-binary-file for possible encryption at a later
time.
•
cfgfmt -E -eSecret -tpTagFile input-text-file output-binary-file
The -E (uppercase) option means that any value specified for the Cisco ATA EncryptKey parameter
in input-text-file is ignored and the output-binary-file is encrypted with the Secret key. However,
the EncryptKey parameter and its value, if specified in input-text-file, will be included in
output-binary-file.
atadefault.cfg Configuration File
You can create a configuration file, called atadefault.cfg, that is common to all Cisco ATAs. This
configuration file is applied to a Cisco ATA only if a unique configuration file (ata) does not
exist for the Cisco ATA on the TFTP server during the Cisco ATA power-up procedure.
You can use the atadefault.cfg file to provide limited functionality for when you first install the
Cisco ATA. For example, if your service provider provides the ethernet connection and VoIP telephony
service, you may need to call customer service to activate the service. If the atadefault.cfg file is
configured to provide a direct connection to the customer service center, you can simply pick up the
telephone and wait to be connected without using your regular phone.
The following procedure illustrates how to create the Cisco ATA default configuration file, convert it to
the required binary format that the Cisco ATA can read, and store it on the TFTP server so that the
Cisco ATA will download it during the boot-up process:
Procedure
Step 1
Make a copy of the example_uprofile.txt file and rename it atadefault.txt.
Step 2
Make the desired configuration changes by editing the atadefault.txt file, then save the file.
Step 3
Convert the atadefault.txt file to a binary file by running the cfgfmt.exe tool, which is bundled with the
Cisco ATA software.
Note
Step 4
Be sure to name the output file atadefault.cfg.
Store the binary atadefault.cfg configuration file in the TFTP server root directory.
During the boot-up process, the Cisco ATA will download this file as its configuration file unless it first
finds a Cisco ATA-specific configuration file named for the MAC address of the Cisco ATA.
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Configuring the Cisco ATA Using a TFTP Server
Configuring the Cisco ATA to Obtain its Configuration File from the TFTP Server
This section describes three methods for how the Cisco ATA contacts the TFTP server to obtain its
configuration file:
•
Using a DHCP Server, page 3-13
– The Cisco ATA contacts the DHCP server, which provides the IP address of the TFTP server
– The Cisco ATA uses the DHCP server but the DHCP server does not know about the TFTP
server
•
Without Using a DHCP Server, page 3-15
Using a DHCP Server
When using a DHCP server, configuration settings vary depending on whether or not the DHCP server
is under the control of the Cisco ATA system administrator or the service provider. The simplest
configuration is when the DHCP server is under the control of the Cisco ATA administrator, in which
case the DHCP server provides the IP address of the TFTP server. Depending on who controls the DHCP
server, follow the applicable configuration procedure:
•
Procedure if DHCP Server is Under Control of Cisco ATA Administrator, page 3-13
•
Procedure if DHCP Server is not Under Control of Cisco ATA Administrator, page 3-14
This section also includes the topic:
•
Note
Other DHCP Options You Can Set, page 3-14
If no DHCP server is found and the Cisco ATA is programmed to find one, the function button
continues to blink.
Procedure if DHCP Server is Under Control of Cisco ATA Administrator
Procedure
Step 1
On the DHCP server, set one of the following two options:
•
DHCP option 150 (TFTP server IP address)
•
Standard DHCP option 66 (TFTP server name)
If you use DHCP option 150, the Cisco ATA will ignore DHCP option 66. However, if you use DHCP
option 66, you must turn off DHCP option 150 or set its value to 0.
Note
Step 2
You can turn off the DHCP option 150 request by using the Cisco ATA OpFlags parameter (see
the “OpFlags” section on page 5-34).
Make sure to use default values for the following Cisco ATA parameters:
•
TftpURL=0
•
UseTftp=1
•
DHCP=1
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Configuring the Cisco ATA Using a TFTP Server
This completes the parameter settings and DHCP options you need to configure for this procedure. The
Cisco ATA will contact the DHCP server for the IP address of the TFTP server that contains the
Cisco ATA configuration file.
Procedure if DHCP Server is not Under Control of Cisco ATA Administrator
This is the procedure to use if the DHCP server is not under the control of the Cisco ATA administrator,
which means that the URL of the TFTP server must be manually configured.
Procedure
Step 1
Using the voice configuration menu, set the parameter TftpURL to the IP address or URL of the TFTP
server. For more information on setting the TftpURL parameter, see the “TftpURL” section on page 5-5.
For information about using the Cisco ATA voice configuration menu, see the “Voice Configuration
Menu” section on page 3-15.
Note
If you are not using a DHCP server to provide the TFTP server location, you must manually
configure the TftfURL. You can do this by using the voice configuration menu without first
obtaining network connectivity for the Cisco ATA. If you want to configure this value using the
Web configuration page, you first must obtain network connectivity by using the voice
configuration menu to statically configure IP address information (see the “Voice Configuration
Menu” section on page 3-15).
Step 2
Use the default value of 1 for the Cisco ATA parameter DHCP.
Step 3
Use the default value of 1 for the Cisco ATA parameter UseTftp.
This completes the parameter settings you need to configure for this procedure. The Cisco ATA will
contact the manually configured TFTP server that contains the Cisco ATA configuration file.
Other DHCP Options You Can Set
The following parameters can also be configured with DHCP:
•
Boot file name of DHCP header—The ata binary Cisco ATA configuration file,
which can have a maximum of 31 characters and can be any name with printable ASCII characters
•
Client PC address
•
DHCP option 1—Client Subnet Mask
•
DHCP option 3—Routers on the client’s subnet
•
DHCP option 6—One or two Domain Name servers
•
DHCP option 42—One or two Network Time Protocol servers
•
DHCP option 43—Set this option to identify the protocol (for example, SIP)
•
DHCP Option 60 (DHCP_VENDOR_CLASS_ID)—Use this parameter to identify the type of
Cisco ATA box (ATA186 or ATA188).
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Voice Configuration Menu
Without Using a DHCP Server
Use the following procedure if you are not using a DHCP server in your environment but are still using
a TFTP server to obtain the Cisco ATA configuration file:
Procedure
Step 1
Set the DHCP parameter to 0.
Step 2
Set the UseTFTP parameter to 1.
Step 3
Set the Cisco ATA parameter TftpURL to the IP address or URL of the TFTP server. For more
information on setting the TftpURL parameter, see the “TftpURL” section on page 5-5.
Note
Step 4
If you are not using a DHCP server to provide the TFTP server location, you must manually enter
the TftpUrl using either the voice configuration menu or the Web configuration page.
If you have done already done so, statically configure the following parameters using the voice
configuration menu (see the “Voice Configuration Menu” section on page 3-15). These are the
parameters you need to configure for the Cisco ATA to obtain network connectivity:
•
StaticIP
•
StaticRoute
•
StaticNetMask
Other parameters that are normally supplied by DHCP may be provided statically by configuring their
values. These parameters are:
•
DNS1IP
•
DNS2IP
•
NTPIP
•
AltNTPIP
•
Domain
This completes the parameter settings you need to configure in order for the Cisco ATA to contact the
TFTP server (without using DHCP) that will contain the configuration file for the Cisco ATA.
Voice Configuration Menu
The main reasons to use the voice configuration menu are to establish IP connectivity for the Cisco ATA
if a DHCP server is not being used in your network environment, and to reset the Cisco ATA to its
factory values if necessary. You can also use the voice configuration menu if you need to configure a
small number of parameters or if the web interface and TFTP configuration are not available.
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Voice Configuration Menu
Note
Do not use the voice configuration menu to attempt to change any values that you configured by means
of the TFTP configuration file method. Whenever the Cisco ATA refreshes, it downloads its
ata configuration file or atadefault.cfg default configuration file from the TFTP server,
and the values in either of these files will overwrite the values of any corresponding parameters
configured with the voice configuration menu.
See Chapter 5, “Parameters and Defaults,” for a complete list of parameters and their definitions. Also
see Table 4-3 on page 4-12 for an alphabetical listing of configurable features and references to their
corresponding parameters.
This section contains the following topics:
•
Using the Voice Configuration Menu, page 3-16
•
Entering Alphanumeric Values, page 3-17
•
Resetting the Cisco ATA to Factory Default Values, page 3-18
Using the Voice Configuration Menu
To manually configure the Cisco ATA by using the voice configuration menu and the telephone keypad,
perform the following steps:
Procedure
Step 1
Connect an analog touch-tone phone to the port labeled Phone 1 on the back of the Cisco ATA.
Step 2
Lift the handset and press the function button located on the top of the Cisco ATA. You should receive
the initial voice configuration menu voice prompt.
Step 3
Using the telephone keypad, enter the voice menu code for the parameter that you want to configure or
the command that you want to execute, then press #. For a list of voice menu codes, see Appendix B,
“Voice Menu Codes.”
Table 3-5 lists the menu options that you need to configure basic IP connectivity for the Cisco ATA,
after which you can use the Cisco ATA web configuration page to configure additional parameters.
Note
If you are using the voice configuration menu to statically configure the Cisco ATA IP address,
you must disable DHCP by setting its value to 0.
Table 3-5
Parameters that Provide Basic IP Connectivity for the Cisco ATA
Voice Menu
Number
Features
1
StaticIP—IP address of the Cisco ATA.
2
StaticRoute—Default gateway for the Cisco ATA to use.
10
StaticNetMask—Subnet mask of the Cisco ATA.
20
DHCP—Set value to 0 to disable the use of a DHCP server; set value to 1 to enable
DHCP.
21
Review the IP address of the Cisco ATA.
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Voice Configuration Menu
Table 3-5
Step 4
Parameters that Provide Basic IP Connectivity for the Cisco ATA (continued)
Voice Menu
Number
Features
22
Review the default router for the Cisco ATA to use.
23
Review subnet mask of the Cisco ATA.
Follow the voice prompts and enter the appropriate values, then press the # key.
Note
Use the * key to indicate a delimiter (dot). For example, to enter an IP address of 192.168.3.1,
you would enter 192*168*3*1 on your telephone keypad.
Note
When entering values for a field that contains a hexadecimal value, you must convert the
hexadecimal value to a decimal value in order to enter it into the voice configuration menu
system. For example, to enter the hexadecimal value 0x6A, you would enter the number 106 on
the telephone keypad.
The voice configuration menu repeats the value you entered, then prompts you to press one of the
following keys:
•
1=Change your entered value
•
2=Review your entered value
•
3=Save your entered value
•
4=Review the current saved value
Step 5
Press the # key after you have entered the desired key. If you do not press the # key, the system will
automatically timeout after 10 seconds.
Step 6
Cisco strongly recommends that you set a password. Use the voice menu code 7387277 (SETPASS) to
configure a password through the voice configuration menu, after which you are prompted for the
password whenever you attempt to change a parameter value.
Step 7
After completing the configuration through the voice configuration menu, press the # key to exit.
Step 8
Hang up the telephone. The Cisco ATA configuration refreshes. The function button fast-blinks when
the refresh completes.
Entering Alphanumeric Values
Some voice configuration menu options require you to enter alphanumeric characters. Alphanumeric
entry differs from numeric entry because you must press # after each character selected.
If you need to enter an alphanumeric value, the voice prompt tells you to enter an alphanumeric value;
otherwise, enter a numeric value (0 to 9).
Table 3-6 lists the keys on a telephone keypad and their respective alphanumeric characters.
Using Table 3-6 as a guide, enter the appropriate number key on the telephone keypad as many times as
needed to select the number, letter, or symbol required. For example, to enter 58sQ, you would enter:
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Cisco ATA Web Configuration Page
5 # 8 # 7 7 7 7 7 # 7 7 7 7 7 7 7 # #
Table 3-6
Alphanumeric Characters
Key
Alphanumeric Characters
1
1 ./_\ @*space return +-!,?|~^#=$”‘’%<>[] :;{}()&
2
2abcABC
3
3defDEF
4
4ghiGHI
5
5jklJKL
6
6mnoMNO
7
7pqrsPQRS
8
8tuvTUV
9
9wxyzWXYZ
0
0
Resetting the Cisco ATA to Factory Default Values
It is possible that you may, under some circumstances, want to reset the Cisco ATA to its factory default
values. For example, this is the only way to recover a forgotten password without contacting your Cisco
representative.
To perform a factory reset, you must use the voice configuration menu and follow these steps:
Procedure
Step 1
Press the function button on the Cisco ATA.
Step 2
Press the digits 322873738 (FACTRESET) then press # on your telephone keypad.
Step 3
Press 3 on your telephone keypad to confirm that you want to reset the Cisco ATA, then hang up the phone.
Cisco ATA Web Configuration Page
You can use the Cisco ATA web configuration page in a non-TFTP configuration environment, or in a
TFTP configuration environment as a read-only record of individual customer parameters.
You can display the most recent Cisco ATA configuration file from the TFTP server by opening your
web browser and typing the following:
http:///refresh
where ipaddress is the IP address of the Cisco ATA.
Figure 3-1 shows and example of the Cisco ATA web configuration page, which displays all
configurable parameters. The different colors on the screen are for different parameter groupings, as
described in Chapter 5, “Parameters and Defaults.”
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Cisco ATA Web Configuration Page
Note
Do not use the web configuration page to attempt to change any values that you configured by means of
the TFTP configuration file method. Whenever the Cisco ATA refreshes, it downloads its
ata configuration file or atadefault.cfg default configuration file from the TFTP server,
and the values in either of these files will overwrite the values of any corresponding parameters
configured with the web configuration method.
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Cisco ATA Web Configuration Page
Cisco ATA Web Configuration Page
UIPassword:
*
ToConfig:
0
UseTftp:
0
TftpURL:
0
CfgInterval:
3600
EncryptKey:
*
Dhcp:
1
StaticIP:
0.0.0.0
StaticRoute:
0.0.0.0
StaticNetMask:
0.0.0.0
UID0:
9300
PWD0:
****
UID1:
9301
PWD1:
****
GkOrProxy:
192.168.2.97
Gateway:
0
GateWay2:
0.0.0.0
UseLoginID:
0
LoginID0:
0
LoginID1:
0
AltGk:
0
AltGkTimeOut:
0
GkTimeToLive:
300
GkId:
.
UseSIP:
1
SIPRegInterval:
3600
MaxRedirect:
5
SIPRegOn:
1
NATIP:
0.0.0.0
SIPPort:
5060
MediaPort:
16384
OutBoundProxy:
0
NatServer:
0
NatTimer:
0x00000000
LBRCodec:
0
AudioMode:
0x00040004
RxCodec:
2
TxCodec:
2
NumTxFrames:
2
CallFeatures:
0xffffffff
PaidFeatures:
0xffffffff
CallerIdMethod:
0x00019e60
FeatureTimer:
0x00000000
Polarity:
0x00000000
ConnectMode:
0x00060400
AutMethod:
0x00000000
TimeZone:
17
NTPIP:
0.0.0.0
AltNTPIP:
0.0.0.0
DNS1IP:
0.0.0.0
DNS2IP:
0.0.0.0
UDPTOS:
0x000000a0
SigTimer:
0x01418564
OpFlags:
0x00000002
VLANSetting:
0x0000002b
NPrintf:
192.168.2.159.9300
TraceFlags:
0x00000001
RingOnOffTime:
2,4,25
IPDialPlan:
1
DialPlan:
*St4-|#St4-|911|1>#t8.r9t2-|0
DialTone:
2,31538,30831,3100,3885,1,0 BusyTone:
2,30467,28959,1191,1513,0,4
ReorderTone:
2,30467,28959,1191,1513,0,2 RingBackTone:
2,30831,30467,1943,2111,0,1
CallWaitTone:
1,30831,0,5493,0,0,2400,240
CallCmd:
Af;AH;BS;NA;CS;NA;Df;EB;F
AlertTone:
1,30467,0,5970,0,0,480,480,
85712
Figure 3-1
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Refreshing or Resetting the Cisco ATA
You can access the web configuration page from any graphics-capable browser, such as
Microsoft Internet Explorer or Netscape. This provides easy initial access to the Cisco ATA
configuration within the administrator’s private network.
Follow these steps to set parameters using the web configuration page:
Procedure
Step 1
Make sure that your PC and the Cisco ATA are already networked and visible to each another.
Step 2
Open your web browser.
Step 3
Enter the URL for your configuration page. The default URL for the web server is:
http://IP Address/dev
For example, the configuration page for a Cisco ATA with the IP address 192.168.3.225 is:
http://192.168.3.225/dev
Step 4
Select the values for the items that you want to configure. See Chapter 5, “Parameters and Defaults,” for
a complete list of parameters and their definitions. Also see Table 4-3 on page 4-12 for an alphabetical
listing of configurable features and references to their corresponding parameters.
Note
Cisco strongly recommends that you set a password. Use the UIPassword parameter to configure a
password, after which you are prompted for the password whenever you attempt to change a parameter
value. Configuration parameters cannot be accessed through the voice configuration menu if the
password contains one or more letters and can be changed only by using the web interface or the TFTP
configuration method.
Step 5
Click apply to save your changes.
The Cisco ATA automatically refreshes its configuration.
Step 6
Close your web browser.
Refreshing or Resetting the Cisco ATA
Whenever you make configuration changes to your Cisco ATA configuration file, you can refresh or
reset the Cisco ATA for these configuration changes to immediately take effect. If you do not refresh or
reset the Cisco ATA, the configuration changes will take effect the next time the Cisco ATA contacts
the TFTP server, which occurs based on the configured value of the CfgInterval parameter.
Note
A refresh procedure will update the Cisco ATA configuration file. A reset procedure will also update the
Cisco ATA configuration file, and will additionally power-down and power-up the Cisco ATA. A reset
should not be necessary if your only goal is to update the configuration file.
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Upgrading the SIP Signaling Image
Procedure to Refresh the Cisco ATA
To refresh the Cisco ATA, enter the following command from your web browser:
http:///refresh
where ipaddress is the IP address of the Cisco ATA that you are refreshing.
Procedure to Reset the Cisco ATA
To reset the Cisco ATA, enter the following command from your web browser:
http:///reset
where ipaddress is the IP address of the Cisco ATA that you are resetting.
Upgrading the SIP Signaling Image
For instructions on how to upgrade the Cisco ATA to the most recent SIP signaling image, refer to the
following list:
•
To use the recommended TFTP method of upgrading the Cisco ATA, see the “Upgrading the
Signaling Image from a TFTP Server” section on page 8-1.
•
In the rare instance that you are not using the TFTP server to configure the Cisco ATA and to obtain
software upgrades, you must manually upgrade to the latest signaling image immediately after the
Cisco ATA boots up. In this case, see the “Upgrading the Signaling Image Manually” section on
page 8-2.
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C H A P T E R
4
Basic and Additional SIP Services
This section provides information about key basic and additional SIP services that the Cisco ATA
supports:
Note
•
Important Basic SIP Services, page 4-1—This section includes a list of parameters that you must
configure in order for the Cisco ATA to function in a SIP environment.
•
Additional SIP Services, page 4-3—This section contains information about additional, commonly
used SIP features, with references to the parameters for configuring these services.
•
Complete Reference Table of all Cisco ATA SIP Services, page 4-12—This section contains a
complete listing of Cisco ATA services supported for SIP, and includes cross references to the
parameters for configuring these services. This section includes services not described in the
sections about the key basic SIP services and the commonly used additional SIP services.
The term Cisco ATA refers to both the Cisco ATA 186 and the Cisco ATA 188, unless otherwise stated.
Important Basic SIP Services
This section provides descriptions and cross references for configuring required SIP parameters and also
for configuring other important basic SIP services:
•
Required Parameters, page 4-1
•
Establishing Authentication, page 4-2
•
Setting the Codec, page 4-3
•
Configuring Refresh Interval, page 4-3
Required Parameters
If the UseSIP parameter is set to 1 (you are using the SIP protocol), you need to supply values for the
required SIP parameters shown in Table 4-1. The Parameter column provides the name of the parameter
and a cross reference which provides a more-detailed description of the parameter.
Note
See Chapter 5, “Parameters and Defaults,” for information about additional Cisco ATA parameters.
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Important Basic SIP Services
Table 4-1
Required SIP Parameters and Defaults
Parameter
Value Type
Description
Voice
Menu
Access Minimum Maximum
Code
Value
Value
Default
SIPRegInterval,
page 5-15
Integer
Seconds between registration renewal
203
1
86400
3600
MAXRedirect,
page 5-15
Integer
Maximum number of times to try
redirection
202
0
10
5
SIPRegOn, page
5-16
Integer
Enable SIP registration
204
0
1
0
NATIP, page 5-16
IP address
WAN address of the attached router/NAT;
currently only used to support SIP behind
a NAT.
200
0
255
0.0.0.0
SIPPort, page
5-17
Integer
Port to listen for incoming SIP requests
201
1
65535
5060
MediaPort, page
5-17
Integer
Base port to receive RTP media; only used 202
to support SIP behind a NAT
1
65535
16384
OutBoundProxy,
page 5-17
Alphanumeric
string
Proxy server for all outbound SIP requests. 206
—
—
0
GkOrProxy, page
5-10
Alphanumeric
string
SIP proxy server address or registrar
address.
5
—
—
0
UseSIP, page 5-14 Boolean
Set to 1 for SIP mode.
38
—
—
0
ToConfig, page
5-4
Set to 0 after you have completed
configuration of the Cisco ATA. If this
value remains at 1, the Cisco ATA will
unnecessarily continue to contact the
TFTP server.
80001
—
—
1
Boolean
All SIP requests are sent to
OutBoundProxy, when configured, instead
of to the configured GkOrProxy.
Establishing Authentication
The Cisco ATA supports two levels of authentication, depending on the setting of the UseLoginID
parameter:
•
If UseLoginID is set to 0, the user ID (UID0 or UID1) is used with a user-supplied password (PWD0
or PWD1) for authentication.
•
If UseLoginID is set to 1, you must supply a login ID (LoginID0 or LoginID1) and a password
(PWD0 or PWD1) for authentication.
Related Configuration Parameters
•
UseLoginID, page 5-11
•
UID0, page 5-9
•
UID1, page 5-9
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Additional SIP Services
•
LoginID0, page 5-12
•
LoginID1, page 5-12
•
PWD0, page 5-9
•
PWD1, page 5-10
Setting the Codec
The LBRCodec (low-bit-rate codec) parameter determines whether the G.723 or G.729A codec, in
addition to G.711A-law and G.711µ-law, can be used for receiving and transmitting. For configuration
information, see the “LBRCodec” section on page 5-20.
Configuring Refresh Interval
When the value specified in the CfgInterval parameter is reached, the Cisco ATA attempts to refresh its
configuration file from the TFTP server. By opening a web page for the Cisco ATA, you can perform a
refresh before the scheduled refresh. Set the CfgInterval parameter to an interval value (in seconds) for
refreshing the Cisco ATA configuration file. Cisco recommends that the interval be semi-random to
prevent many simultaneous contacts with the TFTP server. For more information, see the “CfgInterval”
section on page 5-5.
When the Cisco ATA contacts the TFTP server, it also checks to see if an upgrade signaling image has
been placed on the TFTP server. If such an image exists, the Cisco ATA will download this image.
Additional SIP Services
This section describes additional SIP services and, where applicable, provides configuration information
and cross references to the parameters for configuring these services. These services are listed
alphabetically.
•
Advanced Audio Configuration, page 4-4
•
Billable Features, page 4-4
•
Comfort Noise During Silence Period When Using G.711, page 4-5
•
Configurable Hook Flash Timing, page 4-5
•
Configurable Mixing of Call Waiting Tone and Audio, page 4-5
•
Configurable On-hook delay, page 4-5
•
Debugging Diagnostics, page 4-5
•
Dial Plan, page 4-6
•
Disabling Access To The Web Interface, page 4-6
•
Distinctive Ringing, page 4-6
•
DNS SRV Support, page 4-6
•
Hardware Information Display, page 4-7
•
NAT Gateway, page 4-7
•
NAT/PAT Translation, page 4-7
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•
Network Timing, page 4-8
•
OutBoundProxy Support, page 4-8
•
Progress Tones, page 4-8
•
Receiver-tagged VIA header, page 4-9
•
Repeat Dialing on Busy Signal, page 4-9
•
SIP Proxy Server Redundancy, page 4-10
•
Stuttering Dial Tone on Unconditional Call Forward, page 4-10
•
User Configurable Call Waiting Permanent Default Setting, page 4-10
•
User Configurable Timeout On No Answer for Call Forwarding, page 4-10
•
Setting Up and Placing a Call Without Using a SIP Proxy, page 4-11
Advanced Audio Configuration
The UDPTOS (specifies the default IP precedence of UDP packets) and AudioMode (audio operating
mode) parameters allow you to tune audio configuration.
Related Parameters
UDPTOS, page 5-32
AudioMode, page 5-20
Billable Features
You can customize specific features on a subscription basis by changing the values of specific bits in
several different parameters. Table 4-2 contains a list of billable features and their related parameters:
Table 4-2
Billable Features and Related Parameters
Feature
Related Parameters
Call Conferencing
PaidFeatures, page 5-24, CallFeatures, page 5-23
Call Forwarding
PaidFeatures, page 5-24, CallFeatures, page 5-23, ConnectMode, page 5-28,
SigTimer, page 5-32
Call Transfer
PaidFeatures, page 5-24, CallFeatures, page 5-23
Call Waiting
PaidFeatures, page 5-24, CallFeatures, page 5-23, SigTimer, page 5-32
Caller ID
PaidFeatures, page 5-24, CallFeatures, page 5-23, CallerIdMethod, page 5-25
Call Return
ConnectMode, page 5-28, PaidFeatures, page 5-24, CallFeatures, page 5-23
Polarity
Polarity, page 5-27
Voice Mail Indicator PaidFeatures, page 5-24, CallFeatures, page 5-23
Note
CallWaitCallerID is an obsolete parameter. Do not use it.
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Comfort Noise During Silence Period When Using G.711
When silence suppression is turned on in ITU G.711, the Cisco ATA calculates and transmits its noise
level to the far end to enable the remote endpoint to generate the appropriate amount of comfort noise.
This provides the remote user with a similar experience to that of a PSTN call and prevents silent gaps
when neither party is talking.
Related Parameter
AudioMode, page 5-20—Bit 0 disables/enables silence suppression.
Configurable Hook Flash Timing
This feature provides the ability to adjust the hook-flash timing to meet local requirements.
Related Parameter
SigTimer, page 5-32—Bits 26 and 27 are for configuring the minimum on-hook time required for a hook
flash event, and bits 28 through 31 are for configuring maximum on-hook time.
Configurable Mixing of Call Waiting Tone and Audio
This feature allows the call-waiting tone to be mixed with the audio in an active call. Therefore, the
call-waiting tone will sound without a pause in the audio.
Related Parameter
ConnectMode, page 5-28—Bit 24
Configurable On-hook delay
This feature is available only for the recipient (callee) of a call. If the callee picks up the phone and then
later hangs up to retrieve another call, the hang-up is not considered on-hook until the specified delay
expires.
Related Parameter
FeatureTimer, page 5-26—Bits 8 to 12
Debugging Diagnostics
You can use the following parameters to troubleshoot operation issues:
•
NPrintf, page 5-36—Specify the IP address and port where debug information is sent.
•
TraceFlags, page 5-36—Use to turn on specific trace features.
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Dial Plan
You can set specific dial plan rules and timeout values. Many of these values are determined on a
country-by-country basis.
Related Parameter
DialPlan, page 5-38
Disabling Access To The Web Interface
To prevent tampering and unauthorized access to the Cisco ATA configuration, the Cisco ATA built-in
web server can be disabled.
Related Parameter
OpFlags, page 5-34—Bit 7
Distinctive Ringing
This feature allows a user to identify a caller based on the ringing pattern the user selects for the
incoming number.
This feature is dependent on the proxy or remote UA, including the Alert-Info header with the
appropriate value in the INVITE message. The Cisco ATA supports standard distinctive ringing pattern
1 to 5 as defined in the standard GR-506-CORE.
The following Alert-Info header values are allowed:
•
Bellcore-dr1
•
Bellcore-dr2
•
Bellcore-dr3
•
Bellcore-dr4
•
Bellcore-dr5
If the Alert-Info header value is not recognized, the Cisco ATA plays the regular ring tone, Bellcore-dr1.
Note
The Bellcore-dr5 ringing pattern is the same as the Bellcore-dr1 ringing pattern.
DNS SRV Support
The Cisco ATA supports DNS SRV lookup for the SIP proxy server. If the GkOrProxy parameter value
begins with _sip._udp. or sip.udp., the Cisco ATA performs a DNS SRV lookup for the SIP proxy server.
A DNS SRV lookup results in one of the following conditions:
•
Zero host is returned or DNS SRV lookup failed. The Cisco ATA then performs a regular DNS
A-record lookup for the given name.
•
One host is returned. The single host is used as the primary proxy and AltGk is the backup proxy,
if specified.
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•
Two or more hosts are returned. The two hosts with the highest priorities are used as the primary
and backup proxy servers (AltGk is ignored in this case).
Related Parameters
•
GkOrProxy, page 5-10
•
AltGk, page 5-13
Hardware Information Display
Cisco ATA hardware information is displayed in the lower-left corner of the Cisco ATA Web
configuration page.
NAT Gateway
Network Address Translation (NAT) supports port mapping and forwarding to standard default SIP
signaling port 5060 and media base port 16384, or other ports as configured in the Cisco ATA. Media
ports are evenly numbered from the base port. NAT must support multiple port mappings. The
Cisco ATA can use up to four media ports to handle conference calls on both lines. For example, if media
base port 16384 is used for one call, the next call uses port 16386 and other calls will use ports 16388
and 16390.
Note
Routers such as D-Link, WinRoute, and WinProxy may not route correctly if both caller and callee are
behind the same NAT.
To configure the Cisco ATA to work in a NAT environment, modify the following parameters:
•
StaticRoute, page 5-7—Enter the LAN IP address of the NAT through which the Cisco ATA will
communicate.
•
NATIP, page 5-16—Enter the WAN IP address of the NAT through which all external SIP user
agents will communicate.
•
SIPPort, page 5-17—Enter a new port for SIP messages (optional).
•
MediaPort, page 5-17—Enter a new base port for RTP media (optional).
NAT/PAT Translation
To maintain Network Address Translation/Port Address Translation (NAT/PAT) for a session, the
Cisco ATA can be configured to periodically send a dummy UDP packet to a server (the Cisco ATA
does not expect any response from the server).
Related Parameters
•
NatTimer, page 5-19—Bits 0 to 11 are for specifying the retransmission period.
•
NatServer, page 5-18—Specify the server to which the dummy packet is sent.
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Network Timing
You can fine tune your network timing with the following parameters:
•
TimeZone, page 5-30—Use for time-stamping incoming calls (offset from Greenwich Mean Time)
with local time.
•
NTPIP, page 5-30—Use for configuring the IP address of the Network Time Protocol server. NTP
is a protocol built on top of TCP that ensures accurate local time-keeping with reference to radio
and atomic clocks located on the Internet.
•
AltNTPIP, page 5-31—Use to configure an alternate NTP server IP address.
•
ConnectMode, page 5-28—Used to control the connection mode of the SIP protocol.
OutBoundProxy Support
If the OutBoundProxy parameter is a fully qualified domain name (FQDN), and DNS returns multiple
IP addresses, the first IP address is used as the primary outbound proxy and the second IP address as the
secondary outbound proxy. If OutBoundProxy is an IP address or if DNS returns only one IP address,
then a backup outbound proxy is not available. The AltGkTimeOut parameter determines the backup
proxy timeout value for the outbound proxy.
If the backup proxy fails, the Cisco ATA automatically switches back to the primary proxy if the unit
has been using the backup proxy for at least 30 seconds. This effectively prevents the Cisco ATA from
switching indefinitely between failing primary and failing backup proxies for the same transactions.
Switching between primary and secondary proxies can occur only for initial INVITE and REGISTER
requests. Other requests, such as CANCEL, BYE, ACK, and re-INVITE, do not retry the backup proxy
but give up if the current proxy fails.
When OutBoundProxy is enabled, the Cisco ATA determines whether to retry to connect with the
backup OutBoundProxy or backup SIP proxy if the INVITE or REGISTER requests fail. If the reason
for failure is an ICMP error (such as an unreachable host), the Cisco ATA retries with the backup
outbound proxy. If failure is due to timeout while waiting for a response or a 5xx response, the
Cisco ATA retries the backup SIP proxy.
Related Parameter
•
OutBoundProxy, page 5-17
•
AltGkTimeOut, page 5-13
Progress Tones
Values for the following parameters (all defined in the “Call-Progress Tone Parameters” section on
page 5-42) must be determined based on the country in which the Cisco ATA is located:
•
DialTone
•
BusyTone
•
ReorderTone
•
RingBackTone
•
CallWaitTone
•
AltertTone
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Receiver-tagged VIA header
You can disable or enable the processing the received = parameter in the Via header. This feature is
disabled by default.
Related Parameter
ConnectMode, page 5-28—Bit 22
Repeat Dialing on Busy Signal
This feature allows the Cisco ATA to repeatedly call a busy number at a periodic interval for a specific
length of time. Both the interval and total time can be specified by the user.
To use this feature, configure FeatureTimer bits 0-7 and add the new command/action values "#37#;kA"
to the existing “H” context and “5;jA” to the existing “S” context in the CallCmd parameter.
This feature is invoked by pressing 5 after the busy tone sounds. The caller then gets a beep confirmation
followed by silence. When the subscriber hangs up, the Cisco ATA starts to redial at the interval
specified in FeatureTimer bits 4-7. When the called party rings, the caller is notified with a special ring.
If the called party picks up the call first, the called party receives a ringback. If the caller picks up the
call first, the caller receives the ringback. This feature is automatically cancelled when the called party
rings.
Note
For this feature to work properly, the remote user agent server must return a 486 (Busy Here) response
to an INVITE request if it detects that the remote party (IP or PSTN) is busy. If the server returns a 183
(Session Progress) response with an SDP before a 486, the Cisco ATA considers the call successful and
automatically cancels repeat dialing.
Related Parameters
•
FeatureTimer, page 5-26—Bits 0 to 3 control the maximum time the Cisco ATA redials a number.
•
FeatureTimer, page 5-26—Bits 4 to 7 control the interval between each redial that the Cisco ATA
performs.A value of zero (0) sets the default redial interval to 15 seconds.
•
CallCmd, page 5-47—The following context commands are used as follows:
Note
Parameter:
Context:
Command/action:
Description:
repeat dialing.
CallCmd
S (may also include 'a' or 'b')
5;jA
This context command adds the service activation code to enable
Parameter:
Context:
Command/action:
Description:
repeat dialing
CallCmd
H
#37#;kA
This context command adds the service deactivation code to disable
For complete information about call commands, see Chapter 6, “Call Commands.”
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SIP Proxy Server Redundancy
SIP proxy server redundancy can be enabled by entering a fully qualified domain name (FQDN) or IP
address (and optional port number) in the GkOrProxy and AltGk parameters, and by configuring the
AltGkTimeOut parameter. If you provide hostnames for GkOrProxy or AltGk, the names are resolved
by the configured DNS. DNS results are hard-coded in cache memory for 10 minutes.
If DNS returns multiple IP addresses, the Cisco ATA uses only the first IP address. If AltGk is set to 0
(disabled) and DNS returns two or more IP addresses for GkOrProxy, then the Cisco ATA uses the first
IP address as the primary proxy and the second IP address as the secondary proxy. If GkOrProxy is an
IP address or DNS returns one IP address, then the backup SIP proxy is not available. A special case
exists if GkOrProxy and AltGk are the same values and are not IP addresses. In this case, the AltGk
parameter is assumed to have the value 0.
Related parameters
•
GkOrProxy, page 5-10
•
AltGk, page 5-13
•
AltGkTimeOut, page 5-13
Stuttering Dial Tone on Unconditional Call Forward
If unconditional call forwarding is enabled, the Cisco ATA plays a continuous stuttering dial tone when
the telephone handset is picked up. This reminds the user that all incoming calls are forwarded to another
number. For more information, see the “Call Forwarding in the United States” section on page A-5 and
the “Call Forwarding in Sweden” section on page A-6.
User Configurable Call Waiting Permanent Default Setting
This feature allows you to specify the default call-waiting setting for every call on a permanent basis by
means of the service activation and deactivation codes.
Related Parameter
ConnectMode, page 5-28—Bit 23
User Configurable Timeout On No Answer for Call Forwarding
This feature allows you to specify the timeout before a call is forwarded to another number on no answer.
This feature is activated by entering the service activation code followed by the phone number and delay.
The entry sequence is as follows:
* #
Delay can be from 1 to 255 seconds. If the delay is zero (0) or not provided by the user, the delay
specified in the SigTimer parameter (bits 20-25), which has a default value of 20 seconds, is in effect.
Example
Using the U.S. Call Command parameter string, the U.S. service activation code is #75 and the
deactivation code is #73.
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To forward calls to the number 555-1212 after a no-answer for 15 seconds, enter the following:
#755551212*15#
To deactivate this feature, enter the following:
#73
Related Parameter
SigTimer, page 5-32—Bits 20 to 25
Setting Up and Placing a Call Without Using a SIP Proxy
The Cisco ATA supports direct IP-to-IP calls without using a SIP proxy. When a call is placed, the Cisco
ATA sends the INVITE request directly to the remote user agent and exepcts the usual 100/180/200
responses from the user agent.
This section contains the following topics:
•
Configuration, page 4-11
•
Placing an IP Call, page 4-12
Configuration
To perform the necessary configuration of the Cisco ATA, follow this procedure:
Procedure
Step 1
Open your Web browser.
Step 2
Enter the URL: http:///dev
where Cisco_ATA_IP_address is the IP address of your Cisco ATA. This takes you to the Cisco ATA
Web configuration page.
Step 3
Step 4
Configure the following parameters as shown:
•
GkOrProxy, page 5-10—Set to the value of 0 (zero).
•
UID0, page 5-9—Set to the unique telephone number of the Phone 1 port of the Cisco ATA.
•
UID1, page 5-9—Set to the unique telephone number of the Phone 2 port of the Cisco ATA.
•
UseSIP, page 5-14—Set to 1 to enable SIP mode.
•
SIPRegOn, page 5-16—Set to 0 to disable SIP registration with a SIP proxy server.
Click the Apply button to save these changes.
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Placing an IP Call
To place an IP call, dial the telephone number and the IP address of the remote user agent. The dial
format is shown below:
Dial Format
**#
Use the star (*) key on the telephone keypad to represent the dot (.) in an IP address. Use the pound (#)
key on the telephone keypad to terminate the dial string and place the call.
Note
URL dialing is not supported.
Example
To place a call to a user agent with an ID of 408-555-1212 at IP address 192.168.1.100, you would enter
the following string on your telephone keypad:
4085551212**192*168*1*100#
Complete Reference Table of all Cisco ATA SIP Services
Table 4-3 is a reference table that lists all configurable features for the Cisco ATA (using SIP), and
includes links to the detailed descriptions of the parameters used for configuring these features.
Table 4-3
Configurable Features and Related Parameters
Configurable Feature
Related Parameter
802.1Q packet tagging
VLANSetting, page 5-35
Audio compression and decompression
LBRCodec, page 5-20
Backup proxy configuration
AltGk, page 5-13
Backup proxy timeout
AltGkTimeOut, page 5-13
Call forward enable/disable
ConnectMode, page 5-28
Call forwarding—Maximum times allowed
MAXRedirect, page 5-15
Call commands
CallCmd, page 5-47, Chapter 6, “Call
Commands”
Call features
CallFeatures, page 5-23
Caller ID format
CallerIdMethod, page 5-25
Call waiting
SigTimer, page 5-32
Call-waiting call ring timeout
FeatureTimer, page 5-26
Call-waiting state specified
ConnectMode, page 5-28
Cisco Discovery Protocol
OpFlags, page 5-34
CNG tone detection
AudioMode, page 5-20
Configuration update interval
CfgInterval, page 5-5
Debug messages—configuring host
NPrintf, page 5-36
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Table 4-3
Configurable Features and Related Parameters (continued)
Configurable Feature
Related Parameter
Dial plan commands
DialPlan, page 5-38
Domain name server
DNS1IP, page 5-31
DNS hostname lookup
ConnectMode, page 5-28
DTMF method
AudioMode, page 5-20
Encryption
EncryptKey, page 5-6
Fax CED tone
AudioMode, page 5-20
Fax mode on a per-call basis
CallFeatures, page 5-23,
PaidFeatures, page 5-24
Fax pass-through
AudioMode, page 5-20,
ConnectMode, page 5-28
G.711 codec
AudioMode, page 5-20
Hook flash
AudioMode, page 5-20, SigTimer, page 5-32
IDs for phone lines
UID0, page 5-9,
UID1, page 5-9
IP-like address in dial plan
IPDialPlan, page 5-38
Login ID
LoginID0, page 5-12,
LoginID1, page 5-12
Low bit-rate codec
LBRCodec, page 5-20
Mixing of tones
ConnectMode, page 5-28
Network Address Translation (NAT)
server—Maintain during session
NatServer, page 5-18
NSE payload number
ConnectMode, page 5-28
NTP IP address
NATIP, page 5-16
On-hook delay
FeatureTimer, page 5-26
Outbound proxy
OutBoundProxy, page 5-17
Paid features
PaidFeatures, page 5-24
Passwords for phone lines
PWD0, page 5-9,
PWD1, page 5-10
Polarity
Polarity, page 5-27
Polarity reversal before and after caller ID signal CallerIdMethod, page 5-25
Received = tag enable/disable
ConnectMode, page 5-28
Receiving-audio codec preference
RxCodec, page 5-21
Redial time if line is busy
FeatureTimer, page 5-26
Refresh Cisco ATA using Web server
OpFlags, page 5-34
REGISTER messages
ConnectMode, page 5-28
Registration removal
ConnectMode, page 5-28
Reset Cisco ATA using Web server
OpFlags, page 5-34
Retransmission interval for NAT server
NatTimer, page 5-19
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Table 4-3
Configurable Features and Related Parameters (continued)
Configurable Feature
Related Parameter
Dial plan commands
DialPlan, page 5-38
Domain name server
DNS1IP, page 5-31
DNS hostname lookup
ConnectMode, page 5-28
DTMF method
AudioMode, page 5-20
Encryption
EncryptKey, page 5-6
Fax CED tone
AudioMode, page 5-20
Fax mode on a per-call basis
CallFeatures, page 5-23,
PaidFeatures, page 5-24
Fax pass-through
AudioMode, page 5-20,
ConnectMode, page 5-28
G.711 codec
AudioMode, page 5-20
Hook flash
AudioMode, page 5-20, SigTimer, page 5-32
IDs for phone lines
UID0, page 5-9,
UID1, page 5-9
IP-like address in dial plan
IPDialPlan, page 5-38
Login ID
LoginID0, page 5-12,
LoginID1, page 5-12
Low bit-rate codec
LBRCodec, page 5-20
Mixing of tones
ConnectMode, page 5-28
Network Address Translation (NAT)
server—Maintain during session
NatServer, page 5-18
NSE payload number
ConnectMode, page 5-28
NTP IP address
NATIP, page 5-16
On-hook delay
FeatureTimer, page 5-26
Outbound proxy
OutBoundProxy, page 5-17
Paid features
PaidFeatures, page 5-24
Passwords for phone lines
PWD0, page 5-9,
PWD1, page 5-10
Polarity
Polarity, page 5-27
Polarity reversal before and after caller ID signal CallerIdMethod, page 5-25
Received = tag enable/disable
ConnectMode, page 5-28
Receiving-audio codec preference
RxCodec, page 5-21
Redial time if line is busy
FeatureTimer, page 5-26
Refresh Cisco ATA using Web server
OpFlags, page 5-34
REGISTER messages
ConnectMode, page 5-28
Registration removal
ConnectMode, page 5-28
Reset Cisco ATA using Web server
OpFlags, page 5-34
Retransmission interval for NAT server
NatTimer, page 5-19
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Table 4-3
Configurable Features and Related Parameters (continued)
Configurable Feature
Related Parameter
Retry interval if line is busy
FeatureTimer, page 5-26
Ringback tone—send to caller
ConnectMode, page 5-28
Ring-cadence pattern
RingOnOffTime, page 5-37
RTP media port
MediaPort, page 5-17
RTP packet size
NumTxFrames, page 5-22
RTP statistics
TraceFlags, page 5-36
Secondary domain name server
DNS2IP, page 5-31
Silence compression
AudioMode, page 5-20
SIP call return
ConnectMode, page 5-28
SIP mode
UseSIP, page 5-14
SIP proxy registrar address
GkOrProxy, page 5-10
SIP proxy registration renewal
SIPRegInterval, page 5-15
SIP registration enable/disable
SIPRegOn, page 5-16
SIP-request listening port
SIPPort, page 5-17
Static network router probe
OpFlags, page 5-34
TFTP file—not using internally generated name
OpFlags, page 5-34
Timeout values
SigTimer, page 5-32
Time zone offset
TimeZone, page 5-30
Tones: BusyTone, CallWaitTone
AlertTone, DialTone, ReorderTone, and
RingBackTone parameters
Call-Progress Tone Parameters, page 5-42
Tracing
TraceFlags, page 5-36
Transmitting-audio codec preference
TxCodec, page 5-22
UDP packet default IP precedence
UDPTOS, page 5-32
VLAN encapsulation
OpFlags, page 5-34
VLAN mode
OpFlags, page 5-34
WAN address of NAT
NATIP, page 5-16
Web configuration—disallowing
OpFlags, page 5-34
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5
Parameters and Defaults
This section provides information on the parameters and defaults that you can use to create your own
Cisco ATA configuration file. This section also includes the voice configuration menu code for each
parameter that has such a code, and each category of parameter type lists the color portion of the web
configuration screen where the parameter is located. Types of parameters include:
•
User Interface (UI) Parameter, page 5-3
•
Configuration Parameter, page 5-4
•
Parameters for Configuration Method, page 5-4
•
Network Parameters, page 5-6
•
Account Information Parameters, page 5-8
•
Backup Server Parameters, page 5-13
•
SIP Parameters, page 5-14
•
Operating Parameters, page 5-19
•
Optional Feature Parameters, page 5-35
The following list contains general configuration information:
•
Your configuration file must begin with #txt.
•
The Cisco ATA uses the following parameter types:
– Alphanumeric string
– Array of short integers
– Boolean (1 or 0)
– Bitmap value—unsigned hexadecimal integer (for specifying bits in a 32-bit integer)
– Extended IP address—IP address followed by port number (for example, 192.168.2.170.9001)
– IP address (e.g. 192.168.2.170)
– Integer (32-bit integer)
– Numeric digit string
Note
The term Cisco ATA is used throughout this manual to refer to both the Cisco ATA 186 and the
Cisco ATA 188, unless differences between the Cisco ATA 186 and Cisco ATA 188 are explicitly stated.
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Parameters and Defaults
Configuration Text File Template
Note
This section contains recommended values for the United States and Sweden as configuration examples
for certain parameters. For information about other countries, contact the Cisco equipment provider for
a specific country.
Configuration Text File Template
This is a listing of the example_uprofile.txt text file, without its annotations, that comes bundled with
the Cisco ATA software.
You can make a copy of this file and use it as a template for creating your own default configuration file
or Cisco ATA-specific configuration file. For instructions on how to create these configuration files, see
the “Creating Unique and Common Cisco ATA Configuration Files” section on page 3-8.
The example_uprofile.txt file contains all the Cisco ATA default values. The sections that follow this
listing describe all the parameters in this file.
#txt
UIPassword:0
UseTftp:1
TftpURL:0
cfgInterval:3600
EncryptKey:0
ToConfig:0
upgradecode:0,0x301,0x0400,0x0200,0.0.0.0,69,0,none
upgradelang:0,0x301,0x0400,0x0200,0.0.0.0,69,0,none
Dhcp:1
StaticIp:0
StaticRoute:0
StaticNetMask:0
GkOrProxy:0
AltGk:0
AltGkTimeOut:0
GkTimeToLive:300
GateWay:0
GateWay2:0.0.0.0
UseLoginID:0
UID0:0
UID1:0
PWD0:0
PWD1:0
LoginID0:0
LoginID1:0
GkId:.
RxCodec:1
TxCodec:1
LBRCodec:0
AudioMode:0x00150015
NumTxFrames:2
CallWaitCallerId:0x003c33d0
Polarity: 0
ConnectMode:0x00060000
AutMethod:0
TimeZone:17
NTPIP:0
AltNTPIP:0
DNS1IP:0.0.0.0
DNS2IP:0.0.0.0
UDPTOS:0xA0
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RingOnOffTime:2,4,25
DialTone:2,31538,30831,1380,1740,1,0,0,1500
BusyTone:2,30467,28959,1191,1513,0,4000,4000,0
ReorderTone:2,30467,28959,1191,1513,0,2000,2000,0
RingBackTone:2,30831,30467,1943,2111,0,16000,32000,0
CallWaitTone:1,30831,0,5493,0,0,2400,2400,4800
AlertTone:1,30467,0,5970,0,0,480,480,1920
DialPlan:*St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.IPDialPlan: 1
CallCmd:Af;AH;BS;NA;CS;NA;Df;EB;Ff;EP;Kf;EFh;HQ;Jf;AFh;HQ;I*67;gA*82;fA#90v#;OI;H#72v#;bA#
74v#;cA#75v#;dA#73;eA*67;gA*82;fA*70;iA*69;DA*99;xA;Uh;GQ;
SIPPort:5060
SIPRegOn:0
SIPRegInterval:120
MaxRedirect:5
OutBoundProxy:0
NatServer:0
NatTimer:0
NPrintf:0
TraceFlags:0x00000000
EchoIP:192.168.2.9
SigTimer:0x01418564
OpFlags:0x2
VLANSetting:0x0000002b
FeatureTimer:0x00000000
The sections that follow describe these parameters.
User Interface (UI) Parameter
This parameter is located in the purple portion of the Cisco ATA Web Configuration Page.
UIPassword
Description
This parameter controls access to web page or voice configuration menu interface. To set a password,
enter a value other than zero. To have the user prompted for this password when attempting to perform
a factory reset or upgrade using the voice configuration menu, see the “OpFlags” section on page 5-34.
To clear a password, change the value to 0.
You cannot recover a forgotten password unless you reset the entire configuration of the Cisco ATA (see
the “Resetting the Cisco ATA to Factory Default Values” section on page 3-18). If you forget a password,
you can contact your Cisco representative.
Note
When UIPassword contains letters, you cannot enter the password from the telephone keypad.
Value Type
Alphanumeric string
Range
Maximum nine characters
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Configuration Parameter
Default
0
Voice Configuration Menu Access Code
7387277
Configuration Parameter
This parameter is located in the light-yellow portion of the Cisco ATA Web Configuration Page.
ToConfig
Description
After you configure the Cisco ATA, set the parameter to 0, or the Cisco ATA will unnecessarily contact
the TFTP server.
Value Type
Boolean
Range
0 or 1
Default
1—This indicates that the operating parameters have not previously been set.
Voice Configuration Menu Access Code
80001
Parameters for Configuration Method
This section describes the following parameters, which are located in the grey portion of the Cisco ATA
Web Configuration Page:
•
UseTFTP, page 5-4
•
TftpURL, page 5-5
•
CfgInterval, page 5-5
•
EncryptKey, page 5-6
UseTFTP
Settings
1—Use the TFTP server for Cisco ATA configuration.
0—Do not use the TFTP server for Cisco ATA configuration.
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Value Type
Boolean
Range
0 or 1
Default
1
Voice Configuration Menu Access Code
305
TftpURL
Description
Use this parameter to specify the IP address or URL of the TFTP server. This string is needed if the DHCP
server does not provide the TFTP server IP address. When the TftpURL parameter is set to a non-zero value,
this parameter has priority over the TFTP server IP address supplied by the DHCP server.
Optionally, you can include the path prefix to the TFTP file to download.
For example, if the TFTP server IP address is 192.168.2.170 or www.cisco.com, and the path to
download the TFTP file is in /ata186, you can specify the URL as 192.168.2.170/ata186 or
www.cisco.com/ata186.
Note
From the voice configuration menu, you can only enter the IP address; from the web server, you can enter
the actual URL.
Value Type
Alphanumeric string
Range
Maximum number of characters: 31
Default
0
Voice Configuration Menu Access Code
905
CfgInterval
Description
Use this parameter to specify the number of seconds between each configuration update. The Cisco ATA
will also upgrade its signaling image if it detects that the TFTP server contains an upgraded image.
For example, when using TFTP for configuration, the Cisco ATA contacts TFTP each time the interval
expires to get its configuration file.
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Network Parameters
You can set CfgInterval to a random value to achieve random contact intervals from the Cisco ATA to
the TFTP server.
Value Type
Decimal
Range
60 to 4294967295
Default
3600
Voice Configuration Menu Access Code
80002
EncryptKey
Description
This parameter specifies the encryption key that is used to encrypt the Cisco ATA configuration file on
the TFTP server.
The cfgfmt tool, which is used to create a Cisco ATA binary configuration file (see the “Using the
EncryptKey Parameter and cfgfmt Tool” section on page 3-11), automatically encrypts the binary file,
using the rc4 encryption algorithm, when the EncryptKey parameter has a value other than 0.
Note
If the Cisco ATA configuration file is not encrypted, the value must be set to 0.
Value Type
Alphanumeric string
Range
Maximum number of characters: 8
Default
0
Voice Configuration Menu Access Code
320
Network Parameters
This section describes the following parameters, which are located in the orange portion of the
Cisco ATA Web Configuration Page:
•
DHCP, page 5-7
•
StaticIp, page 5-7
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Network Parameters
•
StaticRoute, page 5-7
•
StaticNetMask, page 5-8
DHCP
Description
A DHCP server can be used to automatically set the Cisco ATA IP address, the network route IP address,
the subnet mask, DNS, NTP, TFTP, and other parameters.
•
1—Enable DHCP
•
0—Disable DHCP
Value Type
Boolean
Range
0 or 1
Default
1
Voice Configuration Menu Access Code
20
StaticIp
Description
Use this parameter to statically assign the Cisco ATA IP address if the DHCP parameter is set to 0.
Value Type
IP address
Default
0.0.0.0
Voice Configuration Menu Access Code
1
StaticRoute
Description
Use this parameter to statically assign the Cisco ATA route if the DHCP parameter is set to 0.
Value Type
IP address
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Account Information Parameters
Default
0.0.0.0
Voice Configuration Menu Access Code
2
StaticNetMask
Description
Use this parameter to statically assign the Cisco ATA subnet mask if the DHCP parameter is set to 0
Value Type
IP address
Default
255.255.255.0
Voice Configuration Menu Access Code
10
Account Information Parameters
This section describes the following parameters, which are located in the blue portion of the Cisco ATA
Web Configuration Page:
•
UID0, page 5-9
•
PWD0, page 5-9
•
UID1, page 5-9
•
PWD1, page 5-10
•
GkOrProxy, page 5-10
•
Gateway, page 5-11
•
Gateway2, page 5-11
•
UseLoginID, page 5-11
•
LoginID0, page 5-12
•
LoginID1, page 5-12
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UID0
Description
This parameter is the User ID for the Phone 1 port. If the value is set to zero, the port will be disabled
and no dial tone will sound.
Value Type
Alphanumeric string
Range
Maximum number of characters: 31
Default
0
Voice Configuration Menu Access Code
3
PWD0
Description
This parameter is the password for the Phone 1 port.
Value Type
Alphanumeric string
Range
Maximum number of characters: 31
Default
0
Voice Configuration Menu Access Code
4
UID1
Description
This parameter is the User ID for the Phone 2 port. If the value is set to zero, the port will be disabled
and no dial tone will sound.
Value Type
Alphanumeric string
Range
Maximum number of characters: 31
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Default
0
Voice Configuration Menu Access Code
13
PWD1
Description
This parameter is the password for the Phone 2 port.
Value Type
Alphanumeric string
Range
Maximum number of characters: 31
Default
0
Voice Configuration Menu Access Code
14
GkOrProxy
Description
This parameter is the proxy address or registrar address.
For a SIP proxy server, this can be an IP address with or without a port parameter such as
123.123.110.45, 123.123.110.45.5060, or 123.123.110.45:5061, or a URL such as sip.cisco.com, or
sip.ata.cisco.com:5061. For an IP address, a '.' or ':' can be used to delimit a port parameter. For a URL,
a ':' must be used to indicate a port.
Note
If the SIP proxy server and registration server reside on separate hardware, enter the SIP registration
server address in this field.
If the hostname specified in GkOrProxy has a prefix of _sip._udp or sip.udp, the Cisco ATA first
attempts to perform a DNS SRV lookup on the hostname.
If the SRV lookup returns two hosts, they become primary and backup proxies according to their priority
(as specified in the DNS SRV RFC), and the hostname specified in the AltGk parameter is ignored.
If the SRV lookup returns only one host, this host is the primary proxy, and the hostname specified in
the AltGk parameter is the backup proxy.
Value Type
Alphanumeric string
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Range
Maximum number of characters: 31
Default
0—Disables proxy registration and proxy-routed calls.
In this case, you can make direct IP calls by dialing the user-id@IP:port of the callee, where user-id must
be a numeric value, '@' is dialed as "**", and '.' and ':' are dialed as a '*'.
The following list shows some examples of direct SIP IP dialing:
•
1234**192*168*1*10*5060
•
102*210*9*101*5061
•
4084281002**100*123*89*10
Voice Configuration Menu Access Code
5
Gateway
Description
Not applicable to SIP.
Gateway2
Description
Not applicable to SIP.
UseLoginID
Description
0—Use UID0 and UID1 as the authentication ID.
1—Use LoginID0 and LoginID1 as the authentication ID.
Value Type
Boolean
Range
0 or 1
Default
0
Voice Configuration Menu Access Code
93
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LoginID0
Description
This parameter is the Login ID for line 0.
Note
UID0 is used for authentication if UseLoginID is 0.
Value Type
Alphanumeric string
Range
Maximum number of characters: 51
Default
0
Voice Configuration Menu Access Code
46
LoginID1
Description
This parameter is the Login ID for line 1.
Note
UID1 is used for authentication if UseLoginID is 0.
Value Type
Alphanumeric string
Range
Maximum number of characters: 51
Default
0
Voice Configuration Menu Access Code
47
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Backup Server Parameters
Backup Server Parameters
This section describes the following parameters, which are located in the lavender portion of the
Cisco ATA Web Configuration Page:
•
AltGk, page 5-13
•
AltGkTimeOut, page 5-13
•
GkTimeToLive, page 5-14
•
GkId, page 5-14
AltGk
Description
You have the option of using this parameter to specify a backup proxy. However, if a DNS SRV
performed on the GkOrProxy parameter returns more than one host, the AltGk parameter is ignored.
Value Type
Alphanumeric string
Range
Maximum number of characters: 31
Default
0
Voice Configuration Menu Access Code
6
AltGkTimeOut
Description
You can use this parameter to specify the timeout in seconds before the Cisco ATA fails back to the
primary proxy server from the backup proxy server. Re-registration does not occur until the current
registration period expires.
Value Type
Integer
Default
0—The Cisco ATA continues to use the backup proxy server until it fails before attempting to fail back
to the primary proxy server.
Range
30 to 4294967295 seconds
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SIP Parameters
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GkTimeToLive
Description
Not applicable to SIP.
GkId
Description
Not applicable to SIP.
SIP Parameters
This section describes the following parameters, which are located in the yellow portion of the
Cisco ATA Web Configuration Page:
•
UseSIP, page 5-14
•
SIPRegInterval, page 5-15
•
MAXRedirect, page 5-15
•
SIPRegOn, page 5-16
•
NATIP, page 5-16
•
SIPPort, page 5-17
•
MediaPort, page 5-17
•
OutBoundProxy, page 5-17
•
NatServer, page 5-18
•
NatTimer, page 5-19
UseSIP
Description
0—Use H.323 mode.
1—Use SIP mode.
Value Type
Boolean
Range
0 or 1
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Default
0
Voice Configuration Menu Access Code
38
SIPRegInterval
Description
Use this parameter to configure the number of seconds between Cisco ATA registration renewal with the
SIP proxy server. The Cisco ATA renews the registration at some percentage of time earlier than the
specified interval to prevent a registration from expiring.
Value Type
Integer
Range
1 to 86400
Default
3600
Voice Configuration Menu Access Code
203
MAXRedirect
Description
This parameter specifies the maximum number of times that a called number is allowed to forward the
call to another number.
Value Type
Integer
Range
0 to 10
Default
5
Voice Configuration Menu Access Code
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SIP Parameters
SIPRegOn
Description
0—Disable SIP registration.
1—Enable SIP registration. When this flag is enabled, the Cisco ATA registers with the SIP Proxy Server
that is specified in the GkorProxy parameter. The Cisco ATA also registers with the interval that is
specified in the SIPRegInterval parameter.
Value Type
Boolean
Range
0 or 1
Default
0
Voice Configuration Menu Access Code
204
NATIP
Description
This is the WAN address of the attached router/NAT; currently only used to support SIP behind a NAT.
Value Type
IP address
Default
0.0.0.0
Voice Configuration Menu Access Code
200
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SIP Parameters
SIPPort
Description
This parameter is used to configure the port through which the Cisco ATA listens for incoming SIP
requests and sends outgoing SIP requests.
Value Type
Integer
Range
1 to 65535
Default
5060
Voice Configuration Menu Access Code
201
MediaPort
Description
Use this parameter to specify the base port where the Cisco ATA transmits and receives RTP media. This
parameter must be an even number. Each connection uses the next available even-numbered port for RTP.
Value Type
Integer
Range
1 to 65535
Default
16384
Voice Configuration Menu Access Code
202
OutBoundProxy
Description
The SIP Outbound Proxy Server is a SIP proxy server which can be different from the Registration Proxy
Server (specified in the GkOrProxy parameter) and to which all outgoing SIP requests are sent. Outgoing
SIP responses are not affected by this out-bound-proxy and are still sent according to the VIA header
and source address of the incoming SIP requests.
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SIP Parameters
If the outgoing SIP request has a ROUTE header, the first route in the header is removed if it resolves to
the same IP address as the out-bound-proxy. This process guards against the case when the
out-bound-proxy also inserts its IP address into the RECORD-ROUTE header.
The OutBoundProxy parameter can be an IP address with or without a port parameter, such as
123.123.110.45, 123.123.110.45.5060, or 123.123.110.45:5061, or a URL such as sip.cisco.com,
sip.ata.cisco.com:5061. For IP addresses, a period (.) or colon (:) can be used to delimit a port parameter.
For a URL, a colon (:) must be used to indicate a port. If no port parameter is specified, the port 5060 is
assumed.
Value Type
Alphanumeric string
Range
Maximum number of characters: 31
Default
0
Voice Configuration Menu Access Code
206
NatServer
Description
This parameter allows you to specify a server to which a dummy, single-byte UDP packet is sent to
maintain a Network Address Translation (NAT) during a session.
NatServer can contain up to 47 characters in fully qualified domain name (FQDN) or IP format with an
optional port parameter (separated from the address by a colon); for example, xyz.cisco.com:1234. If no
port is specified, the default port of 5060 is assumed.
Value Type
IP address or FQDN format
Range
Maximum number of characters: 47
Default
5060 is the default port if no port is specified.
Voice Configuration Menu Access Code
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Operating Parameters
NatTimer
Description
This parameter allows you to specify a retransmission interval for sending a dummy packet to NatServer.
The interval is in seconds and is specified in bits 0-11 of this parameter. The upper 20 bits are reserved
and should be set to 0.
Value Type
Bitmap
Default
0, which means that no dummy packets will be sent to the NatServer.
Voice Configuration Menu Access Code
208
Operating Parameters
This section describes the following parameters, which are located in the green portion of the Cisco ATA
Web Configuration Page:
•
LBRCodec, page 5-20
•
AudioMode, page 5-20
•
RxCodec, page 5-21
•
TxCodec, page 5-22
•
NumTxFrames, page 5-22
•
CallFeatures, page 5-23
•
PaidFeatures, page 5-24
•
CallerIdMethod, page 5-25
•
FeatureTimer, page 5-26
•
Polarity, page 5-27
•
ConnectMode, page 5-28
•
AutMethod, page 5-30
•
TimeZone, page 5-30
•
NTPIP, page 5-30
•
AltNTPIP, page 5-31
•
DNS1IP, page 5-31
•
DNS2IP, page 5-31
•
UDPTOS, page 5-32
•
SigTimer, page 5-32
•
OpFlags, page 5-34
•
VLANSetting, page 5-35
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Operating Parameters
LBRCodec
Description
This parameter allows you to specifiy which low-bit-rate codecs are available. The Cisco ATA is capable
of supporting two G.723.1 connections or one G.729 connection. When G.723.1 is selected as the low-bit-rate
codec, each FXS port is allocated with one G.723.1 connection. When G.729 is selected, only one FXS port
is capable of operating with the G.729 codec. The allocation of the G.729 resource to the FXS port is dynamic.
The G.729 resource, if available, is allocated to an FXS port when a call is initiated or received; the resource
is released when a call is completed.
The following values are valid:
•
0—Select G.723.1 as the low-bit-rate codec.
•
3—Select either G.729 as the low-bit-rate codec.
Related Parameters
•
RxCodec, page 5-21
•
TxCodec, page 5-22
Value Type
Integer
Range
0 or 3
Default
0
Voice Configuration Menu Access Code
300
AudioMode
Description
This parameter represents the audio operating mode. The lower 16 bits are for the Phone 1 port, and the
upper 16 bits are for the Phone 2 port. Table 5-1 on page 5-21 provides definitions for each bit.
Value Type
Bitmap
Default
0x00150015
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Table 5-1
AudioMode Parameter Bit Definitions
Bit Number
Definition
0
0/1—Disable/enable G.711 silence suppression.
1
0—Enable selected low-bit-rate codec in addition to G.711. This setting is the default.
1—Enable G.711 only.
2
0/1—Disable/enable fax CED tone detection.
3
0/1—Enable/disable fax CNG tone detection.
4-5: DtmfMethod
0—Always in-band (send and receive, do not send SDP info)
1—By negotiation (send SDP info, enable receive, decode others’ SDP information, send
depends on others’ SDP information)
2—Always out-of-band (send SDP info, enable receive, decode others’ SDP information, always
send).
3—Reserved.
6-15
Reserved.
RxCodec
Description
Use this parameter to specify receiving-audio codec preference. The following values are valid:
•
0—G.723 (can be selected only if LBRCodec is set to 0)
•
1—G.711A-law
•
2—G.711µ-law
•
3—G.729a (can be selected only if LBRCodec is set to 3)
Value Type
Integer
Range
0-3
Default
2
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TxCodec
Description
Use this parameter to specify the transmitting-audio codec preference. The following values are valid:
•
0—G.723 (can be selected only if LBRCodec is set to 0)
•
1—G.711A-law
•
2—G.711µ-law
•
3—G.729A (can be selected only if LBRCodec is set to 3)
Value Type
Integer
Range
0-3
Default
2
Voice Configuration Menu Access Code
37
NumTxFrames
Description
Use this parameter to select the default RTP packet side in number of frames per packet. The Cisco ATA
default frame sizes are as follows:
•
G.711 and G.729—10 ms
•
G.723.1—30 ms
For example, to receive 20 ms of G.729 packets, set the parameter to 2.
Value Type
Integer
Range
1-6
Default
2
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Operating Parameters
CallFeatures
Description
Disable/enable CallFeatures by setting each corresponding bit to 0 or 1.
The lower 16 bits are for the Phone 1 port, and the upper 16 bits are for the Phone 2 port. Table 5-2
provides definitions of each bit.
Note
The subscribed features that can be permanently disabled by the user are CLIP_CLIR, call waiting and
Fax mode. A subscribed service enable/disabled by the user can be disabled/enabled dynamically on a
per-call basis.
Value Type
Bitmap
Default
0xffffffff
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Table 5-2
CallFeatures Parameter Bit Definitons
Bit
Number
Definition
0
Forward unconditionally
1
Forward on busy
2
Forward on no answer
3
CLIP_CLIR
4
Call waiting
5
three-way calling
6
Blind transfer
7
Transfer with consultation. This service allows the user to transfer the remote party to a different number by first
calling that number and consulting with the callee.
8
Caller ID. This service enables the Cisco ATA 186 to generate a Caller ID signal to drive a Caller ID display
device attached to the FXS line.
9
Call return
10
Message waiting indication
11
Call Waiting Caller ID. This is available only if the Method bit in CallerIdMethod is set to Bellcore (FSK).
15
Fax mode. This service allows the user to set the Cisco ATA to Fax mode on a per-call basis. For Fax mode, use
the following settings:
•
G711 codec only
•
No silence suppression
•
No FAX tone detection
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PaidFeatures
Description
Unsubscribe/subscribe to CallFeatures by setting each corresponding bit to either 0 or 1. The lower 16
bits are for the Phone 1 port, and the upper 16 bits are for the Phone 2 port. Table 5-3 provides
definitions of each bit.
Value Type
Bitmap
Default
0xffffffff
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Table 5-3
PaidFeatures Parameter Bit Definitions
Bit Number
Definition
0
Forward unconditionally
1
Forward on busy
2
Forward on no answer
3
CLIP_CLIR
4
Call waiting
5
three-way calling
6
Blind transfer
7
Transfer with consultation. This service allows the user to transfer the remote party to a different number by
first calling that number and consulting with the callee.
8
Caller ID. This service enables the Cisco ATA 186 to generate a Caller ID signal to drive a Caller ID display
device attached to the FXS line.
9
Call return
10
Message waiting indication
11
Call Waiting Caller ID. This is available only if the Method bit in CallerIdMethod is set to Bellcore (FSK).
15
Fax mode. This service allows the user to set the Cisco ATA to Fax mode on a per-call basis. For Fax mode, use
the following settings:
•
G.711 codec only
•
No silence suppression
•
No FAX tone detection
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CallerIdMethod
Description
This 32-bit parameter specifies the signal format to use for both FXS ports for generating Caller ID
format. Possible values are:
•
Bits 0-1 (method)—0=Bellcore (FSK), 1=DTMF, values 2 and 3 are reserved.
If method=0, set the following bits:
•
Bit 2—Reserved.
•
Bit 3 to 8—Maximum number of digits in phone number (valid values are 1 to 20; default is 12)
•
Bit 9 to 14—Maximum number of characters in name (valid values are 1 to 20; default is 15)
•
Bit 15—If this bit is enabled (it is by default), send special character O (out of area) to CID device
if the phone number is unknown.
•
Bit 16—If this bit is enabled (it is by default), send special character P (private) to CID device if the
phone number is restricted.
•
Bits 17 to 27—Reserved.
If method=1, set the following bits:
•
Bits 3-6—Start digit for known numbers (valid values are 12 for “A,” 13 for “B,” 14 for “C,” and
15 for “D.”)
•
Bits 7-10—End digit for known numbers (valid values are 11 for “#,” 12 for “A,” 13 for “B,” 14 for
“C,” and 15 for “D.”)
•
Bits 11—Polarity reversal before and after Caller ID signal (value of 0/1 disables/enables polarity
reversal)
•
Bits 12-16—Maximum number of digits in phone number (valid values are 1 to 20)
•
Bits 17 to 19—Start digit for unknown or restricted numbers (valid values are 4 for “A,” 5 for “B,”
6 for “C,” and 7 for “D.”)
•
Bits 20 to 22—End digit for unknown or restricted numbers (valid values are 3 for “#,” 4 for “A,” 5
for “B,” 6 for “C,” and 7 for “D.”)
•
Bits 23 to 24—Code to send to the CID device if the number is unknown (valid values are 0 for “00,”
1 for “0000000000,” and 2 for “2.” 3 is reserved and should not be used.
•
Bits 25 to 26—Code to send to the CID device if the number is restricted (valid values are 0 for “10,”
and 1 for “1.” 2 and 3 are reserved and should not be used.
•
Bits 27 to 31—Reserved.
Examples
The following examples are recommended values for the CallerID Method parameter:
•
USA=0x19e60
•
Sweden=0x0ff61 or 0x006aff61
•
Denmark=0x0fde1 or 0x033efde1
Value Type
Bitmap
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Default
0x00019e60
Voice Configuration Menu Access Code
316
FeatureTimer
Description
This parameter provides configurable timing values for various features, as shown below.
•
Bits 0-3—Maximum time to spend redialing if line is busy
– Range: 0 - 15
– Factor: five-minute increments
– Values: 0 - 75 minutes
– Default: 0 (= 30 minutes)
•
Bits 4-7—Retry interval if line is busy
– Range: 0 - 15
– Factor: 15-second increments
– Values: 0 - 225 seconds
– Default: 0 (= 15 seconds)
•
Bits 8-12—On-hook delay before a call is disconnected. This feature works only when the
Cisco ATA is the terminating endpoint of the call. The user can hang up the phone in one room and
pick up the phone in another room without disconnecting the line.
– Range: 0 - 31
– Factor: five-second increments
– Values: 0 - 155 seconds
– Default: 0 (no delay)
•
Bits 13-15—Amount of time the Cisco ATA waits for a "486 Busy Here" response from a PSTN
gateway after receiving a "183 Session Progress" response.
– Range: 0 - 7
– Factor: one-second increments
– Values: 0 to 7 seconds
– Default: 0 (no waiting)
•
Bits 16-31—Reserved.
Value Type
Bitmap
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Default
0x00000000
Voice Configuration Menu Access Code
317
Polarity
Description
You can control line polarity of the Cisco ATA FXS ports when a call is connected or disconnected by
configuring the Polarity bitmap parameter as follows:
•
Bit 0: CALLER_CONNECT_POLARITY. Polarity to use when the Cisco ATA is the caller and the
call is connected.
– 0 =Use forward polarity (Default)
– 1 =Use reverse polarity
•
Bit 1: CALLER_DISCONNECT_POLARITY. Polarity to use when the Cisco ATA is the caller and
the call is disconnected.
– 0 =Use forward polarity (Default)
– 1 =Use reverse polarity
•
Bit 2: CALLEE_CONNECT_POLARITY. Polarity to use when the Cisco ATA is the callee and the
call is connected.
– 0 =Use forward polarity (Default)
– 1 =Use reverse polarity
•
Bit 3: CALLEE_DISCONNECT_POLARITY. Polarity to use when the Cisco ATA is the callee and
the call is disconnected.
– 0 =Use forward polarity (Default)
– 1 =Use reverse polarity
Note
Bits 4-31 are reserved.
Value Type
Bitmap
Default
0x00000000
Voice Configuration Menu Access Code
304
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ConnectMode
Description
This parameter is a 32-bit bitmap used to control the connection mode of the selected call signaling
protocol. Table 5-4 on page 5-28 provides bit definitions for this parameter.
Value Type
Bitmap
Default
0x00060400
Voice Configuration Menu Access Code
311
Table 5-4
ConnectMode Parameter Bit Definitions
Bit Number
Definition
0—H.323 only
0—Enable normal start.
1—Enable fast start.
1—H.323 only
0/1—Disable/enable h245 tunneling.
2
0—Use the dynamic payload type 126/127 as the RTP payload type (fax pass-through mode) for G.711
µ-law/G.711 A-law.
1—Use the standard payload type 0/8 as the RTP payload type (fax pass-through mode) for G.711
µ-law/G.711 A-law.
3—H.323 only
0/1—Disable/enable the requirement for the alternate gatekeeper to register.
4—H.323 only
0—Denotes a non-Cisco CallManager environment.
1—Enable the Cisco ATA to operate in a Cisco CallManager environment.
5—H.323 only
0/1—Enable/disable two-way cut-through of voice path before receiving CONNECT message.
6—H.323 only
0/1—Disable/enable using the Progress Indicator to determine if ringback is supplied by the far end with
RTP.
7
0/1—Disable/enable fax pass-through redundancy.
8-12
Specifies the fax pass-through NSE payload type. The value is the offset to the NSE payload base number
of 96. The valid range is 0-23; the default is 4.
For example, if the offset is 4, the NSE payload type is 100.
13
0—Use G.711µ-law for fax pass-through codec.
1—Use G.711A-law for fax pass-through codec.
14-15
0—Use fax pass-through.
1—Use codec negotiation in sending fax.
2—Reserved.
3—Reserved.
16—SIP only
0/1—Disable/enable SIP to remove the registration before adding a new one.
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Table 5-4
ConnectMode Parameter Bit Definitions (continued)
Bit Number
Definition
17—SIP only
0/1—Disable/enable call forwarding performed by the Cisco ATA. In SIP, call forwarding can be
performed locally by the Cisco ATA or it can be performed by the SIP proxy. If this bit is disabled, the
Cisco ATA forwards the entire dial string, including service activation code, to the SIP proxy for
processing.
18—SIP only
0/1—Disable/enable SIP call return performed by the Cisco ATA.
19
0/1—Disable/enable the Cisco ATA to send a ringback tone to the caller.
20—SIP only
0/1—Disable/enable SIP to perform action=proxy in a REGISTER message.
21—SIP only
0/1—Disable/enable SIP to perform action=redirect in a REGISTER message.
22—SIP only
0/1—Disable/enable SIP to process a received= tag in the VIA header to extract the exernal IP addresses
used by the Network Address Translation (NAT) router.
When the Cisco ATA is operating behind a NAT, the NATIP parameter must be set to the external IP
address of the NAT router. This allows the correct IP address to be placed in the Contact and SDP
headers.
In release 2.14 or later, you may leave the NAT IP address at the default value of "0" or "0.0.0.0" and let
the ATA automatically scan the Via header for a "received=" parameter. The parameter, if present, would
indicate that the Cisco ATA is operating behind a firewall.
The Cisco ATA proceeds as follows:
1.
If the "received=" parameter is in an INVITE response, the current INVITE is canceled and a new
INVITE is sent with the new IP address extracted from the "received=" parameter
in the Contact and SDP headers.
In addition, the Cisco ATA will cancel all previous registrations and re-register with the new IP
address in the Contact header. This step is performed only if registration is currently in an idle state.
2.
If the "received=" parameter is in a REGISTER response as a result of a REGISTER command, the
Cisco ATA will cancel all previous registrations and re-register with the new IP address extracted
from the "received=" parameter in the Contact header.
Note
For the Cisco ATA to automatically detect its presence behind a NAT, the SIP proxy server
or remote user agent server must include the "received=" parameter in the Via header in the
responses to the Cisco ATA if the proxy detects that the source address and port do not match
those in the Via header.
23
0/1—Disable/enable the end user to specify the default call waiting state for every call. This can be done
on a permanent basis.
24
0/1—Disable/enable the mixing of audio and call waiting tone during a call.
25—SIP only
0/1—Disable/enable DNS lookup of hostname “From” header hostname for call return.
26 to 31
Reserved.
Note
You cannot simultaneously set bits 20 and 21 to 1. Also, if you set both these bits 0, the action parameter
is not included in the REGISTER message, forcing the proxy server to perform the next step in the call
process.
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AutMethod
Description
This parameter is not used for SIP.
TimeZone
Description
This parameter is the timezone offset from Greenwich Mean Time (GMT) for time-stamping incoming
calls with local time (to use for Caller ID display, for example).
Local time is generated by the following formula:
•
Local Time=GMT + TimeZone, if TimeZone <= 12
•
Local Time=GMT + TimeZone - 25, if TimeZone > 12
Value Type
Integer
Range
0-24
Default
17
Voice Configuration Menu Access Code
302
NTPIP
Description
This parameter is the NTP IP address, required if DHCP server does not provide one.
The Cisco ATA requires an NTP Server from which to obtain Coordinated Universal Time (UTC) to
time-stamp incoming calls (H.323 and SIP) to drive an external Caller-ID device.
DHCP may also supply a NTP server. If NTPIP is specified, it overwrites the value supplied by DHCP.
NTPIP is ignored if its value is 0 or 0.0.0.0.
The user must not specify a port parameter. The Cisco ATA uses the default NTP port only.
Value Type
IP address
Default
0.0.0.0
Voice Configuration Menu Access Code
141
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AltNTPIP
Description
This parameter is the alternate NTP IP address, if you want redundancy. You can set this parameter to 0
or point to the same NTPIP if only one NTP server exists.
Value Type
IP address
Default
0.0.0.0
Voice Configuration Menu Access Code
142
DNS1IP
Description
This parameter is the primary domain name server (DNS) IP address, if the DHCP server does not
provide one. If DHCP provides DNS, DNS1IP and DNS2IP (if they are non-zero) overwrite the
DHCP-supplied values. The user must not specify a port parameter. The Cisco ATA uses the default DNS
port only.
Value Type
IP address
Default
0.0.0.0
Voice Configuration Menu Access Code
916
DNS2IP
Description
This parameter is the secondary domain name server (DNS) IP address, if the DHCP server does not
provide one. If DHCP provides DNS, DNS1IP and DNS2IP (if they are non-zero) overwrite the
DHCP-supplied values. The user must not specify a port parameter. The Cisco ATA uses the default DNS
port only.
Value Type
IP address
Default
0.0.0.0
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Voice Configuration Menu Access Code
917
UDPTOS
Description
This parameter specifies the IP precedence (ToS bit) of UDP packets. Set the lower eight bits only, as
follows:
•
Bits 0-1: Unused
•
Bit 2: Reliability bit—1=request high reliability
•
Bit 3: Throughput bit—1=request high throughput
•
Bit 4: Delay bit—1=request low delay
•
Bits 5-7: Specify datagram precedence. Values range from 0 (normal precedence) to 7 (network
control).
Value Type
Bitmap
Default
0xB8
Voice Configuration Menu Access Code
255
SigTimer
Description
This parameter controls various timeout values. Table 5-5 on page 5-33 contains bit definitions of this
parameter.
Value Type
Bitmap
Default
0x01418564
Voice Configuration Menu Access Code
318
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Table 5-5
SigTimer Parameter Bit Definitions
Bit Number
Definition
0-7
Call waiting period—The period between each burst of call-waiting tone.
Range: 0 to 255 in 0.1 seconds
Default: 100 (0x64=100 seconds)
8-13
Reorder delay—The delay in playing the reorder (fast busy) tone after the far-end caller hangs up.
Range: 0 to 62 in seconds
Default—5 (seconds)
63—Never play the reorder tone.
14-19
Ring timeout—When a call is not answered, this is the amount of time after which Cisco ATA rejects the
incoming call.
Range—0 to 63 in 10 seconds
Default—6 (60 seconds)
0—Never times out
20-25
No-answer timeout—The time to declare no answer and initiate call forwarding on no answer (used in SIP
only
Range—0 to 63 in seconds
Default—20 (0x14=20 seconds)
26-27
Minimum hook flash time—The minimum on-hook time required for hook flash event.
Range: 0 to 3
Default: 0 (60 ms)
Other possible values: 1=100 ms, 2=200 ms, 3=300 ms.
28-31
Maximum hook flash time—The maximum on-hook time allowed for hook flash event.
Range: 0 to 15
Default: 0 (1000 ms)
Other possible values: 1=100 ms, 2=200 ms, 3=300 ms, 4=400 ms, 5=500 ms, 6=600 ms, 7=700 ms, 8=800
ms, 9=900 ms, 10=1000 ms, 11=1100 ms, 12=1200 ms, 13=1300 ms, 14=1400 ms, 15=1500 ms.
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OpFlags
Description
This parameter enables/disables various operational features.
See Table 5-6 on page 5-34 for bit definitions of this parameter.
Value Type
Bitmap
Default
0x2
Voice Configuration Menu Access Code
323
Table 5-6
OpFlags Parameter Operational Features to Turn On or Off
Bit Number
Definition
0
If Bit 0 = 0, the TFTP configuration filename supplied by the DHCP server overwrites the default filename for
each Cisco ATA.
If Bit 0 = 1, the default Cisco ATA filename is always used.
1
If Bit 1 = 0, the Cisco ATA probes the static network router during the power-up process.
If Bit 1 = 1, static network router probing is disabled.
2
Reserved.
3
If Bit 3=1, the Cisco ATA does not request DHCP option 150 in the DHCP discovery message; some DHCP
server do not respond if option 150 is requested.
4
If Bit 4 = 1, the Cisco ATA uses the VLAN ID specified in the VLANSetting parameter for VLAN IP encapsulation
(see the “VLANSetting” section on page 5-35).
5
If Bit 5=1, the Cisco ATA does not use VLAN IP encapsulation.
6
If Bit 6=1, the Cisco ATA does not perform CDP discovery.
7
If Bit 7=1, the Cisco ATA does not allow web configuration. Once the web server is disabled, you must
configure the Cisco ATA with the TFTP or voice configuration menu methods.
Examples
1.
If the existing OpFlags value is 0x2, select menu option 323 from the voice configuration menu and enter
the value 130 (0x82). This disables web configuration.
If you later attempt to access the Cisco ATA web configuration page, the following error messages will be
displayed.
– Netscape: The document contained no data. Try again later, or contact the
server's administrator.
– Internet Explorer:
2.
The page cannot be displayed.
If the existing OpFlags value is 0x82, select menu option 323 from the voice configuration menu and enter
the value 2 (0x2). This disables web configuration.
8
If Bit 8=1, the Cisco ATA does not allow HTTP refresh access with the http://ip/refresh command.
9
If Bit 9=1, the Cisco ATA does not allow HTTP reset access with the http://ip/reset command.
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Table 5-6
OpFlags Parameter Operational Features to Turn On or Off (continued)
Bit Number
Definition
10
Reserved.
11
If Bit 11=0,the Cisco ATA requests the device hostname from the DHCP server.
If Bit 11=1, the Cisco ATA uses the device hostname that is specified in DHCP option 12.
12-27
Reserved.
28-31
To configure the Cisco ATA to prompt the user for the UIPassword when the user attempts to perform a factory
reset or upgrade using the voice configuration menu, configure bits 28 to 31 with the value of 6. Any other value
for these bits means that the Cisco ATA will not prompt the user for the UIPassword in these cases.
VLANSetting
Description
This parameter is for firmware version 2.15 and 2.14ms, and above.
Bitmap definitions are as follows for the VLANSetting parameter:
•
Bits 0-2—Specify VLAN CoS bit value (802.1P priority) for TCP packets.
•
Bits 3-5—Specify VLAN CoS bit value (802.1P priority) for UDP packets.
•
Bits 6-17—Reserved.
•
Bits 18-29—User-specified 802.1Q VLAN ID.
•
Bits 30-31—Reserved.
Value Type
Bitmap
Default
0x0000002b
Voice Configuration Menu Access Code
324
Optional Feature Parameters
This section describes the following parameters, which are located in the wheat-colored portion of the
Cisco ATA Web Configuration Page:
•
NPrintf, page 5-36
•
TraceFlags, page 5-36
•
RingOnOffTime, page 5-37
•
IPDialPlan, page 5-38
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•
DialPlan, page 5-38
•
Call-Progress Tone Parameters, page 5-42
•
CallCmd, page 5-47
NPrintf
Description
Use this parameter to specify the IP address and port of a host to which all Cisco ATA debug messages
are sent. The program prserv.exe, which comes bundled with the Cisco ATA software, is needed to
capture the debug information.
Syntax
,
Example
If the program prserv.exe is running on a host with IP address 192.168.2.170 and listening port 9001, set
NPrintf to 192.168.2.170.9001. This causes the Cisco ATA to send all debug traces to that IP address.
Value Type
Extended IP address
Default
0
Voice Configuration Menu Access Code
81
TraceFlags
Description
Use this parameter to turn on specific trace features for diagnostic use. Bit values are as follows:
•
Bits 0 to 1:
– 0 (default) is for simple debug messages.
– 1 is for detailed debug messages.
– 2 and 3 are reserved.
•
Bits 2-7—Reserved
•
Bit 8—RTP statistics log (values 0/1 to disable/enable with default of 0) has the following format:
Recv[channel number]:
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Bit 8 is not used at this time.
Note
•
Bits 9 to 31—Reserved.
Value Type
Bitmap
Default
0x00000000
Voice Configuration Menu Access Code
313
RingOnOffTime
Description
This parameter specifies the ringer cadence pattern, expressed as a triplet of integers “a,b, and c”.
•
a—Number of seconds to turn the ring ON.
•
b—Number of seconds to turn the ring OFF.
•
c—The ring frequency, fixed at 25.
Value Type
List of three integer values, separated by commas
Range
1-65535
Default
2, 4, 25
Recommended Values:
•
United States —2,4,25
•
Sweden — 1,5,25
Voice Configuration Menu Access Code
929
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IPDialPlan
Description
This Iparameter allows for detection of IP-like destination address in DialPlan. Three values are valid:
•
0—String is dialed as is and not treated as an IP address.
•
1—When the Cisco ATA detects two asterisks (**), IPDialPlan takes over. The user enters the pound
(#) key to terminate the digit collection, and the interdigit timeout default is not used.
•
2—When IPDialPlan is set to 2, three asterisks (***) are required for IPDialPlan to take effect.
All other values are currently undefined.
Value Type
Integer
Range
0, 1 or 2
Default
1
Voice Configuration Menu Access Code
310
DialPlan
Description
The programmable dial plan is designed for the service provider to customize the behavior of the
Cisco ATA for collecting and sending dialed digits. The dial plan allows the Cisco ATA user to specify
the events that trigger the sending of dialed digits. These events include the following:
•
The termination character has been entered.
•
The specified dial string pattern has been accumulated.
•
The specified number of dialed digits has been accumulated.
•
The specified inter-digit timer has expired.
Value Type
Alphanumeric string
Range
Maximum number of characters is 199.
Default
*St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.Voice Configuration Menu Access Code
926
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Additional DialPlan Information
The DialPlan section contains the following additional topics that describe commands and rules for
creating your own dial plan:
•
About Dial Plan Commands, page 5-39
•
Dial Plan Blocking (In Rule), page 5-41
•
'H' Rule to Support Hot/Warm Line, page 5-41
•
'P' Rule to Support Dial Prefix, page 5-42
About Dial Plan Commands
The following list contains rules for Cisco ATA dial plans:
Note
•
. —Wildcard, match any digit entered.
•
- —Additional digits can be entered. This command can be used only at the end of a dial plan rule
(for example, 1408t5- is legal usage of the - command, but 1408t5-3... is illegal).
•
>#—Defines the # character as a termination character. When the termination character is entered,
the dial string is automatically sent. The termination character can be entered only after at least one
user-entered digit matches a dial plan rule. Alternatively, the command >* can be used to define *
as the termination character.
•
tn— Defines the timeout value n, in the unit of seconds, for the interdigit timer. Valid values are 0-9 and
a-z, where a-z indicates a range of 10 to 36.
•
rn—Repeat the last pattern n times, where n is 0-9 or a-z. The values a-z indicate a range of 10 to
36. Use the repeat modifier to specify more rules in less space.
The commands ># and tn are modifiers, not patterns, and are ignored by the rn command.
•
|—Used to separate multiple dial plan rules.
•
^—Logical not. Match any character except the character immediately following the ^ command.
•
S—Seize rule matching. If a dial plan rule matches the sequence of digits entered by the user to this
point, and the modifier S is the next command in the dial plan rule, all other rules are negated for
the remainder of the call (for example, a dial plan beginning with *S will be the only one in effect
if the user first enters the * key).
Note
All rules apply in the order listed (whichever rule is completely matched first will immediately send the
dial string).
Note
No syntax check is performed by the actual implementation. The administrator has the responsibility of
making sure that the dial plan is syntactically valid.
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Dial Plan Example 1 (Default Dial Plan)
The following dial plan:
*St4-|#St4-|911|1>#t8.r9t2-|0>#t811.rat4-|^1t4>#.-
consists of the following rules:
•
*St4-—If the first digit entered is *, all other dial plan rules are voided. Additional digits can be
entered after the initial * digit, and the timeout before automatic dial string send is four seconds.
•
#St4—Same as above, except with # as the initial digit entered.
•
911—If the dial string 911 is entered, send it immediately.
•
1>#t8.r9t2—If the first digit entered is 1, the timeout before automatic send is eight seconds. The
terminating character # can be entered at any time to manually send the dial string. After the 11th
digit is entered, the timeout before an automatic send changes to two seconds. The user can enter
more digits until the dial string is sent by the timeout or by the user entering the # character.
•
0>#t811.rat4—If the first digit entered is 0, the timeout before automatic send is eight seconds, and
the terminating character # can be entered at any time to manually send the dial string. If the first
three digits entered are 011, then, after an additional 11 digits are entered, the timeout before an
automatic send changes to four seconds. The user can enter more digits until the dial string is sent
by the timeout or by the user entering the # character.
•
^1t4>#.—If the first digit entered is anything other than 1, the timeout before an automatic send is
four seconds. The terminating character # can be entered at any time to manually send the dial string.
The user can enter more digits until the dial string is sent by the timeout or by the user entering the
# character.
Dial Plan Example 2
The following dial plans:
.t7>#......t4-|911|1t7>#..........t1-|0t4>#.t7or
.t7>#r6t4-|911|1t7>#.r9t1-|0t4>#.t7-
consist of the following rules:
•
.t7>#r6t4-—You must enter at least one digit. After the first digit is entered and matched by the dial
plan, the timeout before an automatic send is seven seconds, and the terminating character # can be
entered at any time to manually send the dial string. After seven digits are entered, the timeout
before an automatic send changes to two seconds. The - symbol at the end of the rule allows further
digits to be entered until the dial string is sent by the timeout or the user entering the # character.
•
911—If the dial string 911 is entered, send this string immediately.
•
1t7>#.r9t1—If the first digit entered is 1, the timeout before an automatic send is seven seconds,
and the terminating character # can be entered at any time to manually send the dial string. After the
11th digit is entered, the timeout before an automatic send changes to one second. The user can enter
more digits until the dial string is sent by the timeout or by the user entering the # character.
•
0t4>#.t7—If the first digit entered is 0, the timeout before an automatic send is four seconds, and
the terminating character # can be entered at any time to manually send the dial string. After the
second digit is entered, the timeout before an automatic send changes to seven seconds. The user
can enter more digits until the dial string is sent by the timeout or by the user entering the # character.
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Optional Feature Parameters
Dial Plan Blocking (In Rule)
Dial plan blocking can be used to reduce the occurrences of invalid dialed digits being sent and can
prevent the dialed string of a specified pattern from being sent. By adding dial plan blocking, dialed
digits are discarded after the interdigit timer expires unless one of the specified matching rules is met.
In addition, the default nine-second global interdigit timeout value is also modified with the value specified
in the dial plan blocking command:
In
where n specifies the global interdigit timeout and the valid values are 1-9 and a-z (10-35).
For example, to enter an interdigit timeout of 12 seconds and discard dialed digits unless 911 is entered,
you would use the following command:
Ic| 911
Specifying your own interdigit timeout also changes the behavior of the dial plan so that the entire dial
string, rather than being sent at timeout, is sent only as a result of a matching rule or time intended by a
matching rule.
'H' Rule to Support Hot/Warm Line
Hotline/Warmline, also known as Private Line Automatic Ringdown (PLAR), is a line used for priority
telephone service. If the Hotline feature is configured, the Cisco ATA immediately dials a pre-configured
number as soon as the handset goes off hook. If the Warmline feature is configured, the Cisco ATA dials a
pre-configured number if no digits were entered before the specified timer value expired when the handset
went offhook.
Syntax
Hdnnnn
where d is a delay-in-seconds parameter 0-9,a-z (to support 0 to 35 seconds delay), and nnnn is the
variable-length phone number to call when no digits are entered for d seconds after offhook.
•
Example 1: H05551212 (Hotline configuration; the Cisco ATA immediately dials 555-1212 when the
handset goes off hook.)
•
Example 2: H55551212 (Warmline configuration; the Cisco ATA waits for five seconds and dials
555-1212 if no digits were entered when the handset went off hook.)
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'P' Rule to Support Dial Prefix
This rule is for automatic pre-pending the dial string as entered by the user with a specified prefix.
Syntax
Ptnnnn
where t is a single leading trigger character; if t is the first entered digit when making a new call, it
triggers the prepending of a variable-length prefix (as specified by nnnn) in the dial string. The t
character can take one of the following values:
0-9,*,#, 'n' (= any of 1-9), 'N' (any of 'n' and 0), 'a' (any of 'n',* and #), or 'A' (any of 'a' and 0);
Example:
Pn12345: Prepends 12345 to the dial string when the first entered digit is any of 1-9. The triggered digit
is not removed from the dial string.
Call-Progress Tone Parameters
This section contains the following topics:
•
List of Call-Progress Tone Parameters, page 5-42
•
Tone Parameter Syntax, page 5-42
•
How to Calculate Scaling Factors, page 5-43
•
Recommended Values, page 5-44
•
Specific Call-Progress Tone Parameter Information, page 5-44
List of Call-Progress Tone Parameters
The following list contains the names of the call-progress tone parameters:
•
DialTone
•
BusyTone
•
ReorderTone
•
RingBackTone
•
CallWaitTone
•
AlertTone
Tone Parameter Syntax
Each tone is specified by nine intergers, as follows:
ntone, freq0, freq1, level0, level1, steady, on-time, off-time, total-tone-time
•
ntone is the number of frequency components (0, 1 or 2).
•
freq[0] (Hz) is the transformed frequency of the first frequency component (-32768 to 32767).
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Only positive values can be configured to the Cisco ATA 186. For negative values, use the 16-bit
2’s-complement value. For example, enter -1 as 65535 or 0xffff.
Note
•
freq[1] is the transformed frequency of the second frequency component (-32768 to 32767).
•
level[0] is the transformed amplitude of the first frequency component (-32768 to 32767).
•
level[1] is the transformed amplitude of the second frequency component (-32768 to 32767).
•
steady controls whether the tone is constant or intermittent. A value of 1 indicates a steady tone and
causes the Cisco ATA to ignore the on-time and off-time parameters. A value of 0 indicates an on/off
tone pattern and causes the Cisco ATA to use the on-time and off-time parameters.
•
on-time controls the length of time the tone is heard in milliseconds (ms) expressed as an integer
from 0 to 0xffff sample at 8000 samples/second.
•
off-time controls the length of time between audible tones in milliseconds (ms) expressed as an
integer from 0 to 0xffff sample at 8000 samples/second.
•
total-tone-time controls the length of time the tone is audible (0 to 0xffff). If this value is set to 0,
the tone will play until another call event stops the tone. For DialTone, BusyTone, ReorderTone, and
RingBackTone, the configurable value is the number of 10 ms (100 = 1 second) units.
For the other tones, the value is the number of samples at 8000 samples/second, where the following
information applies:
– Frequency ranges from 0 to 4000 (Hz)
– Transformed Frequency = 32767 · cos (2pi·Frequency/8000)
– Amplitude ranges from 0 to 32767
– Transformed Amplitude = A · 32767 · sin (2pi·Frequency/8000)
The scaling factor A determines the volume level of the tone. To calculate scaling factors, see the
“How to Calculate Scaling Factors” section on page 5-43.
Note
All tones are persistent (until the Cisco ATA changes state) except for the call-waiting tone and the
confirm tone. The call-waiting tone, however, repeats automatically once every 10 seconds while the
call-waiting condition exists.
How to Calculate Scaling Factors
Use the following formula to calculate the scaling factor A:
A=0.5 * 10^((k+10-(n-1)*3)/20)
In this formulas, k is the desirable volume in dBm; n is the number of frequency components. The ^
symbol means to the order of.
Example
If a one-frequency component of -20 dBm volume level is desirable, then:
A=0.5 * 10^((-20+10)/20) = 0.16
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Optional Feature Parameters
Recommended Values
The following settings are recommended for the US:
•
DialTone = "2,31538,30831,3100,3885,1,0,0,1000" (approximately -10 dBm)
•
BusyTone = "2,30467,28959,1191,1513,0,4000,4000,0" (approximately -21 dBm)
•
ReorderTone = "2,30467,28959,1191,1513,0,2000,2000,0" (approximately -21 dBm)
•
RingBackTone = "2,30831,30467,1943,2111,0,16000,32000,0" (approximately -16 dBm)
•
CallWaitTone = "1,30831,0,5493,0,0,2400,2400,4800" (approximately -10 dBm)
•
AlertTone = "1,30467,0,5970,0,0,480,480,1920" (approximately -10 dBm)
The following settings are recommended for Sweden:
•
DialTone = "1,30959,0,4253,0, 1, 0, 0,1500" (approximately -5 dBm)
•
BusyTone = "1,30959,0,2392,0, 0, 2000, 2000,0" (approximately -10 dBm)
•
ReorderTone = "1,30959,0,2392,0, 0, 2000, 6000,0" (approximately -10 dBm)
•
RingBackTone = "1,30959,0,2392,0, 0, 8000, 40000,0" (approximately -10 dBm)
•
CallWaitTone = "1,30959,0,2392,0, 0, 1600, 4000,11200" (approximately -10 dBm)
•
AlertTone = "1,30959,0,2392,0, 0, 480, 480,1920" (approximately -10 dBm)
Specific Call-Progress Tone Parameter Information
Brief descriptions, and lists of default values and the voice configuration menu code for each Cisco ATA
tone parameter, are described in the following sections:
•
DialTone, page 5-44
•
BusyTone, page 5-45
•
ReorderTone, page 5-45
•
RingbackTone, page 5-46
•
CallWaitTone, page 5-46
•
AlertTone, page 5-47
DialTone
Description
The Cisco ATA plays the dial tone when it is ready to accept the first digit of a remote address to make
an outgoing call.
Default values for the nine-integer array
•
ntone—2
•
freq0—31538
•
freq1—30831
•
level0—1380
•
level1—1740
•
steady—1
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•
on-time—0
•
off-time—0
•
total time to play tone—1000
Voice Configuration Menu Access Code
920
BusyTone
Description
The Cisco ATA plays the busy tone when the callee is busy.
Default values for the nine-integer array
•
ntone—2
•
freq0—30467
•
freq1—28959
•
level0—1191
•
level1—1513
•
steady—0
•
on-time—4000
•
off-time—4000
•
total time to play tone—0
Voice Configuration Menu Access Code
921
ReorderTone
Description
The Cisco ATA plays the reorder tone (also known as congestion tone) if the outgoing call failed for
reasons other than busy.
Default values for the nine-integer array
•
ntone—2
•
freq0—30467
•
freq1—28959
•
level0—1191
•
level1—1513
•
steady—0
•
on-time—2000
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•
off-time—2000
•
total time to play tone—0
Voice Configuration Menu Access Code
922
RingbackTone
Description
The Cisco ATA plays the ring-back tone when the callee is being alerted by the called device.
Default values for the nine-integer array
•
ntone—2
•
freq0—30831
•
freq1—30467
•
level0—1943
•
level1—2111
•
steady—0
•
on-time—16000
•
off-time—32000
•
total time to play tone—0
Voice Configuration Menu Access Code
923
CallWaitTone
Description
The Cisco ATA plays the call-waiting tone when an incoming call arrives while the user is connected to
another party.
Default values for the nine-integer array
•
ntone—1
•
freq0—30831
•
freq1—0
•
level0—5493
•
level1—0
•
steady—0
•
on-time—2400
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off-time—2400
•
total time to play tone—4800
Voice Configuration Menu Access Code
924
AlertTone
Description
The Cisco ATA plays the alert tone to prompt the user to enter a phone number when invoking a
supplementary service, such as call-forwarding, or blind transfer.
Default values for the nine-integer array
•
ntone—1
•
freq0—30467
•
freq1—0
•
level0—5970
•
level1—0
•
steady—0
•
on-time—480
•
off-time—480
•
total time to play tone—1920
Voice Configuration Menu Access Code
925
CallCmd
Description
Command table that controls call commands such as turning on/off caller ID.
For detailed information on the CallCmd parameter, see Chapter 6, “Call Commands.”
Value Type
Alphanumeric string
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Range
Maximum of 248 characters
Default
•
US command table:
CallCmd:Af;AH;BS;NA;CS;NA;Df;EB;Ff;EP;Kf;EFh;HQ;Jf;AFh;HQ;I*67;gA*82;fA#90v#;OI;H
#72v#;bA#74v#;cA#75v#;dA#73;eA*67;gA*82;fA*70;iA*69;DA*99;xA;Uh;GQ;
•
Sweden command table:
CallCmd:BS;NA;CS;NA;Df;EB;Ff0;ARf1;HPf2;EPf3;AP;Kf1;HFf2;EFf3;AFf4;HQ;Jf1;HFf2;EFf
3;AFf4;HQ;Af4;HQ;I*31#;gA#31#;gA*90*v#;OI;H*21*v#;bA*61*v#;dA*67*v#;cA#21#;eA#61
#;eA#67#;eA*31#;gA#31#;gA*43#;hA#43#;iA*69#;DA*99#;xA
Voice Configuration Menu Access Code
930
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6
Call Commands
This section provides detailed information on call commands for the Cisco ATA:
•
Call Command Structure, page 6-1
•
Syntax, page 6-2
•
Call Command Example, page 6-5
•
Call Command Behavior, page 6-7
Service providers can offer many supplementary services, which can be activated, configured, or
deactivated in more than one way. The CallCmd parameter allows you to define the behavior of
supplementary services that the Cisco ATA supports.
Note
The term Cisco ATA refers to both the Cisco ATA 186 and the Cisco ATA 188, unless otherwise stated.
Note
This section contains call command information for the United States and Sweden. For information
about other countries, contact the Cisco equipment provider for a specific country.
Call Command Structure
The entry in the CallCmd field is a character string composed of a sequence of instructions, which
consist of a combination of three elements:
•
Context—The Cisco ATA supplementary service operation is dependent upon a state and transition
process. For example, the most common state is IDLE, in which the Cisco ATA is on-hook, waiting
for an incoming call. Picking up the telephone handset causes the Cisco ATA to transition to the
PREDIAL state, in which the user hears a dial tone and the Cisco ATA is waiting to detect DTMF
digits. The Context portion of a Call Command string specifies the state for which the commands
are defined.
•
Input-Sequence—The input sequence is simply the input from the user, a combination of hook-flash
and DTMF digits.
•
Action—This specifies the action taken by the Cisco ATA. The action depends on the
Input-Sequence that the user enters and the Context in which it is entered.
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Call Commands
Syntax
Syntax
The CallCmd string has the following structure:
Context-Identifier Command . . . Command; . . . Context-Identifier Command;
•
Table 6-1 provides a list of Context-Identifiers, which show the state of the Cisco ATA.
•
Command consists of the following items:
Input-Sequence; Action-Identifier-1 Action-Identifier-2 [Input-Sequence]
– Input Sequence consists of one or more characters from the set shown in Table 6-2.
– Table 6-3 provides a list of Action Identifiers. Action-Identifier-1 is for the first thread of a
call; Action-Identifier-2 is for the second thread of a call. Each Action Identifier is one
character.
Each Context-Identifier is followed by one or more commands to allow a variable number of actions to
be triggered by relevant user input commands for any state. Each command is composed of an
Input-Sequence that the user enters when the Cisco ATA is in a given state and two Action-Identifier
characters which define the action that the Cisco ATA performs in response to the Context-Identifier and
Input-Sequence. If the Cisco ATA takes only one action, one of the two Action-Identifier characters is a
null action.
Example 6-1
Syntax Example Using One Command
Af;AH;
In this simple example, the first “A” is the Context-Identifier, which means the Cisco ATA is in the
CONFERENCE state, as shown in Table 6-1. The “f” is the input sequence, which is hook-flash, as
shown in Table 6-2. Following the semicolon, the two action identifiers are “A” and “H”. These
identifiers mean “NONE” and “Disconnect the call,” respectively, as shown in Table 6-3. Based on these
action identifiers, the Cisco ATA disconnects the most recent callee, and remains connected to the first
party. The state of the Cisco ATA becomes CONNECTED. Table 6-4 explains more about the various
states of the Cisco ATA.
Example 6-2
Syntax Example Using Two Commands
CN;CAf;OF;
In this example, the first “C” is the Context Identifier, which means the Cisco ATA is in the
PREDIAL_HOLDING state, as shown in Table 6-1. The “N” is the first input sequence, which is any
part of the set of digits 0|1|2|3|4|5|6|7|8|9, as shown in Table 6-2. Following the first semicolon, the two
action identifiers are “C” and “A”, which mean “Continue to Dial” and “NONE,” respectively, as shown
in Table 6-3.
Following this pair of action identifiers is another input sequence, “f”, which means hook-flash, as
shown in Table 6-2. Next is the semicolon, always required after the input sequence, followed by the
corresponding action pair, “O” and “F”. These identifiers mean “Release the Call” and “Retrieve the
Call,” respectively, as shown in Table 6-3.
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Context-Identifiers
Table 6-1
Context-Identifiers
Identifier
Context (State of Cisco ATA)
A
CONFERENCE
B
PREDIAL
C
PREDIAL_HOLDING
D
CONNECTED
E
CONNECTED_HOLDING
F
CONNECTED_ALERTING
G
HOLDING
H
CONFIGURING
I
CONFIGURING_HOLDING
J
3WAYCALLING
K
CALLWAITING
L
IDLE
M
RINGING
N
DIALING
O
CALLING
P
Reserved (ANSWERING)
Q
Reserved (CANCELING)
R
Reserved (DISCONNECTING)
S
WAITHOOK
T
DIALING_HOLDING
U
CALLING_HOLDING
V
Reserved (ANSWERING_HOLDING)
W
Reserved (HOLDING_HOLDING)
X
Reserved (CANCELING_HOLDING)
Y
Reserved (DISCONNECTING_HOLDING)
Z
Reserved (HOLDING_ALERTING)
a
WAITHOOK_ALERTING
b
WAITHOOK_HOLDING
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Input Sequence Identifiers
Table 6-2
Input Sequence Identifiers
Identifier
Input Sequence
0-9,#*
DTMF digits
f
hook flash
o
off-hook
@
anytime; for example, @f means anytime hookflash occurs
h
on-hook
S
#|*
N
0|1|2|3|4|5|6|7|8|9
D
N|S
v
a variable number (1 or more) of characters
from the above list. It must be followed by a
character which acts as the terminator of this
variable part.
Action Identifiers
Table 6-3
Action Identifiers
Identifier
Action
A
NONE
B
Seizure (User intends to dial or configure)
C
Continue to dial
D
Call Return
E
Hold the active call
F
Retrieve the waiting call
G
Cancel the call attempt
H
Disconnect the call
I
Blind transfer the call to the number
N
Go to configuration mode
O
Release the call
P
Answer the incoming call
Q
Transfer with consultation
R
Say busy to the caller
a
None
b
Forward all calls to the given number
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Table 6-3
Action Identifiers (continued)
c
Forward on busy to the given number
d
Forward on no answer to the given number
e
Cancel call forward
f
CLIP for the next call
g
CLIR for the next call
h
Enable Call Waiting for the next call
i
Disable Call Waiting for the next call
x
Enable Fax Mode for the next call
y
Disable Fax Mode for the next call
Call Command Example
In addition to call commands that you configure, the Cisco ATA has a default list of call commands to
handle common call scenarios. Configured call commands overwrite default call commands. If any
Context-Identifier or Input-Sequence elements appear in both the default Call Command string and the
manually entered string, the manually entered value takes precedence.
The following string shows a sample Call Command:
Bf;BAN;CA;CN;CAf;OF;Df;EB;I@f;OF;H@f;OA;Lo;BAf;BA;Mo;PA;ND;CAf;OA;Of;GA;Pf;HA;Qf;OA;Rf;OA;
Sf;OA;TD;CAf;OF;Uf;GF;Vf;HF;Wf;FF;Xf;AF;Yf;AF;Zf;AP;bf;OF;af;OP;
In this section, the Call Command string is broken down into its components as follows:
Call Command Fragment;
Context-Identifier
Input-Sequence1; Action1 Action2;
(optional) Input-Sequence2; Action1 Action2;
Note
If you use a second input sequence, this sequence follows the Action Identifier pair without a
separating semicolon.
Refer to the preceding tables to determine the meanings of the identifiers.
Example 6-3
Call Command String
Bf;BAN;CA;
Predial
hook-flash; Seizure NONE
0|1|...|9; Continue-to-dial NONE;
CN;CAf;OF;
Predial_Holding
0|1|...|9; Continue-to-dial NONE
hook-flash; Release-the-call Retrieve-the-waiting-call;
Df;EB;
Connected
hook-flash; Hold-the-active-call Seizure;
I@f;OF;
Configuring_Holding
hook-flash (at any time); Release-the-call Retrieve-the-waiting-call;
H@f;OA;
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Configuring
hook-flash (at any time); Release-the-call NONE;
Lo;BAf;BA;
Idle
off-hook; Seizure NONE;
hook-flash; Seizure NONE;
Mo;PA;
Ringing
off-hook; Answer-the-incoming-call NONE;
ND;CAf;OA
Dialing
0|1|...|9|#|*; Continue-to-dial NONE
hook-flash; Release-the-call NONE;
Of;GA;
Calling
hook-flash; Cancel-the-call-attempt NONE;
Pf;HA;
Answering
hook-flash; Disconnect-the-call NONE;
Qf;OA;
Canceling
hook-flash; Release-the-call NONE;
Rf;OA;
Disconnecting
hook-flash; Release-the-call NONE;
Sf;OA;
Waithook
hook-flash; Release-the-call NONE;
TD;CAf;OF;
Dialing_Holding
0|1|...|9|#|*; Continue-to-dial NONE;
hook-flash; Release-the-call NONE;
Uf;GF;
Calling_Holding
hook-flash; Cancel-the-call-attempt Retrieve-the-waiting-call;
Vf;HF;
Answering_Holding
hook-flash; Disconnect-the-call Retrieve-the-waiting-call;
Wf;FF;
Holding_Holding
hook-flash; Retrieve-the-waiting-call Retrieve-the-waiting-call;
Xf;AF;
Canceling_Holding
hook-flash; NONE Retrieve-the-waiting-call;
Yf;AF;
Disconnecting_Holding
hook-flash; NONE Retrieve-the-waiting-call;
Zf;AP;
Holding_Alerting
hook-flash; NONE Answering;
bf;OF;
Waithook_Holding
hook-flash; Release-the-call Retrieve-the-waiting-call;
af;OP;
Waithook_Holding
hook-flash; Release-the-call Answer-the-incoming-call;
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Call Command Behavior
Table 6-4 summarizes differing Call Command behavior based on the U.S. and Sweden default call
commands.
U.S. Call Command Default
Af;AH;BS;NA;CS;NA;Df;EB;Ff;EP;Kf;EFh;HQ;Jf;AFh;HQ;I*67;gA*82;fA#90v#;OI;H#72v#;bA#74
v#;cA#75v#;dA#73;eA*67;gA*82;fA*70;iA*69;DA*99;xA;Uh;GQ;Af;AH;
Sweden Call Command Default
BS;NA;CS;NA;Df;EB;Ff0;ARf1;HPf2;EPf3;AP;Kf1;HFf2;EFf3;AFf4;HQ;Jf1;HFf2;EFf3;AFf4;HQ;A
f4;HQ;I*31#;gA#31#;gA*90*v#;OI;H*21*v#;bA*61*v#;dA*67*v#;cA#21#;eA#61#;eA#67#;eA*31#
;gA#31#;gA*43#;hA#43#;iA*69#;DA*99#;xA;Uh;GQ;
Table Notations
The following notations are used in Table 6-4:
Note
Table 6-4
•
FE—Far end
•
AFE—Active Far End, which is a connected far end that is not placed on hold
•
WFE—Waiting Far End, which is a connected far end being placed on hold, or an incoming caller
waiting to be answered
•
R—Hook Flash
•
ONH—On Hook
•
OFH—Off Hook
•
0-9,*,#—DTMF digits
•
v—a variable length string, usually a phone number, and does not include #
•
CWT—call-waiting tone
The notations in Table 6-4 include abbreviations for input sequence behavior. Refer to the tables and
syntax examples shown earlier in this section. The Summary of Commands column in Table 6-4 is based
on the actual command syntax used in the default Call Command strings for the United States and
Sweden.
Call Command Behavior
Cisco ATA State and its Definition
IDLE: Phone is on-hook; Cisco ATA is
waiting for incoming call
Summary of Commands (Input Sequence and Actions)
•
OFH—Start dial tone and go to PREDIAL state.
•
New incoming call or a waiting call (started before it enters IDLE)—Start
ringing the phone and go to the RINGING state.
PREDIAL: Phone just went off-hook but United States and Sweden:
no DTMF has been entered yet;
• # , *—Stop dial-tone, go to the CONFIG state, and prepare to accept a
Cisco ATA plays dial-tone
complete configuration sequence.
•
0-9: Stop dial tone, start invoking dial-plan rules, and go to the DIALING
state to accept a complete phone number.
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Call Command Behavior
Table 6-4
Call Command Behavior (continued)
Cisco ATA State and its Definition
Summary of Commands (Input Sequence and Actions)
DIALING: User is entering phone
number, which is parsed with the given
dial-plan rules
•
R—Abort dialing, restart dial tone, and revert to PREDIAL state.
•
Invalid phone number—Abort dialing, plays fast-busy, and go to
WAITHOOK state.
CONFIG: User configuring a
supplementary service in the United
States
•
*69—Call Return
•
#72v#—Forward unconditional to number specified in 'v'l (PacBell use 72#).
•
#73—Cancel any call forwarding (PacBell use 73#).
•
#74v#—Forward on busy to number specified in 'v' (PacBell does not enable
this service from the phone).
•
#75v#—Forward on no answer to number specified in 'v' (Pac Bell does not
enable this service from the phone).
•
*67—CLIR in the next call (if global profile is CLIP)
•
*82—CLIP for the next call (if global user profile is CLIR)
•
*70—Disable call waiting in the next call.
•
*99—Enable Fax Mode in the next call (non-standard).
•
Dial-tone—Revert to PREDIAL state.
•
Any complete configuration sequence—Carry out the configuration
command, restart dial-tone, and revert to PREDIAL state.
•
*21*v#—Forward unconditionally to number specified in 'v'.
•
*67*v#—Forward on busy to number specified in 'v'.
•
*61*v#—Forward on no answer to number specified in 'v'.
•
#21#—Cancel any call forwarding.
•
#67#—Cancel any call forwarding.
•
#61#—Cancel any call forwarding.
•
#31#—CLIR in the next call.
•
*31#—CLIR in the next call.
•
*43#—Enable call waiting in the next call (Sweden allows globally disable
call waiting).
•
#43#—Disable call waiting in the next call.
•
*69#—Call Return
•
(non-standard)*99#—Enable Fax Mode in the next call (non-standard).
CONFIG: User configuring a
supplementary service in Sweden
All Regions:
CALLING: Phone number is sent;
Cisco ATA is waiting for response from
the far end
•
R or any unrecognized sequence—Abort configuration, restart dial tone and
revert to PREDIAL state.
•
Any complete configuration sequence—Carry out the configuration
command, restart dial tone, and revert to PREDIAL state.
•
R—Cancel the outgoing call, restarts dial-tone, and revert to PREDIAL
state.
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Call Command Behavior
Table 6-4
Call Command Behavior (continued)
Cisco ATA State and its Definition
RINGING: Cisco ATA is ringing the
phone to alert user of an incoming call
Summary of Commands (Input Sequence and Actions)
•
OFH—Stop ringing, answer the call, and go to CONNECTED state.
CONNECTED: The Cisco ATA is
United States and Sweden:
connected with one far end party;
• R—Hold current call, play dial-tone to dial second number, and go to
Cisco ATA may be the caller or the callee
PREDIAL_HOLDING state.
WAITHOOK: Far end hangs up while in
CONNECTED state; Cisco ATA plays
fast-busy after five seconds in this state
CONNECTED_ALERTING: Cisco ATA
receives another call while in
CONNECTED state; Cisco ATA plays
Call Waiting tone periodically (every 10
seconds for US; every second for
Sweden)
•
R—Stop fast-busy, start dial-tone, and go to PREDIAL state.
United States:
•
R—Place current call on-hold, answer the waiting call, and go to
CALLWAITING state.
Sweden:
•
R0—Continue current call, reject the waiting call, and revert to
CONNECTED state.
•
R1—Disconnect current call, answer the waiting call, and go to
CONNECTED state.
•
R2—Place current call on-hold, answer waiting call, and go to
CALLWAITING state.
•
R3—Continue with current call, answer the waiting call and go to
CONFERENCE state.
All Regions:
•
ONH—Disconnect current call and go to IDLE state (the Cisco ATA then
automatically starts ringing the phone, and goes to RINGING state).
•
AFE hangs up—Go to WAITHOOK_ALERTING state, continue to play
CWT.
•
WFE cancels the call—Stop CWT and revert to CONNECTED state.
United States:
CALL WAITING: Cisco ATA is
connected to two far end users on the
• R—Place the AFE on-hold and retrieve the WFE.
same line; one is in active conversation
• ONH—Transfer the WFE to the AFE, drop out of the call, and go to
(the active far end or AFE) while the
PREDIAL state.
other is on-hold (the waiting far end or
WFE). This state is initially entered when Sweden:
the Cisco ATA is connected to one of the
far ends while the other far end calls into • R1—Disconnect current call, answer the waiting call, and go to
CONNECTED state.
the Cisco ATA.
• R2—Place the AFE on-hold and retrieve the WFE.
•
R3—Retrieve the WFE, and go to CONFERENCE state.
•
R4—Transfer the WFE to the AFE, drop out of the call, and go to PREDIAL
state.
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Call Commands
Call Command Behavior
Table 6-4
Call Command Behavior (continued)
Cisco ATA State and its Definition
Summary of Commands (Input Sequence and Actions)
United States:
3WAYCALLING: Cisco ATA is
connected to two far end users on the
• R—Retrieve the WFE and go to CONFERENCE state.
same line; one of them is in active
• ONH—Transfer the WFE to the AFE, drop out of the call, and go to
conversation (the active far end or AFE)
PREDIAL state.
while the other is on-hold (the waiting far
end or WFE). This state is initially
Sweden:
entered when the Cisco ATA is connected
to one of the far ends, then places this far • Same as for CALLWAITING state
end on hold and calls the second far end.
CONFERENCE: Cisco ATA is connected United States:
to two active far ends simultaneously;
• R—Disconnect the last callee and stay connected with the first party, and
Cisco ATA performs audio mixing such
revert to CONNECTED state.
that every party can hear the other two
Sweden:
parties but not themselves.
•
PREDIAL_HOLDING: Cisco ATA user
places a connected call on-hold and
prepares to dial a second number;
Cisco ATA plays dial-tone.
R4—Transfer one FE to the other, drop out of the call, and go to PREDIAL
state.
United States and Sweden:
•
*,#—Stop dial-tone, go to CONFIG_HOLDING state, and prepare to collect
a configuration command.
•
0-9—Stop dial-tone, go to DIALING_HOLDING state, and prepare to
complete dialing a second phone number.
All Regions:
•
Stop dial-tone, retrieve the WFE, and revert to CONNECTED state.
CONFIG_HOLDING: A connected FE is United States:
placed on hold, while the Cisco ATA is
• *67—CLIR for the next call
entering a configuration command.
• *82—CLIP for the next call
•
#90v#—Blind transfer to the number specified in 'v'; disconnect the call and
go to PREDIAL state.
Sweden:
•
#31# or *31#—CLIR in the next call
•
*90*v#—Blind transfer to the number specified in 'v'; disconnect the call
and go to PREDIAL (non-standard) state.
All Regions:
DIALING_HOLDING: Cisco ATA user
is entering a second phone number to call
while placing a connected call on hold
•
R or any unrecognized sequence—Abort configuration, restart dial tone, and
go to PREDIAL_HOLDING state.
•
A complete configuration sequence—Carry out the command, and go to
PREDIAL_HOLDING state.
•
Collected digits match a dial-plan rule—Call the new number, and go to
CALLING_HOLDING state
•
R—Abort dialing and revert to PREDIAL_HOLDING state.
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Call Command Behavior
Table 6-4
Call Command Behavior (continued)
Cisco ATA State and its Definition
Summary of Commands (Input Sequence and Actions)
CALLING_HOLDING: Cisco ATA is
waiting for a second far end to respond
while placing a connected call on hold
•
R—Cancel the call and revert to PREDIAL_HOLDING state.
•
ONH—Cancel the call and transfer the waiting party to the callee, and revert
back to PREDIAL state.
WAITHOOK_HOLDING: The AFE
hangs-up to disconnect the current call
while there is a WFE being put on hold
•
R—Retrieve the WFE and go to CONNECTED state.
AITHOOK_ALERTING: The AFE hangs
up while a waiting call alerts
•
R—Stop CWT, answer the waiting call, and go to CONNECTED state.
•
WFE: Cancel the call; stop CWT, go to WAITHOOK state.
•
ONH—Go to IDLE state (in which Cisco ATA automatically starts ringing
the phone, and goes to RINGING state).
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Call Command Behavior
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7
Configuring and Debugging Fax Services
The Cisco ATA provides two modes of fax services that are capable of internetworking with Cisco IOS
gateways over IP networks. These modes are called fax pass-through mode and fax mode.
With fax pass-through mode, the Cisco ATA encodes fax traffic within the G.711 voice codec and passes
it through the Voice Over IP (VoIP) network as though the fax were a voice call. This mode uses the
Cisco proprietary fax upspeed method.
With fax mode, the Cisco ATA presents itself as a device capable of using only G.711 codecs; therefore,
no codec renegotiation or switchover is required. This places minimum functionality and configuration
requirements on remote gateways. Fax mode is recommended for environments in which G.711 fax
upspeed is not available for the supporting Cisco gateways.
This section contains the following topics:
Note
•
Using Fax Pass-through Mode, page 7-1
•
Using FAX Mode, page 7-6
•
Debugging the Cisco ATA 186/188 Fax Services, page 7-7
The term Cisco ATA is used throughout this manual to refer to both the Cisco ATA 186 and the
Cisco ATA 188, unless differences between the Cisco ATA 186 and Cisco ATA 188 are explicitly stated.
Using Fax Pass-through Mode
Fax pass-through mode allows for maximum codec flexibility because users may set up a voice call using
any voice codec, then renegotiate to a G.711 codec for the fax session. To use fax pass-through mode,
first configure the Cisco ATA and supporting Cisco gateways to support the Cisco-proprietary G.711fax
upspeed method. Then, disable fax relay on the far-end gateway—either for the entire gateway or for the
dial peer engaged in the fax call with the Cisco ATA.
The fax upspeed method allows you to use low bit-rate codecs such as G.723 and G.729 for voice calls,
and G.711 codecs for fax calls. With a fax call, the Cisco ATA detects a 2100-Hz CED tone or V.21
preamble flag, then informs the remote gateway of its intent to switchover to G.711 via a peer-to-peer
message. This type of message, carried as a Named Signaling Event (NSE) within the RTP stream, is
used for all fax event signaling. The Cisco ATA can initiate and respond to NSEs and can function as
either an originating or terminating gateway.
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Configuring and Debugging Fax Services
Using Fax Pass-through Mode
Note
The Cisco ATA can also accept standard-based protocol-level codec switch requests, but cannot send
such requests. Therefore, to interoperate with a Cisco gateway, use the Cisco-proprietary codec switch.
This section contains the following topics:
•
Configuring the Cisco ATA for Fax Pass-through mode, page 7-2
•
Configuring Cisco IOS Gateways to Enable Fax Pass-through, page 7-3
Configuring the Cisco ATA for Fax Pass-through mode
Fax Pass-through mode requires configuring two configuration parameters:
AudioMode, page 7-2
ConnectMode, page 7-3
AudioMode
Description
The AudioMode parameter is a 32-bit value. The lower 16 bits apply to the Phone 1 port of the
Cisco ATA and the upper 16 bits apply to the Phone 2 port of the Cisco ATA.
Example
The following is an example of configuring the Phone 1 port of the Cisco ATA for fax pass-through
mode:
0xXXXX0015
Translation
This setting translates to the following bitmap:
xxxx xxxx xxxx xxxx 0000 0000 0001 0101
Note
•
Bit 0 = 1—Enables G.711 silence suppression (VAD)
•
Bit 2 = 1—Enables Fax CED tone detection and switchover upon detection
•
Bit 4 = 1, Bit 5 = 0—DTMF transmission method = out-of-band through negotiation
•
Bit 6 = Bit 7 = 0—Hookflash transmission method = disable sending out hookflash
The values XXXX in the example apply to the Phone 2 port of the Cisco ATA.
To configure the same value for the Phone 2 port of the Cisco ATA, the value would be
The configuration of one port is independent from the configuration of the other port.
0x0015XXXX.
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Configuring and Debugging Fax Services
Using Fax Pass-through Mode
ConnectMode
Description
The ConnectMode parameter is a 32-bit value. The parameter settings apply to both lines of the
Cisco ATA. Configure ConnectMode after configuring AudioMode for fax pass-through mode. Cisco
recommends you use the following ConnectMode setting to interoperate with a Cisco IOS gateway.
Recommended Setting
0x90000400
Translation
This setting translates to the bitmap:
1001 0000 0000 0000 0000 0100 0000 0000
Bit 2 and bits 7 through 15 are the only relevant bits for fax pass-through mode. These bits from the
example are isolated below:
xxxx xxxx xxxx xxxx 0000 0100 0xxx x0xx
•
Bit 2 = 0—Uses RTP payload number 126/127 for fax upspeed to G.711µ−law/G.711A-law. Set this
value to 1 if you want to use RTP payload number 0/8 for fax upspeed.
•
Bit 7 = 0—Disables fax pass-through redundancy. Set this bit to 1 to enable redundancy. With
redundancy enabled, the Cisco ATA sends each packet twice. Because of bandwidth and
transmission time costs, use this option only if network quality is poor and all other gateways used
in the network support this feature.
•
Bits {12, 11, 10, 9, 8} = {0, 0, 1, 0, 0}—Sets the offset to NSE payload-type number 96 to 4. Setting
the offset to 4 results in the Cisco ATA sending an NSE payload-type value of 100 by default. Valid
offset values range from 2 to 23 (NSE payload type value of 98 to 119). Set this value to match the
value for your Cisco gateways.
Most Cisco MGCP-based gateways, such as Cisco 6608, use NSE payload type 101 by default. Most
Cisco H.323/SIP-based gateways use NSE payload type 100 by default.
•
Bit 13 = 0—Uses G.711µ−law for fax pass-through upspeed. Set this bit to 1 to use G.711A for fax
pass-through upspeed.
•
Bit 14 = Bit 15 = 0—Enables fax pass-through mode using the Cisco proprietary method
(recommended). Set both of these bits to 1 to disable fax pass-through mode.
Configuring Cisco IOS Gateways to Enable Fax Pass-through
To configure your IOS gateways to network with Cisco ATA, do the following:
Procedure
Step 1
Enable Fax Pass-through Mode, page 7-4
Step 2
Disable Fax Relay Feature, page 7-5
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Configuring and Debugging Fax Services
Using Fax Pass-through Mode
Note
For detailed information on setting up your IOS gateways and on feature availability, refer to the
document Cisco Fax Services over IP.
Enable Fax Pass-through Mode
The supporting Cisco gateway can enable fax pass-through mode using system-level or dial-peer-level
commands.
System Level commands
Enable the fax pass-through feature using the following system-level commands:
Procedure
Step 1
Run the following command:
voice service voip
Step 2
Run the following command:
modem passthrough NSE [payload-type number] codec {g711µ/law | g711alaw} [redundancy]
[maximum-sessions value]
The definitions of the command parameters are as follows:
•
The payload-type parameter default is 100. Valid values are from 98 to 119.
The NSE payload number must be the same on both the Cisco ATA and the Cisco gateway.
•
The codec parameter must be G.711µ−law for faxes sent over a T1 trunk or G.711A-law for faxes
sent over an E1 trunk.
•
The redundancy parameter enables RFC 2198 packet redundancy. It is disabled by default.
•
The maximum sessions parameter defines the number of simultaneous fax pass-through calls with
redundancy. The default is 16. Valid values are 1 to 26.
When using the voice service voip and modem passthrough nse commands on a terminating gateway
to globally set up fax or modem pass-through with NSEs, you must also ensure that each incoming call
will be associated with a VoIP dial peer to retrieve the global fax or modem configuration. You associate
calls with dial peers by using the incoming called-number command to specify a sequence of digits that
incoming calls can match. You can ensure that all calls will match at least one dial peer by using the
following commands:
Router(config)# dial-peer voice tag voip
Router(config-dial-peer)# incoming called-number .
Step 3
For the Cisco ATA ConnectMode parameter, turn off bits 14 and 15. This enables the sending of fax
pass-through signals and the detection of incoming fax pass-through signals using the Cisco proprietary
method.
Note
The NSE payload-type number, fax pass-through codec (G.711µ-law or G.711A-law) and redundancy
parameters must have the same settings for the Cisco ATA that they have for supporting Cisco gateways.
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Configuring and Debugging Fax Services
Using Fax Pass-through Mode
Dial-Peer Level Commands
You can enable fax pass-through mode for communication between a Cisco IOS gateway and the
specified Cisco ATA using the following dial-peer level commands:
Procedure
Step 1
Perform the command:
dial-peer voice tag voip
Step 2
Perform the command:
modem passthrough {NSE [payload-type number] codec {g711µlaw | g711alaw} [redundancy] |
system}
a.
The default of this command is:
modem passthrough system
When using the default configuration, the dial-peer fax pass-through configuration is defined by the
voice service voip command. When the system option is used, no other parameters are available.
When the NSE is configured in the fax pass-through command at the dial-peer level, the fax
pass-through definition in the dial-peer command takes priority over the definition in the voice
service voip command.
b.
The payload-type number, codec, and redundancy parameters can also be used.
For example, the command:
modem passthrough NSE codec g711µlaw
means that the Cisco ATA will use the NSE payload-type number 100, G.711µ-law codec, and no
redundancy in fax pass-through mode.
Step 3
When setting up dial-peer for fax pass-through, it is necessary to set up a pair of dial-peers for inbound
and outbound calls between the Cisco ATA and Cisco IOS gateways. You do this by specifying the
destination-pattern and incoming-called number. The destination-pattern should point to the
Cisco ATA, while the incoming-called number should apply to all numbers that the Cisco ATA is allowed
to dial.
Disable Fax Relay Feature
Fax relay may be enabled by default for some IOS gateways. If you do not disable the fax relay feature,
it may override the precedence of fax/modem pass-through and cause the fax transmission to fail. It is
necessary to disable fax relay at the dial-peer or system level with the following command:
fax rate disable
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Configuring and Debugging Fax Services
Using FAX Mode
Using FAX Mode
Use fax mode when the gateways in the network do not support fax pass-through mode or dial-peer
configuration.
You can set one or both lines of the Cisco ATA to G.711-only fax mode. This mode allows the fax
machine connected to the Cisco ATA to communicate directly with the far endpoint with no fax signaling
event occurring between the two gateways.
This section contains the following topics:
•
Configuring the Cisco ATA for Fax Mode, page 7-6
•
Configuring the Cisco ATA for Fax Mode on a Per-Call Basis, page 7-7
•
Configuring the Cisco IOS Gateway for Fax Mode, page 7-7
Configuring the Cisco ATA for Fax Mode
G.711-only fax mode operation requires configuration of one parameter—AudioMode.
Description
The AudioMode parameter is a 32-bit value. The lower 16 bits apply to the Phone 1 port of the
Cisco ATA, and the upper 16 bits to the Phone 2 port. The following is an example of the Phone 1 port
of the Cisco ATA configured for G.711-only fax mode:
Example
0xXXXX0012
Translation
This setting translates to the bitmap:
xxxx xxxx xxxx xxxx 0000 0000 0001 0010
Note
•
Bit 0 = 0—Disables G.711 silence suppression (VAD).
•
Bit 1 = 1—Uses G.711 only, does not user the low bit-rate codec.
•
Bit 2 = 0—Disables Fax CED tone detection.
•
Bit 4 = 1, Bit 5 = 0—DTMF transmission method: out-of-band through negotiation
•
Bit 6 = Bit 7 = 0—Hookflash transmission method: disables sending out hookflash
The values XXXX in the example do not apply to the Phone 1 port of the Cisco ATA.
To configure the same value for the Phone 2 port of the Cisco ATA, the value would be 0x0012XXXX. The
configuration of one port is independent from the configuration of the other port.
Note
The AudioMode configuration overrides the values of the following three parameters: RxCodec,
TxCodec, and LBRCodec. For example, if these three parameters are each set to 0 (for G.723), the
Cisco ATA would still use G.711 if AudioMode is set to 0x00120012. With this configuration, the
Cisco ATA sends both G.711µ-law and G.711A-law as preferred codecs to a peer voice gateway.
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Configuring and Debugging Fax Services
Debugging the Cisco ATA 186/188 Fax Services
Configuring the Cisco ATA for Fax Mode on a Per-Call Basis
Note
The per-call-basis fax mode feature is only available for the H.323 and SIP protocols.
If you want to activate fax mode on a per-call basis, configure the following parameters:
Procedure
Step 1
CallFeatures and PaidFeatures Bit 15 (for line1—mask 0x8000) and Bit 31 (for line2—mask
0x80000000) = 1: This sets the default to enable fax mode on a per-call basis.
Step 2
AudioMode Bit 2 = 0: This disables fax CED tone detection.
Step 3
CallCmd includes *99;xA (99 is the default; the value can be changed to any prefix code.)
To activate a call from your fax machine, enter *99 (default), then enter the telephone number to which
you want to send the fax. The next call will automatically revert to normal mode.
Configuring the Cisco IOS Gateway for Fax Mode
On the Cisco gateway, disable both fax relay and fax pass-through at the dial-peer level or system level
with the following commands:
Procedure
Step 1
Run the command:
fax rate disable
Step 2
Run the command:
no modem passthrough
Debugging the Cisco ATA 186/188 Fax Services
This section includes the following debugging topics for fax services:
•
Common Problems When Using IOS Gateways, page 7-7
•
Using prserv for Diagnosing Fax Problems, page 7-9
•
Using rtpcatch for Diagnosing Fax Problems, page 7-12
Common Problems When Using IOS Gateways
Table 7-1 lists typical problems and actions that might solve these problems for situations in which the
Cisco ATA is using fax over a Cisco IOS gateway.
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Debugging the Cisco ATA 186/188 Fax Services
Table 7-1
Solving Common Fax Problems
Problem
Action
The far-end gateway is not Cisco recommends IOS version 12.2 (11)T or higher for the Cisco 2600
loaded with correct
and Cisco 3600, and IOS version 12.1 (3)T or higher for Cisco AS5300.
software image.
The Cisco 6608 supports both the NSE and NTE methods of fax
pass-through mode, beginning with software version D004030145S16608.
To use fax pass-through mode with the Cisco 6608, the user must select
6608 NSE mode, and the NSE payload type must be reconfigured to match
the Cisco ATA.
The Cisco ATA is not
loaded with the proper
software.
Cisco recommends using software version 2.14 or higher.
Cisco recommends using Cisco ATA models 186-I1, 186-I2, 188-I1, or
User is operating
Cisco ATA software on an 188-I2 (hardware platforms).
outdated model.
For fax mode, the AudioMode configuration parameter should be set to
The Cisco ATA is not
configured for fax mode or 0xXXXX0012 (X = value not applicable) for the Phone 1 port of the
Cisco ATA, and 0x0012XXXX for the Phone 2 port.
fax pass-through mode.
For fax pass-through mode, AudioMode should be set to 0xXXXX0015
for the Phone 1 port of the Cisco ATA, and 0x0015XXXX for the Phone
2 port.
The remote gateway is not When the Cisco ATA is configured for fax pass-through mode, all remote
configured for modem/fax gateways must be configured with modem/fax pass-through mode either
on a dial-peer level or system level.
pass-through mode.
Fax relay is not disabled
on the remote gateway.
Fax relay is enabled by default on some Cisco gateways. When fax relay
is enabled, it can override fax pass-through mode and cause fax failure.
Examples of the CLI commands to disable fax relay for IOS gateways are
as follows:
•
fax rate disable for H.323/SIP gateways
•
mgcp fax t38 inhibit for MGCP gateways
Fax/modem pass-through
method on the remote
gateway is not compatible
with the Cisco NSE-based
method.
Some Cisco gateways (such as Cisco VG248, and Cisco 6608) may use
signaling messages based on RFC2833 for G.711 upspeed when loaded
with older software images. This method is incompatible with the Cisco
NSE-based method.
NSE payload types differ
between gateways.
The Cisco ATA has a configurable NSE packet payload-type value whose
default is 100. This value is compatible with the implementations of most
Cisco gateways. However, some Cisco gateways use 101 as the NSE
payload type.
You must check to make sure that the image on your gateway supports the
Cisco NSE-based fax/modem pass-through. Otherwise, you must
configure the Cisco ATA to use fax mode.
Ensure that all gateways in your environment use the same NSE payload
type if you wish to successfully use fax pass-through mode.
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Using prserv for Diagnosing Fax Problems
This section contains the following topics:
•
prserv Overview, page 7-9
•
Analyzing prserv Output for Fax Sessions, page 7-9
prserv Overview
prserv is a tool that runs on a Microsoft Windows-based PC and serves as a log server that captures
debug information that the Cisco ATA sends to your PC IP address/port. The debug information is saved
into a readable text file.
To enable your Cisco ATA to send debug information, you need to set the NPrintf configuration
parameter to your PC IP address and an available port, as shown in the following procedure:
Procedure
Step 1
.
is the IP address of your PC.
is any unused port (any number from 1024 to 65535) on your PC.
Note
Step 2
You can the Nprintf parameter on the Cisco ATA configuration web page or with the TFTP-based
configuration method.
To operate the debug capture program prserv.exe, place the prserv program in a folder on your PC. At
the DOS prompt, enter:
C:>prserv
is the port number you have selected. If is omitted, the default port number is 9001.
As prserv receives debug information from the Cisco ATA, it displays the information on the DOS screen
and saves it to the output file .log.
Once you are finished capturing debug information, you can stop prserv by entering Ctrl-C at the DOS
prompt. If you restart the process without changing the name of the log file, any new debug information
is appended to the end of the original file.
Analyzing prserv Output for Fax Sessions
The debug log obtained from prserv is for detecting simple configuration problems.
Note
A comprehensive understanding of the fax events requires the use of the rtpcatch tool (see the “Using
rtpcatch for Diagnosing Fax Problems” section on page 7-12).
Table 7-2 lists log events relevant to analyzing a fax session.
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Table 7-2
Debug Log Examples
Log event
Description
[ch] Enable encoder
Voice encoder type pt is enabled for the channel ch, where pt can be 0
for G.711µ-law, 4 for G.723.1, 8 for G.711A-law, and 18 for G.729.
For example, [0]Enable encoder 4 indicates that the Cisco ATA
transmitted G.723.1-encoded voice packets.
[ch] DPKT 1st:
, pt
The first voice packet that the Cisco ATA received was of RTP payload
type pt for the channel ch with timestamp of timestamp1, and the local
decoding timestamp was set to timestamp2.
For example, [0]DPKT 1st: 1491513359 1491512639, pt
that the first RTP packet that the Cisco ATA received was
G.723.1-encoded for channel 0.
4
indicates
[ch] codec: => Voice codec switchover occurred. The voice encoder type switched from
pt1 to pt2 for the channel ch.
For example, [0]codec: 4 => 0 indicates that the local voice encoder
on the Cisco ATA switched from G.723.1 to G.711µ-law.
[ch] Rx MPT PT=
NSE pkt
Channel ch received an NSE packet of event with payload type of NSEpt.
For event, c0XXXXXX indicates a CED tone event, and c1XXXXXX
indicates a phase reversal event.
For example, [0]Rx MPT PT=100 NSE pkt c0000000 indicates that the
Cisco ATA received a CED tone event NSE packet with payload type of
100.
[ch] Tx MPT PT= NSE Channel ch transmitted an NSE packet of event with payload type of
pkt
NSEpt. For event, c0XXXXXX indicates a CED tone event, and
c1XXXXXX indicates a phase reversal event.
For example, [0]Tx MPT PT=100 NSE pkt c0000000 indicates that the
ATA transmitted a CED tone event NSE packet with payload type of
100.
Debugging FAX Pass-through Mode
When the Cisco ATA is configured to use fax pass-through mode, the fax call session can be established
with an arbitrary voice codec. Once the voice call has been established, fax machines can signal their
presence by means of a CED tone or V.21 preamble flag, after which the gateways send NSE packets to
initiate switchover.
Note
For fax pass-through mode, check the Cisco ATA debug log to verify that it is acting as an originating
gateway as well as a terminating gateway.
Terminating-Gateway Example
When the Cisco ATA is used as a terminating gateway for a fax session, make sure the following
conditions are true:
•
The Cisco ATA transmits CED-tone-event NSE packets.
•
The encoder switchover to G.711 occurs during the NSE-packet transaction.
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An example debug log for a terminating gateway scenario is show below:
[0]Tx MPT PT=100 NSE pkt c0000000
[0]codec: 4 => 0
[0]Rx MPT PT=100 NSE pkt c0000000
Note
The NSE response to the CED tone event is not mandatory; some gateways may not send back an NSE
response.
Originating-Gateway Example
When the Cisco ATA is used as an originating gateway for a fax session, make sure that the following
conditions are true:
•
The Cisco ATA receives and responds to CED-tone-event NSE packets.
•
The NSE payload type is the same for the received and transmitted NSE packets.
•
The encoder switchover to G.711 occurs during NSE-packet transaction.
An example debug log for an originating gateway scenario is shown below:
[0]Rx MPT
[0]Tx MPT
[0]codec:
[0]Rx MPT
[0]Rx MPT
Note
PT=100
PT=100
4 => 0
PT=100
PT=100
NSE pkt c0000000
NSE pkt c0000000
NSE pkt c0000000
NSE pkt c0000000
If your gateway is using a legacy IOS software image, it may not send NSE packets but instead may rely
on a straightforward codec switchover mechanism. In this case, a codec switchover event occurs rather
than an NSE packet transaction.
Possible Reasons for Failure
If your Cisco ATA does not receive CED-tone-event NSE packets and codec switchover does not occur,
the failure may be due to the following reasons:
•
The terminating gateway is not configured with fax/modem pass-through.
•
The fax pass-through mode used by the terminating gateway may not be compatible with the Cisco
NSE method.
If the log shows proper NSE packet transaction and G.711 upspeed for your fax session but the session
still fails, check that the following conditions are true:
•
The Cisco ATA software image version is 2.14 or above.
•
The Cisco ATA model number is ATA186-I1, ATA186-I2, ATA188-I1, or ATA188-I2.
•
The fax relay option for the remote gateways has been disabled.
Debugging FAX Mode
When the Cisco ATA is configured with fax mode, only G.711 codecs are used. You must confirm that
only 0 (for G.711µ-law) or 8 (for G.711A-law) appear in the Enable encoder and DPKT 1st debug lines.
The following example of a debug log shows that G.711µ-law is used:
[0]Enable encoder 0
[0]DPKT 1st: 1491513359 1491512639, pt 0
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If the numeric codes for the G.711 codecs do not appear in the log, you need to check your AudioMode
parameter setting on the Cisco ATA.
If the correct G.711 codecs appear in the log but your fax sessions still fail, check that the following
conditions are true:
•
The Cisco ATA software image version is 2.14 or above.
•
The Cisco ATA model number is ATA186-I1, ATA186-I2, ATA 188-I1, or ATA188-I2.
•
The fax relay option for the remote gateways has been disabled.
Using rtpcatch for Diagnosing Fax Problems
This section contains the following topics:
•
rtpcatch Overview, page 7-12
•
Example of rtpcatch, page 7-13
•
Analyzing rtpcatch Output for Fax Sessions, page 7-16
•
Using rtpcatch to Analyze Common Causes of Failure, page 7-17
•
rtpcatch Limitations, page 7-19
rtpcatch Overview
rtpcatch is a tool that provides comprehensive information for a VoIP connection. The tool runs on a
Microsoft Windows-based PC and is capable of parsing an output capture file from Network Associates
(NAI) Sniffer Pro and identifies significant fax pass-through and fax relay events.
Major functions
rtpcatch includes the following major functions:
•
Reads session data from Sniffer Pro capture files.
•
Analyzes media streams.
•
Stores media streams to files.
•
Reports RTP statistics such as the number of RTP packets, the number of RTP frames, the number
of lost packets, the number of filler packets during silence suppression periods, and the number of
erased packets.
How to Use
To use rtpcatch, follow these steps:
Procedure
Step 1
Create a working directory for rtpcatch and place the executable file rtpcatch.exe in this directory.
Step 2
Copy your Network Associates Sniffer Pro capture files into this directory.
Step 3
At the DOS prompt of this directory, enter the following command:
:>rtpcatch [] [options]
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– is the NAI Sniffer capture file.
– is the prefix prepended to the output filenames.
Output Files
The output files of rtpcatch include a summary file and audio stream files.
The summary file is .sum if is specified, otherwise it is file.sum.
Stream files are labeled with an integer tag beginning with 00. Stream files are also tagged with the
extension pcm for G.711A/G.711µ-law, 723 for G723.1, 729 for G729, t38 for T.38, and cfr for Cisco
Fax Relay.
Options
rtpcatch options include:
•
-fax—to
output the fax events for a connection.
The output includes "FAX summary 1" as the interleaved event list for all directions, and "FAX
summary 2" as the event list for each direction. The reported events include voice codec change,
NSE signalling, and fax relay events.
•
-port —to
discard any packets sent from/to this port.
If the NAI Sniffer capture file includes Cisco ATA prserv packets, these packets can interfere with
rtpcatch analysis. Some prserv packets might be interpreted as NTE or NSE events. To prevent
such interference, you can either disable debugging output on the Cisco ATA (do this by setting the
Nprintf configuration parameter to 0), configure your NAI Sniffer to filter out the prserv packets,
or run rtpcatch with the -port options.
Note
rtpcatch works best for analyzing a single VoIP session. Command-line options can be entered in any
order.
Example of rtpcatch
The section contains an example of using rtpcatch and includes an explanation of its output:
Output
C:\>rtpcatch faxpassthru -fax
[
[
[
[
[
[
[
[
[
[
25]open file: 00.723, (G723) 2.213:10000 => 2.116:10002
26]open file: 01.723, (G723) 2.116:10002 => 2.213:10000
29] <00> 1 silence pkts from TS 1760 (seq# 3)
42] <00> 2 silence pkts from TS 4400 (seq# 9)
47] <00> 2 silence pkts from TS 5600 (seq# 11)
55] <00> 2 silence pkts from TS 7760 (seq# 15)
101]open file: 02.pcm, (G711u) 2.116:10002 => 2.213:10000
106] <02> 2
lost pkts from seq# 39
107]open file: 03.pcm, (G711u) 2.213:10000 => 2.116:10002
110] <03> 1 silence pkts from TS 19440 (seq# 41)
------------ Summary --------------
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Input file: faxpassthru.cap
<00.723>: (G723) 2.213:10000 => 2.116:10002
total 38 pkts(70 frames), lost 0 pkts, fill 7 silence pkts
<01.723>: (G723) 2.116:10002 => 2.213:10000
total 38 pkts(76 frames), lost 0 pkts, fill 0 silence pkts
<02.pcm>: (G711u) 2.116:10002 => 2.213:10000
total 2181 pkts(2181 frames), lost 2 pkts, fill 0 silence pkts
<03.pcm>: (G711u) 2.213:10000 => 2.116:10002
total 2179 pkts(2179 frames), lost 0 pkts, fill 1 silence pkts
---------- FAX Summary 1 ---------[
[
[
[
[
[
[
25]<2.213=>2.116>
26]<2.116=>2.213>
101]<2.116=>2.213>
102]<2.116=>2.213>
103]<2.116=>2.213>
105]<2.213=>2.116>
107]<2.213=>2.116>
Codec G723
Codec G723
Codec G711u/D
NSE PT 100, EVT 192: Up-Speed, CED tone Detected
NSE PT 100, EVT 193: ECAN OFF, Phase Reversal Detected
NSE PT 100, EVT 192: Up-Speed, CED tone Detected
Codec G711u/D
---------- FAX Summary 2 ---------PATH: 2.213:10000 => 2.116:10002
[
25]Codec G723
[ 105]NSE PT 100, EVT 192: Up-Speed, CED tone Detected
[ 107]Codec G711u/D
PATH: 2.116:10002 => 2.213:10000
[
26]Codec G723
[ 101]Codec G711u/D
[ 102]NSE PT 100, EVT 192: Up-Speed, CED tone Detected
[ 103]NSE PT 100, EVT 193: ECAN OFF, Phase Reversal Detected
Explanation
The output is printed on screen and saved in the file file.sum.
The following lines are described:
•
[
25]open file: 00.723, (G723) 2.213:10000 => 2.116:10002
This indicates that rtpcatch reached NAI Sniffer packet number 25 and opened a new file named
00.723 to store an audio stream consisting of G.723-compressed data. The audio path originates
from the IP address ending with 2.213 and port 10000 (written as <2.213:1000>) and terminates at
the IP address ending with 2.116 and port 10002.
•
[
29] <00>
1 silence pkts from TS 1760 (seq# 3)
This indicates that rtpcatch detected one silence RTP packet in the audio path <00> and the silence
packet began at timestamp 1760. This occurred at packet number 29 with the RTP sequence number
3.
•
[
106] <02>
2
lost pkts from seq# 39
This indicates that rtpcatch detected two lost RTP packets in the audio path <02>. The missing
packets began with sequence number 39. This occurred at packet number 106.
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•
------------ Summary -------------Input file: faxpassthru.cap
<00.723>: (G723) 2.213:10000 => 2.116:10002
total 38 pkts(70 frames), lost 0 pkts, fill 7 silence pkts
This indicates that the input filename is faxpassthru.cap. The output file 00.723 contains the
G.723-compressed stream from <2.123:10000> to <2.116:10002>; 38 packets (70 frames) were
processed by rtpcatch. No lost packets were detected and seven silence packets were found.
•
---------- FAX Summary 1 ---------[
[
[
[
[
[
[
25]<2.213=>2.116>
26]<2.116=>2.213>
101]<2.116=>2.213>
102]<2.116=>2.213>
103]<2.116=>2.213>
105]<2.213=>2.116>
107]<2.213=>2.116>
Codec G723
Codec G723
Codec G711u/D
NSE PT 100, EVT 192: Up-Speed, CED tone Detected
NSE PT 100, EVT 193: ECAN OFF, Phase Reversal Detected
NSE PT 100, EVT 192: Up-Speed, CED tone Detected
Codec G711u/D
This indicates that the audio streams originating at <2.213> and <2.216> are G.723-compressed.
The audio stream from <2.116> was then up-sped to G.711µ-law at packet number 101. The NSE
signaling packets were sent at packet number 102, 103 and 105. Finally, the audio stream from
<2.113> was up-sped to G.711µ-law.
•
---------- FAX Summary 2 ---------PATH: 2.213:10000 => 2.116:10002
[
25]Codec G723
[ 105]NSE PT 100, EVT 192: Up-Speed, CED tone Detected
[ 107]Codec G711u/D
PATH: 2.116:10002 => 2.213:10000
[
26]Codec G723
[ 101]Codec G711u/D
[ 102]NSE PT 100, EVT 192: Up-Speed, CED tone Detected
[ 103]NSE PT 100, EVT 193: ECAN OFF, Phase Reversal Detected
This summarizes the fax events for each path.
The audio stream events reported by rtpcatch include:
– beginning of new audio codec
– silence packets
– lost packets
– erased packets (as in G.729)
The NSE events reported by rtpcatch include:
– event 32, Fax Mode, CED tone Detected (RFC2833)
– event 34, Modem Mode, ANSam tone Detected (RFC2833)
– event 192, Up-Speed, CED tone Detected
– event 193, ECAN OFF, Phase Reversal Detected
– event 194, ECAN ON, Silence Detected
– event 200, T38 Fax Mode, V.21 Detected
– event 201, T38 Fax Mode ACK
– event 202, T38 Fax Mode NACK
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– event 203, Modem Relay Mode, CM Tone Detected
– event Cisco Fax Relay (with RTP payload type 96)
– event Cisco Fax Relay ACK (with RTP payload type 97)
Analyzing rtpcatch Output for Fax Sessions
The following examples show the proper fax events when gateways are configured to operate in the
following modes:
•
Cisco ATA fax mode
•
Cisco ATA fax pass-through mode
•
T.38 fax relay mode
•
Cisco fax relay mode
Example 7-1
Fax Mode
---------- FAX Summary 1 ---------[
25]<2.131=>3.200> Codec G711u
[
26]<3.200=>2.131> Codec G711u
Analysis
Both sides use G.711 for the entire fax session.
Example 7-2
Fax Pass-through Mode
---------- FAX Summary 1 ---------[
25]<2.213=>2.116> Codec G723
[
26]<2.116=>2.213> Codec G723
[ 101]<2.116=>2.213> Codec G711u/D
[ 102]<2.116=>2.213> NSE PT 100, EVT 192: Up-Speed, CED tone Detected
[ 103]<2.116=>2.213> NSE PT 100, EVT 193: ECAN OFF, Phase Reversal Detected
[ 105]<2.213=>2.116> NSE PT 100, EVT 192: Up-Speed, CED tone Detected
[ 107]<2.213=>2.116> Codec G711u/D
Analysis
Note
•
Both sides initially use G.723.
•
<2.116> switches to G.711µ-law using a dynamic payload type.
•
NSE signaling packets are sent from <2.116>.
•
An optional NE signaling packet is sent from <2.213>.
•
<2.113> switches to G.711µ-law using a dynamic payload type.
EVT 193 may not appear for some fax transmission.
Example 7-3
Fax Pass-through Mode
---------- FAX Summary 1 ---------[
37]<3.200=>2.53> Codec G723
[
41]<2.53=>3.200> Codec G723
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[
[
[
136]<3.200=>2.53> Codec G711u/D
137]<3.200=>2.53> NSE PT 100, EVT 192: Up-Speed, CED tone Detected
140]<2.53=>3.200> Codec G711u/D
Analysis
•
Both sides initially use G.723.
•
<3.200> switches to G.711µ-law using a dynamic payload type.
•
NSE signaling packets are sent from <3.200>.
•
<2.53> switches to G.711µ-law using a dynamic payload type.
Example 7-4
T38 Fax Relay Mode
---------- FAX Summary 1 ---------[
15]<2.53=>3.99> Codec G711u
[ 486]<3.99=>2.53> Codec G711u
[ 1277]<3.99=>2.53> Codec T38
[ 1278]<2.53=>3.99> Codec T38
Analysis
•
Both sides initially use G.711µ-law.
•
Both sides switch to T.38
Example 7-5
Cisco Fax Relay
---------- FAX Summary 1 ---------[
8]<2.53=>3.99> Codec G711u
[ 248]<3.99=>2.53> Codec G711u
[ 798]<2.53=>3.99> NSE PT 96, Cisco Fax Relay
[ 799]<3.99=>2.53> NSE PT 97, EVT 192: Up-Speed, CED tone Detected
[ 800]<2.53=>3.99> NSE PT 97, Cisco Fax Relay ACK
[ 801]<2.53=>3.99> Codec C_FxRly
[ 803]<3.99=>2.53> NSE PT 96, EVT 192: Up-Speed, CED tone Detected
[ 804]<2.53=>3.99> NSE PT 97, Cisco Fax Relay ACK
[ 805]<3.99=>2.53> Codec C_FxRly
Analysis
•
Both sides initially use G.711µ-law.
•
NSE signaling packets are sent between <2.53> and <3.99>.
•
Both sides switch to Cisco fax relay.
Using rtpcatch to Analyze Common Causes of Failure
The following examples show the rtpcatch output of failed fax sessions. <3.200> is ATA; <2.53> is a
Cisco gateway.
Example 7-6
Cisco ATA Configuration Failure
---------- FAX Summary 1 ----------
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[
[
37]<2.53=>3.200> Codec G723
39]<3.200=>2.53> Codec G723
Analysis
•
<2.53> is the originating gateway and <3.200> is the terminating Cisco ATA.
•
The Cisco ATA and the <2.53> gateway use G.723 codec.
Possible Causes for Failure
•
The Cisco ATA is not configured with fax mode or fax pass-through mode.
•
If the Cisco ATA is the gateway for a fax sender, the remote gateway is not configured with fax
pass-through mode.
Example 7-7
Fax Mode Failure
---------- FAX Summary 1 ---------[
37]<2.53=>3.200> Codec G711
[
39]<3.200=>2.53> Codec G711
[ 1820]<2.53=>3.200> NSE PT 96, Cisco Fax Relay
[ 1966]<2.53=>3.200> NSE PT 96, Cisco Fax Relay
Analysis
•
<2.53> is the originating gateway and <3.200> is the terminating Cisco ATA.
•
The Cisco ATA and the <2.53> gateway begin with G.711 codec.
•
The <2.53> gateway sends Cisco fax relay event packets.
Possible Cause for Failure
•
Cisco fax relay option is not disabled on the gateway.
Example 7-8
Fax Pass-through Mode Failure
---------- FAX Summary 1 ---------[
2]<2.53=>3.200> Codec G723
[
4]<3.200=>2.53> Codec G723
[ 106]<3.200=>2.53> Codec G711u/D
[ 107]<3.200=>2.53> NSE PT 100, EVT 192: Up-Speed, CED tone Detected
[ 1436]<3.200=>2.53> NSE PT 100, EVT 192: Up-Speed, CED tone Detected
Analysis
•
<2.53> is the originating gateway, and <3.200> is the terminating Cisco ATA.
•
The Cisco ATA upspeeds to G.711µ-law and sends G.711 upspeed NSE signaling packets.
•
The <2.53> gateway does not respond to the NSE signaling packets.
Possible Causes for Failure
•
Fax/modem pass-through option is not enabled on the gateway.
•
Fax/modem pass-through NSE payload type are configured differently on the Cisco ATA and the
gateway.
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Example 7-9
Fax Pass-through Mode Failure
---------- FAX Summary 1 ---------[
37]<2.53=>3.200> Codec G723
[
39]<3.200=>2.53> Codec G723
[ 143]<3.200=>2.53> Codec G711u/D
[ 144]<3.200=>2.53> NSE PT 100, EVT 192: Up-Speed, CED tone Detected
[ 1602]<3.200=>2.53> NSE PT 100, EVT 192: Up-Speed, CED tone Detected
[ 1604]<2.53=>3.200> Codec G711u/D
[ 1820]<2.53=>3.200> NSE PT 96, Cisco Fax Relay
[ 1966]<2.53=>3.200> NSE PT 96, Cisco Fax Relay
Analysis
•
<2.53> is the originating gateway, and <3.200> is the terminating Cisco ATA.
•
The Cisco ATA upspeeds to G.711µ-law and sends G.711 upspeed NSE signaling packets.
•
The <2.53> gateway upspeeds to G.711µ-law and then sends Cisco fax relay event packets.
Possible Cause for Failure
•
Cisco fax relay option is not disabled on the gateway.
Example 7-10 Fax Pass-through Mode Failure
---------- FAX Summary 1 ---------[
33]<3.200=>2.53> Codec G729
[
39]<2.53=>3.200> Codec G729
[ 562]<2.53=>3.200> NTE PT 101, EVT
[ 563]<2.53=>3.200> NTE PT 101, EVT
[ 565]<2.53=>3.200> NTE PT 101, EVT
[ 566]<2.53=>3.200> Codec G711u/D
[ 568]<2.53=>3.200> NTE PT 101, EVT
[ 580]<3.200=>2.53> Codec G711u/D
34: Modem Mode, ANSam tone Detected (RFC2833)
34: Modem Mode, ANSam tone Detected (RFC2833)
34: Modem Mode, ANSam tone Detected (RFC2833)
34: Modem Mode, ANSam tone Detected (RFC2833)
Analysis
•
<3.200> is the originating Cisco ATA, and <2.53> is the terminating gateway.
•
Both sides initially use G.729.
•
<2.53> gateway sends NTE signaling packets, then upspeeds to G.711µ-law.
•
<3.200>The Cisco ATA switches to G.711µ-law also, but never sends NTE signaling packets.
•
Fax transmission fails because <2.53> gateway does not receive any NTE packets, and it drops the
fax call.
Possible Cause for Failure
•
The Cisco ATA does not support the NTE signaling method and requires that the gateways use the
NSE signaling method.
rtpcatch Limitations
•
rtpcatch performs optimally when analyzing capture files containing only one VoIP session.
•
rtpcatch detects only G.711A, G.711µ-law, G.723, G.729, T.38, Cisco fax relay, modem
pass-through with or without redundancy packets, RTCP packets and NSE packets.
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Configuring and Debugging Fax Services
Debugging the Cisco ATA 186/188 Fax Services
•
rtpcatch can handle a maximum of 20 prserv ports using the -port option.
•
rtpcatch may not detect T.38 packets correctly.
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8
Upgrading the Cisco ATA Signaling Image
This section describes two methods for upgrading the Cisco ATA software for the SIP protocol:
•
Upgrading the Signaling Image from a TFTP Server, page 8-1—This is the Cisco-recommended
method for the SIP protocol. This method is the most efficient method and requires only a one-time
configuration change.
•
Upgrading the Signaling Image Manually, page 8-2—This method can be used if you must manually
upgrade the image of one Cisco ATA. However, this method is not the recommended upgrade
method because it is not as simple as the TFTP method.
This section also describes procedures for verifying a successful image upgrade:
•
Caution
Note
Confirming a Successful Signaling Image Upgrade, page 8-5—Procedures for using your Web
browser or the voice configuration menu are included.
Do not unplug the Cisco ATA while the function button is blinking. Doing so can cause permanent
damage to the device. The function button blinks during an upgrade.
The term Cisco ATA is used throughout this manual to refer to both the Cisco ATA 186 and the
Cisco ATA 188, unless differences between the Cisco ATA 186 and Cisco ATA 188 are explicitly stated.
Upgrading the Signaling Image from a TFTP Server
You can configure the Cisco ATA to automatically download the latest signaling image from the TFTP
server. You do this configuring the parameter upgradecode in your Cisco ATA configuration file. (You
also would use this procedure if you wanted to perform a cross-protocol signaling image upgrade.) For
more information about setting up the configuration file, see the “Creating Unique and Common
Cisco ATA Configuration Files” section on page 3-8.
Syntax of upgradecode Parameter
upgradecode:3,0x301,0x0400,0x0200,tftp_server_ip,69,image_id,image_file_name
Definitions
•
The hexadecimal values that precede the tftp_server_ip variable must always be the values shown
in the syntax.
•
tftp_server_ip is the TFTP server that contains the latest signaling image file.
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Upgrading the Signaling Image Manually
•
image_id is a unique 32-bit integer that differs with each upgrade. You can determine this 32-bit
integer value by using the build date on the image file name and prepending it with "0x". For
example, if the image_file_name is ata186-v2-14-020514a.kxz, then the build date is 020508a, and
the image_id is 0x020508a).
•
image_file_name is the firmware upgrade-image file name. The image_file_name format is:
ata186-v{M}-{N}-{yymmdd}{a-f}{ext}
– - M is the major version number
– - N is the minor version number (always two digits)
– - yymmdd is a two-digit year, two-digit month, and two-digit day
– - a-f is the build letter (- yymmdd and a-f together form the build date of the image)
– - ext must be ".kxz" for upgrading from version 2.11 and below, and can be ".zup" for
upgrading from version 2.12 and up for the Cisco ATA186, but it must be ".zup" for upgrading
the Cisco ATA188.
Process
Whenever the Cisco ATA administrator stores a new signaling image (denoted by a change to the
image_id), the Cisco ATA upgrades its firmware with the new image_file. To contact the TFTP server,
the Cisco ATA uses the TFTP server IP address that is contained within the value of the upgradecode
parameter.
Example
The upgradecode parameter value could be:
upgradecode:3,0x301,0x0400,0x0200,192.168.2.170,69,0x020723a,ata186-v2-15-020
723a.zup
This instructs the Cisco ATA to upgrade its firmware to ata186-v2-15-020723a.zup by downloading the
ata186-v2-15-020723a.zup file from the TFTP server IP address of 192.168.2.170. This download
occurs after the Cisco ATA downloads its configuration file that contains the directive from the
upgradecode parameter. Also, the upgrade occurs only if the internally cached image_id in Cisco ATA
is different from the value 0x020723a.
Upgrading the Signaling Image Manually
This section describes how to manually upgrade the Cisco ATA with the most recent signaling image.
The executable file that you need is called ata186us.exe, and is bundled in the Cisco ATA
release-software zip file.
This section contains the following topics:
•
Preliminary Steps, page 8-3
•
Running the Executable File, page 8-3
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Upgrading the Signaling Image Manually
Preliminary Steps
Before you run the executable file, be sure to complete the following procedure:
Procedure
Step 1
If you are a registered CCO user. go to the following URL:
http://www.cisco.com/cgi-bin/tablebuild.pl/ata186
Step 2
Locate the zip file that contains the software for the applicable release and signaling image you are using.
The contents of each file are described next to the file name. Extract the signaling image file (this file
has an extension of .zup—For example, ata186-v2-15-020723a.zup) and store it on the PC that has
connectivity with the Cisco ATA.
Step 3
Set the Cisco ATA parameter UseTftp to 0.
Note
Step 4
Remember to set this parameter back to 1 before you use the TFTP upgrade method at a later
time.
Follow the instructions in the “Running the Executable File” section on page 8-3.
Running the Executable File
This section includes the procedure for running the executable file and using the voice configuration
menu to complete the upgrade process. First check to make sure the upgrade requirements are met and
determine the syntax to use when running the program.
This section contains the following topics:
•
“Upgrade Requirements” section on page 8-3
•
“Syntax” section on page 8-3
•
“Upgrade Procedure” section on page 8-4
Upgrade Requirements
The following list contains the requirements for using the ata186us.exe file and the voice configuration
menu to upgrade the Cisco ATA to the latest signaling image:
•
A network connection between the PC from which you will invoke the executable file and the
Cisco ATA
•
A PC running Microsoft Windows 9X/ME/NT/2000
Syntax
ata186us [-any] {-h[host_ip]} {-p[port]} {-quiet} [-d1 -d2 -d3]
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Definitions
•
-any—Allow upgrade regardless of software and build versions (recommended).
•
-h[host_ip]—Set the upgrade server to a specific IP address in cases where there may be more
than one IP address for the host. The default behavior is that the program will use the first IP address
it obtains when it runs the gethostbyname command.
•
-p[port]—Set the server port to a specific port number (the default port number is 8000; use a
different port number only if you are setting up an upgrade server other than the default).
•
-quiet—Quiet mode; send all output to log file named as [port].log (useful when running the
upgrade server as a daemon).
•
-d1,-d2,-d3—Choose a verbosity level for debugging, with -d3 being the most verbose.
•
image file—This is the name of the signaling image file to which the Cisco ATA will upgrade.
Example
To upgrade the Cisco ATA to the signaling image ata186-v2-15-020723a.zup, you can use the following
syntax:
ata186us -any -d1 ata186-v2-15-020723a.zup
Upgrade Procedure
To perform the upgrade, follow these steps:
Procedure
Step 1
Run the executable file (see the “Syntax” section on page 8-3) from the Microsoft Windows DOS or
command prompt. You will receive instructions on how to upgrade.
Step 2
On the Cisco ATA, press the function button to invoke the voice configuration menu.
Step 3
Using the telephone keypad, enter the following:
100# ip_address_of_PC * port #
This is the IP address of the PC and the port number at the DOS prompt where you invoked the
ata186us.exe file.
For example, if the IP address is 192.168.1.10, and the port number is 8000 (the default), then enter:
100#192*168*1*10*8000#
When the upgrade is complete, the "Upgrade Successful" prompt will sound.
Note
When upgrading many Cisco ATAs manually, you can save the software-upgrade dial-pad sequence in
your telephone's speed-dial, and use this sequence repeatedly.
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Upgrading the Cisco ATA Signaling Image
Confirming a Successful Signaling Image Upgrade
Confirming a Successful Signaling Image Upgrade
You can verify that you have successfully upgraded the Cisco ATA signaling image by using one of the
following methods:
•
Using a Web Browser, page 8-5
•
Using the Voice Configuration Menu, page 8-5
Using a Web Browser
To use your web browser to verify a successful image upgrade, perform the following steps:
Procedure
Step 1
Open your web browser.
Step 2
Enter the IP address of your Cisco ATA Web configuration page:
http:///dev
Step 3
Refresh the page to clear the cache.
The image version number and its build date should appear at the bottom-left corner of the Cisco ATA
Web configuration page.
Using the Voice Configuration Menu
To use the voice configuration menu to verify a successful image upgrade, perform the following steps:
Procedure
Step 1
Pick up the telephone handset attached to the Phone1 port of the Cisco ATA.
Step 2
Press the function button on the Cisco ATA.
Step 3
Press 123# on the telephone keypad to play out the image version number.
Step 4
Press 123123# on the telephone keypad to play out the image build date.
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Confirming a Successful Signaling Image Upgrade
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9
Troubleshooting
This section describes troubleshooting procedures for the Cisco ATA:
Note
•
General Troubleshooting Tips, page 9-1
•
Symptoms and Actions, page 9-2
•
Installation and Upgrade Issues, page 9-3
•
Debugging, page 9-4
•
Frequently Asked Questions, page 9-5
•
Contacting TAC, page 9-6
The term Cisco ATA is used throughout this manual to refer to both the Cisco ATA 186 and the
Cisco ATA 188, unless differences between the Cisco ATA 186 and Cisco ATA 188 are explicitly stated.
General Troubleshooting Tips
The suggestions in this section are general troubleshooting tips.
•
Make sure that the DHCP server is operating correctly. Note that the function button blinks slowly
when the Cisco ATA attempts to acquire the DHCP configuration.
•
If the green activity LED is not flashing after you connect the Ethernet cable, make sure that both
the power cord and the Ethernet connection are secure.
•
If there is no dial tone, make sure that the telephone line cord from the telephone is plugged into the
appropriate port on the Cisco ATA. Make sure that your Cisco ATA is properly registered on your
Call Control system. Test another phone; if this phone does not work either, there may be a problem
with the current configuration or with the Cisco ATA.
•
A busy tone indicates that the party you called is not available. Try your call again later. A fast-busy
tone indicates that you dialed an invalid number.
•
After power up, if the function button continues to blink slowly, the Cisco ATA cannot locate the
DHCP server. Check the Ethernet connection and the availability of the DHCP server.
•
The DHCP server should show an incoming request from the MAC address listed on the product
label or given by the voice prompt.
•
If you place a call to another IP telephone, detect ringing, and the called party answers but you
cannot detect the speaker’s voice, verify that the Cisco ATA and the other IP telephone support at
least one common audio codec: G.711A-law, G.711µ-law, G.723.1, or G.729A.
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Symptoms and Actions
Symptoms and Actions
Symptom Parameters with values set by using the web server interface or voice configuration menu revert
to their original settings.
Possible Cause You are using TFTP for configuration (the UseTFTP parameter is set to 1). The
Cisco ATA has a cached version of its configuration file stored in its flash memory; this is what
displayed or played through the web server interface or voice configuration menu. If UseTFTP is set
to 1, then the cached value of the Cisco ATA configuration file is synchronized with its configuration
file located at the TFTP server. This synchronization update of the cached value occurs at
approximate intervals determined by the CFGInterval parameter value as well as when the Cisco ATA
powers up or resets.
Recommended Action If you are using TFTP for configuration, do not use the web server interface or
voice configuration menu to modify the value of the Cisco ATA configuration file. Use the web server
interface or voice configuration menu only to initially configure the Cisco ATA to contact the TFTP
server for the Cisco ATA configuration file.
Symptom Unable to access the web configuration page.
Possible Cause Software versions earlier than 2.0 require the web configuration page to be enabled
using option 80# on the voice configuration menu.
Recommended Action Upgrade the software.
Symptom The Cisco ATA does not seem to be configured using the TFTP server.
Possible Cause The TFTP server address is not properly set.
Recommended Action Ensure that the TftpURL is correctly set to the URL or IP address of the TFTP
server that is hosting the configuration file for the Cisco ATA. If you are using DHCP to supply the
TFTP server IP address, make sure that the TftpURL is set to 0. Also, unless the TftpURL is an IP
address, be sure that the DNS1IP and DNS2IP values are properly set to resolve the TftpURL
supplied by DHCP.
Symptom The Cisco ATA contacts the TFTP server more often than specified in the CfgInterval
parameter.
Possible Cause The ToConfig parameter is not set to 0.
Recommended Action After the Cisco ATA has a valid configuration file, the ToConfig parameter must
be set to 0. If it is not set to 0, the Cisco ATA will attempt to contact the TFTP server too frequently.
Symptom Cannot place call.
Possible Cause Equipment failure on the network.
Recommended Action Replace defective network equipment.
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Installation and Upgrade Issues
Possible Cause Recipient has not registered the IP phone.
Recommended Action Register the IP phone.
Possible Cause Ethernet cable is not connected.
Recommended Action Make sure that all cables are connected.
Symptom Fast busy tone.
Possible Cause Authentication credential is incorrect.
Recommended Action Verify authentication credential, and revise if necessary.
Possible Cause Recipient has not registered the IP phone.
Recommended Action Register the IP phone.
Possible Cause No common codec between the Cisco ATA and remote end.
Recommended Action Change codec to one that is common with the Cisco ATA and the remote end.
Possible Cause Recipient is in a call with call waiting disabled.
Recommended Action Attempt to place the call at a later time.
Installation and Upgrade Issues
Note
The following issues apply to the manual image-upgrade process only. Image upgrades must be
performed separately.
Symptom The red LED is flashing slowly on the function button.
Possible Cause The Cisco ATA is trying to obtain the DHCP address or the software image is being
upgraded.
Possible Cause The Ethernet cable is unplugged.
Recommended Action Plug in the Ethernet cable.
Symptom Voice prompt returns Upgrade not available message. This can only occur if you are using the
executable-file upgrade method.
Possible Cause You are attempting to upgrade to the existing version.
Recommended Action You do not need to upgrade.
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Debugging
Symptom Voice prompt returns Upgrade failed message. This can only occur if you are using the
executable-file upgrade method.
Possible Cause You have entered an incorrect IP address.
Recommended Action Enter the correct IP address.
Possible Cause Software image is corrupted.
Recommended Action Upgrade software image.
Symptom No dial tone.
Possible Cause No user ID was entered.
Recommended Action Enter the correct user ID.
Symptom Incorrect dial tone.
Possible Cause Check the web interface for your DialTone setting. The default is U.S.
Recommended Action Set the correct country DialTone value.
Debugging
The MS-DOS Windows-based debugging program tool, preserv.exe, is included in every software
upgrade package. The tool is also available from Cisco TAC. The prserv program is used in conjunction
with the NPrintf configuration parameter. This file serves as an upgrade server that captures debug
information sent by the Cisco ATA software to your PC’s IP address and port number. This debug file
(prserv.exe) compiles the information from the Cisco ATA into a readable log file. To capture this
"NPRINTF" information, you must know the IP address of the PC using the prserv program, illustrated
as follows:
IP address.port
where IP address is the IP address of your PC, and port is 9001. If another process on your PC already
uses port 9001, you may use some other value (legal values are from 1024 to 65535). If no port value is
entered, the default value is 9001.
To enter the IP address and port number, use voice menu option 81#. You must enter the IP address and
port number in alphanumeric format, which requires entering the * key after every character entered. To
enter the "." character, you must enter the sequence 1 1#.
For example, for a computer with the IP address 172.28.78.90 and port number 9001
(172.28.78.90.9001), you would enter the following on your telephone handset:
1* 7* 2* 1 1* 2* 8* 1 1* 7* 8* 1 1* 9* 0* 1 1* 9* 0* 0* 1* *
To operate the debug capture program prserv.exe, place the prserv program in a folder on your PC; then
at the DOS prompt of the folder where you have placed it, enter:
C:> prserv port.log
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Frequently Asked Questions
where port is the port number you have selected. If you do not enter port.log, debug information still
appears on your screen, but it is not saved to a log file.
After you finish capturing debug information, you can stop the log program by entering Ctrl-C at the
DOS prompt. The log file created is named port.log. If you restart the process without changing the name
of the log file, any new debug information is appended to the end of the original file.
Contact Cisco TAC for more information. See the “Obtaining Technical Assistance” section on page xvi
for instructions.
You should also have access to a sniffer or LAN analyzer.
Caution
For security reasons, Cisco recommends that you do not use the web interface over the public network.
Disable the web interface, using the UIPassword parameter, before the Cisco ATA is moved from the
service provider site.
Frequently Asked Questions
Q. How can I recover the box if I forgot the password?
A. There are two important passwords. One is the UIPassword, which protects access to the Cisco ATA
Web Server interface; the other is the EncryptKey, which protects access to the TFTP configuration file.
If you forget the value for the UIPassword but still have access to TFTP-stored configuration file, you
can modify the UIPassword via TFTP. However, if you are not configuring the Cisco ATA via TFTP, or
if you forget both passwords, the only way you can recover the box is to have physical access to the box
and do a factory reset on the box via the box voice configuration menu interface (Access Code:
FACTRESET#).
Q. What is the maximum distance from which I can drive an analog device with a Cisco ATA?
A. Table 9-1 provides maximum distances for this question.
Table 9-1
Ring Loads and Distances
Ring Load (per RJ-11 FXS Port)
Maximum Distance
5 REN
200 feet (61 m)
4 REN
1000 feet (305 m)
3 REN
1700 feet (518 m)
2 REN
2500 feet (762 m)
1 REN
3200 feet (975 m)
The Cisco ATA, however, is not designed for long distance. The simple test is to determine if the phone
or phones that are connected to the Cisco ATA work properly in their environment.
Pay attention to the following questions:
1.
Can the Cisco ATA detect on/off hook from the analog phone?
2.
Can the Cisco ATA detect the DTMF signal?
3.
Can you dial the remote side?
4.
Can the Cisco ATA ring the phone?
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Contacting TAC
5.
Is voice quality satisfactory?
If you answer no to any of the above questions, you may have a loop impedance greater than 400 ohm.
In this case, perform the following procedure.
Procedure
Step 1
Increase the wire gauge to reduce the impedance until the Cisco ATA can detect on/off hook and DTMF
signal.
Step 2
If the Cisco ATA cannot ring the phone, find a phone that can ring at a lower ringing voltage. Also, try
to use only one phone instead of multiple phones in parallel.
Q. Does the Cisco ATA support an overhead paging system, and, if so, does the Cisco ATA support
power denial?
A. The Cisco ATA supports an overhead paging system only if that system does not require power
denial (battery removal) when a call is disconnected. However, the Cisco ATA can be configured to
reverse the voltage polarity when a call is connected or disconnected. For more information, see the
“Polarity” section on page 5-27.
Contacting TAC
Qualified customers who need to contact the Cisco Technical Assistance Center (TAC) must provide the
following information:
•
Product codes.
•
Software version number—To identify the software revision number, use the configuration menu
number 123.
•
Hardware version number—To identify the hardware revision number, use the serial number and
MAC address found on the label on the bottom of the Cisco ATA. The MAC address can also be
obtained using voice menu option 24.
•
Software build information—To identify the software build information, use the voice menu option
123123.
•
Cisco ATA serial number.
See the “Obtaining Technical Assistance” section on page xvi for instructions on contacting TAC.
Note
Customers who obtained their equipment through service providers, independent dealers and other third
parties must contact their equipment provider for technical assistance.
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A
Using SIP Supplementary Services
SIP supplementary services are services that you can use to enhance your telephone service. These
services include call forward, call return, call forwarding and conference calling. Use the following
parameters to enable and subscribe to supplementary services:
•
CallFeatures, page 5-23—Use this parameter to enable desired features.
•
PaidFeatures, page 5-24—Use this parameter to subscribe or unsubscribe to enabled features.
•
This section contains the following topics:
•
Changing Call Commands, page A-1
•
Cancelling a Supplementary Service, page A-1
•
Common Supplementary Services, page A-1
Changing Call Commands
To change the command for a supplementary service (for example, to change *69 to *100), change the
context identifiers in the Call Command field on the Web configuration page. For more information, see
Chapter 6, “Call Commands.”
Note
You cannot change supplementary services by means of the voice configuration menu.
Cancelling a Supplementary Service
You can deactivate some supplementary services by pressing *70 before making a call. You can also
configure your system to have services disabled by default and enabled on a call-by-call basis. Use the
32-bit Call Features plan to handle your services in this manner. For more information, see the
“CallFeatures” section on page 5-23.
Common Supplementary Services
The supplementary services described in this section, and their configuration and implementation,
depend on the system of the country in which the service is activated. For information about your
country’s implementation of services, contact your local Cisco equipment provider.
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Appendix A
Using SIP Supplementary Services
Common Supplementary Services
This section contains the following topics:
•
Caller ID, page A-2
•
Call-Waiting Caller ID, page A-2
•
Voice Mail Indication, page A-2
•
Unattended Transfer, page A-3
•
Attended Transfer, page A-4
•
Making a Conference Call in the United States, page A-4
•
Making a Conference Call in Sweden, page A-4
•
Call Waiting in the United States, page A-5
•
Call Waiting in Sweden, page A-5
•
About Call Forwarding, page A-5
•
Call Forwarding in the United States, page A-5
•
Call Forwarding in Sweden, page A-6
•
Call Return in the United States, page A-6
•
Call Return in Sweden, page A-6
•
Calling Line Identification Presentation, page A-6
•
About Calling Line Identification Restriction, page A-6
•
Calling Line Identification Restriction in the United States, page A-7
•
Calling Line Identification Restriction in Sweden, page A-7
Caller ID
When the telephone rings, the Cisco ATA sends a Caller ID signal to the telephone between the first and
second ring (with name, telephone number, time, and date information, if these are available).
Call-Waiting Caller ID
The Cisco ATA plays a call waiting tone, then sends an off-hook Caller ID signal to the telephone
immediately after the first tone burst.
The Cisco ATA sends the name, telephone number, time, and date information, if these are available.
Voice Mail Indication
This feature allows the Cisco ATA to play an intermittent dial tone if there is a message in the user's voice
mail box.
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Using SIP Supplementary Services
Common Supplementary Services
Unattended Transfer
This feature allows a user to transfer an existing call to another telephone number without waiting for
the dialed party to answer before the user hangs up. Two methods exist for performing an unattended
transfer:
•
Semi-unattended Transfer, page A-3
•
Fully Unattended Transfer, page A-3
Semi-unattended Transfer
Perform the following steps to complete a semi-unattended transfer:
Procedure
Step 1
Press the flash button on the telephone handset to put the other party on hold and get a dial tone.
Step 2
Dial the telephone number to which you would like to transfer the other party.
Step 3
Wait for at least one ring and then hang up your phone to transfer the other party.
Fully Unattended Transfer
Perform the following steps to complete a fully unattended transfer:
Procedure
Step 1
Press the flash button on the telephone handset to put the other party on hold and get a dial tone.
Step 2
Press #90 (the transfer service activation code) on your telephone keypad, then enter the phone number
to which you want to transfer the other party, then press #.
Step 3
Hang up your phone.
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Using SIP Supplementary Services
Common Supplementary Services
Attended Transfer
This feature allows a user to transfer an existing call to another telephone number after first consulting
with the dialed party before the user hangs up. Perform the following steps to complete an attended
transfer:
Procedure
Step 1
Press the flash button on the telephone handset to put the existing party on hold and get a dial tone.
Step 2
Dial the telephone number to which the existing party is being transferred.
Step 3
When the callee answers the phone, you may consult with the callee and then transfer the existing party
by hanging up your telephone handset.
Making a Conference Call in the United States
Procedure
Step 1
Dial the first number.
Step 2
When the person you called answers, press the flash or receiver button on the telephone handset. This
will put the first person you called on hold and you will receive a dial tone.
Step 3
Dial the second person and speak normally when that person answers.
Step 4
To conference with both callers at the same time, perform a hook flash.
Step 5
To drop the second call, perform a hook flash.
Step 6
(Optional) To conference in additional callers, the last person called with a Cisco ATA can call an
additional person, that new person can then call someone else, and so on. This is known as
daisy-chaining.
Making a Conference Call in Sweden
Procedure
Step 1
Dial the first number.
Step 2
When the person you called answers, press the flash or receiver button on the telephone handset. This
will put the first person you called on hold and a dial tone will sound.
Step 3
Dial the second person and speak normally when that person answers.
Step 4
Perform a hook flash, then press 2 on your telephone keypad to return to the first person. You can
continue to switch back and forth between the two callers.
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Using SIP Supplementary Services
Common Supplementary Services
Step 5
To conference with both callers at the same time, perform a hook flash, then press 3 on the telephone
keypad. Once you conference all three callers, the only way to drop a caller is for that caller to hang up.
Step 6
(Optional) To conference in additional callers, the last person called with a Cisco ATA can call an
additional person, that new person can call someone else, and so on. This is known as “daisy-chaining.”
Call Waiting in the United States
If someone calls you while you are speaking on the telephone, you can answer by performing a hook
flash. You cannot conference in all three callers, but the first person you called could call someone else
and daisy-chain them into the conference.
When the Cisco ATA is configured to use Call Waiting by default, press *70 on your telephone keypad
to disable Call Waiting for the duration of the next call.
Call Waiting in Sweden
If someone calls you while you are speaking on the telephone, you can answer by performing a hook
flash then pressing 2 on your telephone keypad, or you can conference them with the person to whom
you are already speaking by performing a hook flash then pressing 3. You can also perform a hook flash
then press 3 later during the call to create a conference call.
Performing a hook flash then pressing 1 hangs up the first caller and answers the second call. If there is
no answer after one minute, the caller receives three beeps and a busy signal.
To enable call waiting for Sweden, press *43#. When the Cisco ATA is configured to use Call Waiting
by default, press #43# to disable Call Waiting for the duration of the next call.
About Call Forwarding
In SIP, the Cisco ATA can control call forwarding and call return.
There are three types of call forwarding:
•
Forward Unconditional—Forwards every call that comes in.
•
Forward When Busy—Forwards calls when the line is busy.
•
Forward on No Answer—Forwards calls when the telephone is not answered after the configured
period of 0-63 seconds.
You can activate only one of these services at a time.
Call Forwarding in the United States
Forward Unconditional
Press #72 on your telephone keypad; enter the number you want to forward call to; then press # again.
Forward When Busy
Press #74 on your telephone keypad; enter the number to forward the calls to; then press # again.
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Using SIP Supplementary Services
Common Supplementary Services
Forward On No Answer
Press #75 on your telephone keypad; enter the number you want to forward the calls to; then press #
again.
Cancelling Call Forwarding
To cancel call forwarding, press #73 on your telephone keypad
Call Forwarding in Sweden
Forward Unconditional
Press *21* on your telephone keypad; enter the number you want to forward calls to; then press #. To cancel,
press #21#.
Forward When Busy
Press *67* on your telephone keypad; enter the number to forward the calls to; then press #. To cancel, press
#61#.
Forward On No Answer
Press *61* on your telephone keypad; enter the number you want to forward the calls to; then press #.
To cancel, press #67#.
Forward On No Answer with a Specified Call Forward Delay
Press *61*on your telephone keypad; enter the number you want to forward the calls to; then press * and the
number of seconds for the call forward delay; then press # again. To cancel, press #67# on your telephone
keypad.
Call Return in the United States
Press *69 on your telephone keypad to activate call return in the United States.
Call Return in Sweden
Press *69# on your telephone keypad to activate call return in Sweden.
Calling Line Identification Presentation
Calling Line Identification Presentation (CLIP) shows your identity to callers with Caller ID.
Press *82 on your telephone keypad to activate CLIP.
About Calling Line Identification Restriction
Calling Line Identification Restriction (CLIR) hides your identity from callers with Caller ID.
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Using SIP Supplementary Services
Common Supplementary Services
Calling Line Identification Restriction in the United States
Press *67 on your telephone keypad to activate CLIR. This feature is disabled when you hang up.
Calling Line Identification Restriction in Sweden
Press *31# on your telephone keypad to activate CLIR. This feature is disabled when you hang up.
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Using SIP Supplementary Services
Common Supplementary Services
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A P P E N D I X
B
Voice Menu Codes
This section contains a quick-reference list of the voice configuration menu options for the Cisco ATA.
This section contains the following tables:
•
Cisco ATA Voice Menu Codes—Information Options, page B-1
•
Cisco ATA Voice Menu Codes—Configuration Parameters, page B-2
•
Cisco ATA Voice Menu Codes—Software Upgrade, page B-4
Note
Follow each voice menu code with #.
Note
The term Cisco ATA refers to both the Cisco ATA 186 and the Cisco ATA 188, unless otherwise stated.
Table B-1 lists codes to return basic Cisco ATA information.
Table B-1
Cisco ATA Voice Menu Codes—Information Options
Option
Voice Menu
Code
Description
Build information
123123
Build date of the Cisco ATA software
Review IP address
21
Returns IP address of the Cisco ATA
Review MAC address
24
Returns media access control (MAC) address of the
Cisco ATA
Review network route IP address
22
Returns IP address of the network route
Review subnet mask
23
Returns subnet mask of the network route
Version number
123
Returns version number of the Cisco ATA software
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Appendix B
Voice Menu Codes
Table B-2 lists configuration codes.
Table B-2
Cisco ATA Voice Menu Codes—Configuration Parameters
Option
Voice Menu
Code
Description
Alternate NTP IP address
78
IP address of the alternate NTP server
Audio mode
312
Allows finer control of the audio component to suit
certain user applications
Call features
314
Subscribed features statically enabled by the user
Caller ID method
316
Specifies the signal format when generating the
Caller ID format to use
TFTP Configuration Interval
80002
Interval (in seconds) between configuration
updates when TFTP configuration is used,
Connection mode
311
Controls the connection mode of the call signaling
protocol
Dynamic Host Configuration
Protocol (DHCP)
20
Controls whether the Cisco ATA can automatically
obtain configuration parameters from a server over
the network
DNS 1 IP
916
IP address of the primary DNS server
DNS 2 IP
917
IP address of the secondary DNS server
Encrypt key
320
Encrypts the configuration file on the TFTP server
Num Tx frames
35
Number of frames transmitted per packet
Gatekeeper/proxy server IP
address
5
SIP registration proxy server IP address
IP address
1
IP address of the Cisco ATA
LBR codec
300
Low bit rate codec selection
Login ID 0
46
Alternate user ID used for authentication
Login ID 1
47
Alternate user ID used for authentication
Media port
202
Specifies which base port the Cisco ATA uses to
receive RTP media streams
Network route address
2
Network router address
NPrintf address
81
IP address of a host to which all Cisco ATA debug
messages are sent
NTP server address
141
IP address of the NTP server
Paid features
315
Features subscribed to by the user
Polarity
304
Controls connect and disconnect polarity
PWD 0
4
Password associated with the primary phone line
(UID0 or LoginID0)
PWD 1
14
Password associated with the secondary phone line
(UID1 or LoginID1)
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Voice Menu Codes
Table B-2
Cisco ATA Voice Menu Codes—Configuration Parameters (continued)
Option
Voice Menu
Code
Rx codec
36
Selects the audio codec type to use to decode
received data. The call-receiving station
automatically adjusts to the call-initiating station's
audio codec type if the call-receiving station
supports that audio codec.
Set password
7387277
Configuration interface password
Signal timers
318
Timeout values controlling the starting or stopping
of a signaling event
SIP max number of redirects
205
Maximum number of redirections the Cisco ATA
will attempt to reach a callee when making a call
SIP NAT IP address
200
WAN IP of the NAT
SIP outbound proxy
206
IP address of the outbound proxy server to which
all outgoing SIP requests are sent
SIP port
201
Specifies which port the Cisco ATA listens to for
incoming SIP messages
SIP protocol
38
Selects the signaling protocol
SIP registration On
204
Enables SIP registration
SIP registration period
203
Interval (in seconds) between each registration
renewal to the SIP registration server
Subnet mask
10
Specifies the subnet mask for the Cisco ATA
TFTP URL
905
IP address of the TFTP server when TFTP
configuration is used
Timezone
302
Specifies offset to GMT—used to time-stamp
incoming calls for caller ID
ToConfig
80001
Identifies unconfigured or already-configured
Cisco ATAs
Trace Fflags
313
Enables logging of debug information
Tx Codec
37
Selects transmitting audio codec preference
UDP TOS bits
255
Determines the precedence and delay of UDP IP
packets
UID 0
3
User ID (telephone number) for the PHONE 1 port
UID 1
13
User ID (telephone number) for the PHONE 2 port
Use login ID
93
Determines which pair (UIDx, PWDx or LoginIDx,
PWDx) to use for authentication
Use TFTP
305
Enables TFTP as configuration method
Description
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Appendix B
Voice Menu Codes
Table B-3 lists codes used in the software upgrade process. For information about these codes, see
Appendix , “Upgrading the Cisco ATA Signaling Image.”
Table B-3
Cisco ATA Voice Menu Codes—Software Upgrade
Option
Voice Menu
Code
Description
Upgrade software
100
Used in the software process to enter the IP address of
the PC
Upgrade language to English
101
When upgrading software, changes or upgrades the
voice prompt language to English
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C
Cisco ATA Specifications
This section describes Cisco ATA specifications:
Note
•
Physical Specifications, page C-1
•
Electrical Specifications, page C-2
•
Environmental Specifications, page C-2
•
Immunity Specifications, page C-2
•
Physical Interfaces, page C-3
•
Ringing Characteristics, page C-3
•
Software Specifications, page C-3
•
SIP Compliance Reference Information, page C-5
The term Cisco ATA refers to both the Cisco ATA 186 and the Cisco ATA 188, unless otherwise stated.
Physical Specifications
Table C-1
Physical Specifications
Description
Specification
Dimensions
1.5 x 6.5 x 5.75 in. (3.8 x 16.5 x 14.6 cm) (H x W x D)
Weight
15 oz (425 g)
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Appendix C
Cisco ATA Specifications
Electrical Specifications
Electrical Specifications
Table C-2
Electrical Specifications
Description
Specification
Power
0.25 to 7.5W (idle to peak)
DC input voltage
+5.0 VDC at 1.5A maximum
Power adaptor
Universal AC/DC
~3.3 x 2.0 x 1.3 in. (~8.5 x 5.0 x 3.2 cm)
~4.8 oz (135 g) for the AC-input external power adaptor
~4 ft (1.2 m) DC cord
6 ft (1.8 m) cord
UL/CUL, CE approved
Class II transformer
Environmental Specifications
Table C-3
Environmental Specifications
Description
Specification
Operating temperature
41 to 104° F (5 to 40° C)
Storage temperature
–4 to 140° F (–20 to 65° C)
Relative humidity
10 to 90% noncondensing, operating, and nonoperating/storage
Immunity Specifications
EN50082-1, including the following:
•
EN61000-3-2, Electromagnetic Compatibility
•
EN61000-3-3, Electromagnetic Compatibility
•
EN61000-4-2, ESD
•
EN61000-4-3, Radiated Immunity
•
EN61000-4-4, Burst Transients
•
EN61000-4-5, Surge
•
EN61000-4-6, Injected RF
•
EN61000-4-11, Dips and Sags
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Cisco ATA Specifications
Physical Interfaces
Physical Interfaces
Table C-4
Physical Interfaces
Description
Specification
Ethernet
Two RJ-45 connectors, IEEE 802.3 10BaseT standard
Analog telephone
Two RJ-11 FXS voice ports
Power
5 VDC power connector
Indicators
Function button with integrated status indicator
Activity LED indicating network activity
Ringing Characteristics
Table C-5
Ringing Characteristics
Description
Specification
Tip/ring interfaces for each RJ-11 FXS port (SLIC)
Ring voltage
40VRMS (typical, balanced ringing only)
Ring frequency
25 Hz
Ring waveform
Trapezoidal with 1.2 to 1.6 crest factor
Ring load
1400 ohm + 40µF
Ringer equivalence number (REN)
Up to 5 REN per RJ-11 FXS port
Loop impedance
Up to 200 ohms (plus 430-ohm maximum telephone
DC resistance)
On-hook/off-hook characteristics
On-hook voltage (tip/ring)
–50V
Off-hook current
27 mA (nominal)
RJ-11 FXS port terminating impedance option
The Cisco ATA186-I1 and Cisco ATA188-I1 provide
600-ohm resistive impedance. The Cisco ATA186-I2
and Cisco ATA188-I2 provide 270 ohm + 750 ohm //
150-nF complex impedance.
Software Specifications
Table C-6
Software Specifications (All Protocols)
Description
Specification
Call progress tones
Configurable for two sets of frequencies and single set of on/off
cadence
Dual-tone multifrequency (DTMF)
DTMF tone detection and generation
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Cisco ATA Specifications
Software Specifications
Table C-6
Software Specifications (All Protocols) (continued)
Description
Specification
Fax
G.711 fax pass-through and G.711 fax mode.
Enhanced fax pass-through is supported on the Cisco ATA.
Success of fax transmissions up to 14.4 kbps depends on
network conditions, and fax modem/fax machine tolerance to
those conditions. The network must have reasonably low
network jitter, network delay, and packet-loss rate.
Line-echo cancellation
Out-of-band DTMF
Configuration
Quality of Service
Security
•
Echo canceller for each port
•
8 ms echo length
•
Nonlinear echo suppression (ERL > 28 dB for frequency
= 300 to 2400 Hz)
•
Convergence time = 250 ms
•
ERLE = 10 to 20 dB
•
Double-talk detection
•
H.245 out-of-band DTMF for H.323
•
RFC 2833 AVT tones for SIP, MGCP, SCCP
•
DHCP (RFC 2131)
•
Web configuration via built-in Web server
•
Touch-tone telephone keypad configuration with voice
prompt
•
Basic boot configuration (RFC 1350 TFTP Profiling)
•
Dial plan configuration
•
Cisco Discovery Protocol
•
Class-of-service (CoS) bit-tagging (802.1P)
•
Type-of-service (ToS) bit-tagging
•
H.235 for H.323
•
RC4 encryption for TFTP configuration files
Voice coder-decoders (codecs)
Note
In simultaneous dual-port operation, the second port is
limited to G.711 when using G.729.
•
G.723.1
•
G.729, G.729A, G.729AB
•
G.723.1
•
G.711A-law
•
G.711µ-law
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Cisco ATA Specifications
SIP Compliance Reference Information
Table C-6
Software Specifications (All Protocols) (continued)
Description
Specification
Voice features
Voice-over-IP (VoIP) protocols
•
Voice activity detection (VAD)
•
Comfort noise generation (CNG)
•
Dynamic jitter buffer (adaptive)
•
H.323 v2
•
SIP (RFC 2543 bis)
•
MGCP 1.0 (RFC 2705)
•
MGCP 1.0/network-based call signalling (NCS) 1.0 profile
•
MGCP 0.1
•
SCCP
SIP Compliance Reference Information
Information on how the Cisco ATA complies with the IETF definition of SIP as described in RFC 2543
is found at the following URL:
http://www-vnt.cisco.com/SPUniv/SIP/documents/CiscoATASIPComplianceRef.pdf
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Cisco ATA Specifications
SIP Compliance Reference Information
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A P P E N D I X
D
SIP Call Flows
This section describes some basic call flows for the Cisco ATA:
Note
•
Supported SIP Request Methods, page D-1
•
Call Flow Scenarios for Successful Calls, page D-2
The term Cisco ATA refers to both the Cisco ATA 186 and the Cisco ATA 188, unless otherwise stated.
Supported SIP Request Methods
The Cisco ATA supports the following SIP request methods:
•
INVITE—Indicates a user or service is being invited to participate in a call session.
•
ACK—Confirms that the client has received a final response to an INVITE request.
•
BYE—Terminates a call and can be sent by either the caller or the callee.
•
CANCEL—Cancels any pending searches but does not terminate a call that has already been
accepted.
•
REGISTER—Registers the address listed in the To header field with a SIP proxy.
•
NOTIFY—Notifies the user of the status of a transfer using REFER. Also used for remote reset.
•
OPTIONS
The following types of responses are used by SIP and generated by the Cisco SIP gateway:
•
SIP 1xx—Informational responses
•
SIP 2xx—Successful responses
•
SIP 3xx—Redirection responses
•
SIP 4xx—Client Failure responses
•
SIP 5xx—Server Failure responses
•
SIP 6xx—Global Failure responses
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Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Call Flow Scenarios for Successful Calls
This section describes call flows for the following scenarios:
•
Cisco ATA-to-SIP Server—Registration without Authentication, page D-2
•
Cisco ATA-to-SIP Server—Registration with Authentication, page D-3
•
Cisco ATA-to-Cisco ATA—Basic SIP to SIP Call without Authentication, page D-6
•
Cisco ATA-to-Cisco ATA—Basic SIP to SIP Call with Authentication, page D-12
Each of the call flows includes a call diagram, action descriptions table, and a sample log file.
Cisco ATA-to-SIP Server—Registration without Authentication
Figure D-1 illustrates the Cisco ATA registering with the SIP server. Authentication is not required for
registration.
The call flow is as follows:
1.
Cisco ATA requests registration.
2.
Registration is completed.
Figure D-1
Cisco ATA-to-SIP Server—Registration without Authentication
Cisco ATA 186
IP network
SIP server
V
1. REGISTER
2. 100 Trying
Table D-1
72117
3. 200 OK
Action Descriptions
Step
Action
Description
Step 1
REGISTER—Cisco ATA to SIP server
Cisco ATA sends a REGISTER message to the SIP server to register
the address in the To header field.
Step 2
100 Trying—SIP Server to Cisco ATA
SIP server returns a 100 Trying message, indicating that the
REGISTER request has been received.
Step 3
200 OK—SIP server to Cisco ATA
SIP server returns a final 200 OK response, confirming that
registration is complete.
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SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-2
Log Listings
1.
REGISTER sip:192.168.2.97 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.194
From:
To:
Call-ID: 88397253@192.168.2.194
CSeq: 1 REGISTER
Contact: ;expires=3600
User-Agent; Cisco ATA v2.10 ata186 (0705a)
Content-Length: 0
2.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.194
Call-ID: 88397253@192.168.2.194
From:
To:
CSeq: 1 REGISTER
Content-Length: 0
3.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.194
Call-ID: 88397253@192.168.2.194
From:
To:
CSeq: 1 REGISTER
Contact: ;expires=“tue, 23 Oct 2001 14:24:57 GMT”
Expires: 3600
Content-Length: 0
Cisco ATA-to-SIP Server—Registration with Authentication
Figure D-2 illustrates the Cisco ATA registering with the SIP server. Authentication is required for
registration.
The call flow is as follows:
1.
Cisco ATA requests registration.
2.
SIP server requests authentication credential.
3.
Authentication is received and registration is completed.
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SIP Call Flows
Call Flow Scenarios for Successful Calls
Figure D-2
Cisco ATA-to-SIP Server—Registration with Authentication
Cisco ATA 186
IP network
SIP server
V
1. REGISTER
2. 100 Trying
3. 407 Proxy Authentication Required
4. REGISTER
6. 200 OK
Table D-3
72118
5. 100 Trying
Action Descriptions
Step
Action
Description
Step 1
REGISTER—Cisco ATA to SIP server
Cisco ATA sends a REGISTER message to the SIP server to
register the address in the To header field.
Step 2
100 Trying—SIP server to Cisco ATA
SIP server returns a 100 trying message, indicating that the
REGISTER request has been received.
Step 3
407 Proxy authentication required— SIP server to
Cisco ATA
SIP server returns a request for authentication.
Step 4
REGISTER—Cisco ATA to SIP server
Cisco ATA attempts to register using its authentication
credential.
Step 5
100 Trying—SIP server to Cisco ATA
SIP server returns a 100 trying message, indicating that the
new REGISTER request has been received.
Step 6
200 OK—SIP server to Cisco ATA
SIP server returns a final 200 OK response, confirming that
the authentication credential has been verified and
registration is complete.
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SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-4
Log Listings
1.
REGISTER sip:192.168.2.81 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.194
From:
To:
Call-ID: 311316842@192.168.2.194
CSeq: 1 REGISTER
Contact: ;expires=3600
User-Agent; Cisco ATA v2.10 ata186 (0705a)
Content-Length: 0
2.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.194
Call-ID: 311316842@192.168.2.194
From:
To:
CSeq: 1 REGISTER
Content-Length: 0
3.
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.194
Call-ID:311316842@192.168.2.194
From:
To:
CSeq: 1 REGISTER
Proxy-Authenticate: DIGEST realm=“CISCO”, nonce=“3bd5e334”
Content-Length: 0
4.
REGISTER sip:192.168.2.81 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.194
From:
To:
Call-ID: 311316842@192.168.2.194
CSeq: 2 REGISTER
Contact: ;expires=3600
User-Agent: Cisco ATA v2.10 ata186 (0705a)
Proxy-Authorization: Digest
username=“9301”,realm=“CISCO”,nonce=“3bd5e334”,uri=“sip:192.168.2.81”,response=“87ac0afeb08222af706
f9e8b5c566ce2”
Content-Length: 0
5.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.194
Call-ID: 311316842@192.168.2.194
From:
To:
CSeq: 2 REGISTER
Content-Length: 0
6.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.194
Call-ID: 311316842@192.168.2.194
From:
To:
CSeq: 2 REGISTER
Contact: ;expires=“tue, 23 Oct 2001 22:37:56 GMT”
Expires: 3600
Content-Length: 0
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Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Cisco ATA-to-Cisco ATA—Basic SIP to SIP Call without Authentication
Figure D-3 illustrates a call from one Cisco ATA to another. Authentication by the SIP server is not
required.
The call flow is as follows:
1.
Call is established between Cisco ATA A and Cisco ATA B.
2.
Call is terminated.
Figure D-3
Cisco ATA-to-Cisco ATA—Basic SIP to SIP Call without Authentication
Cisco ATA 186 A
IP network
SIP server
IP network
V
Cisco ATA 186 B
V
1. INVITE
2. 100 Trying
3. INVITE
4. 100 Trying
5. 180 Ringing
6. 180 Ringing
7. 200 OK
8. 200 OK
9. ACK
10. ACK
2-way voice path
11. BYE
12. 100 Trying
13. BYE
15. 200 OK
72119
14. 200 OK
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
D-6
OL-3410-01
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-5
Action Descriptions
Step
Action
Description
Step 1
INVITE—Cisco ATA A to SIP server
Cisco ATA A sends a call session INVITE request to the SIP
server to pass on to Cisco ATA B.
Step 2
100 Trying—SIP server to Cisco ATA A
SIP server returns a 100 trying message, indicating that the
INVITE request has been received.
Step 3
INVITE—SIP server to Cisco ATA B
SIP server sends the call session INVITE request to Cisco ATA B.
Step 4
100 Trying—Cisco ATA B to SIP server
Cisco ATA B returns a 100 trying message indicating that the
INVITE request has been received.
Step 5
180 Ringing—Cisco ATA B to SIP server
Cisco ATA B sends a 180 ringing response to the SIP server to
pass on to Cisco ATA A.
Step 6
180 Ringing—SIP server to Cisco ATA A
SIP server sends the 180 ringing response to Cisco ATA A.
Step 7
200 OK—Cisco ATA B to SIP server
Cisco ATA B sends a 200 OK message to the SIP server
indicating that a connection has been established.
Step 8
200 OK—SIP server to Cisco ATA A
SIP server passes the 200 OK message to Cisco ATA A.
Step 9
ACK—Cisco ATA A to SIP server
Cisco ATA A sends acknowledgement of the 200 OK response to
the SIP server to pass on to Cisco ATA B.
Step 10
ACK—SIP server to Cisco ATA B
SIP server passes ACK response to Cisco ATA B.
A two-way voice path is established between Cisco ATA A and Cisco ATA B.
Step 11
BYE—Cisco ATA A to SIP server
Cisco ATA A terminates the call session and sends a BYE request
to the SIP server indicating that Cisco ATA A wants to terminate
the call.
Step 12
100 Trying—SIP server to Cisco ATA A
SIP server returns a 100 trying message indicating that the BYE
request has been received.
Step 13
BYE—SIP server to Cisco ATA B
SIP server passes the BYE request to Cisco ATA B.
Step 14
200 OK—Cisco ATA B to SIP server
Cisco ATA B sends a 200 OK message to the SIP server
indicating that Cisco ATA B has received the BYE request.
Step 15
200 OK—SIP server to Cisco ATA A
SIP server passes the BYE request to Cisco ATA A.
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
OL-3410-01
D-7
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-6
Log Listings
1.
INVITE sip:9000@192.168.2.97;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=2819471139
To:
Call-ID: 488337201@192.168.3.175
CSeq: 1 INVITE
Contact:
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=8000 206154 206154 IN IP4 192.168.3.175
s=ATA186 Call
c=IN IP4 192.168.3.175
t=0 0
m=audio 10000 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
2.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.175
Call-ID: 488337201@192.168.3.175
From: ;tag=2819471139
To:
CSeq: 1 INVITE
Content-Length: 0
3.
INVITE sip:9000@192.168.2.194;user=phone SIP/2.0
Record-Route:
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=140fed6e-f61cbd1a-52f223b1-9beb149a-1
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=2819471139
To:
Call-ID: 488337201@192.168.3.175
CSeq: 1 INVITE
Contact:
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=8000 206154 206154 IN IP4 192.168.3.175
s=ATA186 Call
c=IN IP4 192.168.3.175
t=0 0
m=audio 10000 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
D-8
OL-3410-01
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-6
Log Listings (continued)
4.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=140fed6e-f61cbd1a-52f223b1-9beb149a-1
Via: SIP/2.0/UDP 192.168.3.175
Record-Route:
From: ;tag=2819471139
To: ;tag=909616993
Call-ID: 488337201@192.168.3.175
CSeq: 1 INVITE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
5.
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=140fed6e-f61cbd1a-52f223b1-9beb149a-1
Via: SIP/2.0/UDP 192.168.3.175
Record-Route:
From: ;tag=2819471139
To: ;tag=909616993
Call-ID: 488337201@192.168.3.175
CSeq: 1 INVITE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
6.
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.3.175
Record-Route:
From: ;tag=2819471139
To: ;tag=909616993
Call-ID: 488337201@192.168.3.175
CSeq: 1 INVITE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
7.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=140fed6e-f61cbd1a-52f223b1-9beb149a-1
Via: SIP/2.0/UDP 192.168.3.175
Record-Route: ;tag=2819471139
To:
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 199
Content-Type: application/sdp
v=0
o=9000 206275 206275 IN IP4 192.168.2.194
s=ATA186 Call
c=IN IP4 192.168.2.194
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
OL-3410-01
D-9
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-6
Log Listings (continued)
8.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.175
Record-Route: ;tag=2819471139
To:
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 199
Content-Type: application/sdp
v=0
o=9000 206275 206275 IN IP4 192.168.2.194
s=ATA186 Call
c=IN IP4 192.168.2.194
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
9.
ACK sip:9000@192.168.2.97;user=phone SIP/2.0
Route:
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=2819471139
To: ;tag=909616993
Call-ID: 488337201@192.168.3.175
CSeq: 1 ACK
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
10.
ACK sip:9000@192.168.2.194;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=140fed6e-f61cbd1a-52f223b1-9beb149aVia: SIP/2.0/UDP 192.168.3.175
From: ;tag=2819471139
To: ;tag=909616993
Call-ID: 488337201@192.168.3.175
CSeq: 1 ACK
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
11.
BYE sip:9000@192.168.2.97;user=phone SIP/2.0
Route:
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=2819471139
To: ;tag=909616993
Call-ID: 488337201@192.168.3.175
CSeq: 2 BYE
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
12.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 19.168.3.175
Call-ID: 488337201@192.168.3.175
From: ;tag=2819471139
To: ;tag=909616993
CSeq: 2 BYE
Content-Length: 0
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
D-10
OL-3410-01
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-6
Log Listings (continued)
13.
BYE sip:9000@192.168.2.194;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=b499b4be-d7995db7-980cd8af-e5ba35f5-1
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=2819471139
To: ;tag=909616993
Call-ID: 488337201@192.168.3.175
CSeq: 2 BYE
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
14.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.97:5060;branch=b499b4be-d7995db7-980cd8af-e5ba35f5-1
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=2819471139
To: ;tag=909616993
Call-ID: 488337201@192.168.3.175
CSeq: 2 BYE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
15.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=2819471139
To: ;tag=909616993
Call-ID: 488337201@192.168.3.175
CSeq: 2 BYE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
OL-3410-01
D-11
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Cisco ATA-to-Cisco ATA—Basic SIP to SIP Call with Authentication
Figure D-4 illustrates a call from one Cisco ATA to another. Authentication by the SIP server is required.
The call flow is as follows:
1.
Authentication is requested for call initiated by Cisco ATA A.
2.
Call is established between Cisco ATA A and Cisco ATA B.
3.
Call is terminated.
Figure D-4
Cisco ATA-to-Cisco ATA—Basic SIP to SIP Call with Authentication
Cisco ATA 186 A
IP network
SIP server
IP network
V
Cisco ATA 186 B
V
1. INVITE
2. 100 Trying
3. 407 Proxy Authentication Required
4. ACK
5. INVITE
6. 100 Trying
7. INVITE
8.100 Trying
9. 180 Ringing
10. 180 Ringing
11. 200 OK
12. 200 OK
13. ACK
14. ACK
2-way voice path
15. BYE
16. 100 Trying
17. BYE
18. 200 OK
72120
19. 200 OK
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
D-12
OL-3410-01
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-7
Action Descriptions
Step
Action
Description
Step 1
INVITE—Cisco ATA A to SIP server
Cisco ATA A sends a call session INVITE request to the SIP
server to pass on to Cisco ATA B.
Step 2
100 Trying—SIP server to Cisco ATA A
SIP server returns a 100 Trying message, indicating that the
INVITE request has been received.
Step 3
407 Proxy Authentication Required— SIP server SIP server returns a request for authentication to Cisco ATA A.
to Cisco ATA A
Step 4
ACK—Cisco ATA A to SIP server
Cisco ATA A acknowledges the request for authentication.
Step 5
INVITE—Cisco ATA A to SIP server
Cisco ATA A sends a call session INVITE request along with
authentication credential to the SIP server to pass on to
Cisco ATA B.
Step 6
100 Trying—SIP server to Cisco ATA A
SIP server returns a 100 Trying message, indicating that the
INVITE request has been received.
Step 7
INVITE—SIP server to Cisco ATA B
SIP server sends the call session INVITE request to Cisco ATA
B.
Step 8
100 Trying—Cisco ATA B to SIP server
Cisco ATA B returns a 100 trying message indicating that the
INVITE request has been received.
Step 9
180 Ringing—Cisco ATA B to SIP server
Cisco ATA B sends a 180 ringing response to the SIP server to
pass on to Cisco ATA A.
Step 10
180 Ringing—SIP server to Cisco ATA A
SIP server sends the 180 ringing response to Cisco ATA A.
Step 11
200 OK—Cisco ATA B to SIP server
Cisco ATA B sends a 200 OK message to the SIP server
indicating that a connection has been established.
Step 12
200 OK—SIP server to Cisco ATA A
SIP server passes the 200 OK message to Cisco ATA A.
Step 13
ACK—Cisco ATA A to SIP server
Cisco ATA A sends acknowledgment of the 200 OK response to
the SIP server to pass on to Cisco ATA B.
Step 14
ACK—SIP server to Cisco ATA B
SIP server passes ACK response to Cisco ATA B.
A two-way voice path is established between Cisco ATA A and Cisco ATA B.
Step 15
BYE—Cisco ATA A to SIP server
Cisco ATA A terminates the call session and sends a BYE
request to the SIP server indicating that Cisco ATA A wants to
terminate the call.
Step 16
100 Trying—SIP server to Cisco ATA A
SIP server returns a 100 trying message indicating that the BYE
request has been received.
Step 17
BYE—SIP server to Cisco ATA B
SIP server passes the BYE request to Cisco ATA B.
Step 18
200 OK—Cisco ATA B to SIP server
Cisco ATA 186 B sends a 200 OK message to the SIP server
indicating that Cisco ATA 186 B has received the BYE request.
Step 19
200 OK—SIP server to Cisco ATA A
SIP server passes the BYE request to Cisco ATA A.
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
OL-3410-01
D-13
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-8
Log Listings
1.
INVITE sip:9000@192.168.2.81;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=3515135869
To:
Call-ID: 557188650@192.168.3.175
CSeq: 1 INVITE
Contact:
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=8000 177731 177731 IN IP4 192.168.3.175
s=ATA186 Call
c=IN IP4 192.168.3.175
t=0 0
m=audio 10000 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
2.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.175
Call-ID: 557188650@192.168.3.175
From: ;tag=3515135869
To:
CSeq: 1 INVITE
Content-Length: 0
3.
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.3.175
Call-ID: 557188650@192.168.3.175
From: ;tag=3515135869
To:;tag=600eaef7-7c3a549a
CSeq: 1 INVITE
Proxy-Authenticate: DIGEST realm=“CISCO”, nonce=“3bd76584”
Content-Length: 0
4.
ACK sip:9000@192.168.2.81:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=3515135869
To:;tag=600eaef7-7c3a549a
Call-ID: 557188650@192.168.3.175
CSeq: 1 ACK
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
D-14
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Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-8
Log Listings (continued)
5.
INVITE sip:9000@192.168.2.81;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=3515135869
To:
Call-ID: 557188650@192.168.3.175
CSeq: 2 INVITE
Contact:
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Proxy-Authorization: Digest username=“8000”,realm=“CISCO”,nonce=“3bd76584”,
uri=“sip:9000@192.168.2.81”,response=“6e91de67ad976997ffac76f0398ef224”
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=8000 177738 177738 IN IP4 192.168.3.175
s=ATA186 Call
c=IN IP4 192.168.3.175
t=0 0
m=audio 10000 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
6.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.175
Call-ID: 557188650@192.168.3.175
From: ;tag=3515135869
To:
CSeq: 2 INVITE
Content-Length: 0
7.
INVITE sip:9000@192.168.2.194;user=phone SIP/2.0
Record-Route: ;tag=3515135869
To:
Call-ID: 557188650@192.168.3.175
CSeq: 2 INVITE
Contact:
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Proxy-Authorization: Digest username=“8000”,realm=“CISCO”,nonce=“3bd76584”,
uri=“sip:9000@192.168.2.81”,response=“6e91de67ad976997ffac76f0398ef224”
Expires: 300
Content-Length: 253
Content-Type: application/sdp
v=0
o=8000 177738 177738 IN IP4 192.168.3.175
s=ATA186 Call
c=IN IP4 192.168.3.175
t=0 0
m=audio 10000 RTP/AVP 0 18 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
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D-15
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-8
Log Listings (continued)
8.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.81:5060;branch=c0b3510a-819f9d1a-ea43c345-9604747f-1
Via: SIP/2.0/UDP 192.168.3.175
Record-Route:
From: ;tag=3515135869
To: ;tag=100585329
Call-ID: 557188650@192.168.3.175
CSeq: 2 INVITE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
9.
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.81:5060;branch=c0b3510a-819f9d1a-ea43c345-9604747f-1
Via: SIP/2.0/UDP 192.168.3.175
Record-Route:
From: ;tag=3515135869
To: ;tag=100585329
Call-ID: 557188650@192.168.3.175
CSeq: 2 INVITE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
10.
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.3.175
Record-Route:
From: ;tag=3515135869
To: ;tag=100585329
Call-ID: 557188650@192.168.3.175
CSeq: 2 INVITE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
11.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.81:5060;branch=c0b3510a-819f9d1a-ea43c345-9604747f-1
Via: SIP/2.0/UDP 192.168.3.175
Record-Route: ;tag=3515135869
To:
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 199
Content-Type: application/sdp
v=0
o=9000 179263 179263 IN IP4 192.168.2.194
s=ATA186 Call
c=IN IP4 192.168.2.194
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
D-16
OL-3410-01
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-8
Log Listings (continued)
12.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.175
Record-Route: ;tag=3515135869
To:
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 199
Content-Type: application/sdp
v=0
o=9000 179263 179263 IN IP4 192.168.2.194
s=ATA186 Call
c=IN IP4 192.168.2.194
t=0 0
m=audio 10000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
13.
ACK sip:9000@192.168.2.81;user=phone SIP/2.0
Route:
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=3515135869
To: ;tag=100585329
Call-ID: 557188650@192.168.3.175
CSeq: 2 ACK
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
14.
ACK sip:9000@192.168.2.194;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.81:5060;branch=c0b3510a-819f9d1a-ea43c345-9604747f-1
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=3515135869
To: ;tag=100585329
Call-ID: 557188650@192.168.3.175
CSeq: 2 ACK
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
15.
BYE sip:9000@192.168.2.81;user=phone SIP/2.0
Route:
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=3515135869
To: ;tag=100585329
Call-ID: 557188650@192.168.3.175
CSeq: 3 BYE
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
16.
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 19.168.3.175
Call-ID: 557188650@192.168.3.175
From: ;tag=3515135869
To: ;tag=100585329
CSeq: 3 BYE
Content-Length: 0
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
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D-17
Appendix D
SIP Call Flows
Call Flow Scenarios for Successful Calls
Table D-8
Log Listings (continued)
17.
BYE sip:9000@192.168.2.194;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.81:5060;branch=424f3898-9ef87cec-82179ac3-50eeb1d3-1
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=3515135869
To: ;tag=100585329
Call-ID: 557188650@192.168.3.175
CSeq: 3 BYE
User-Agent: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
18.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.81:5060;branch=424f3898-9ef87cec-82179ac3-50eeb1d3-1
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=3515135869
To: ;tag=100585329
Call-ID: 557188650@192.168.3.175
CSeq: 3 BYE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
19.
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.175
From: ;tag=3515135869
To: ;tag=100585329
Call-ID: 557188650@192.168.3.175
CSeq: 3 BYE
Server: Cisco ATA v2.12 ata186 (0928a)
Content-Length: 0
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
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G L O S S A RY
Numerics
10BaseT
10-Mbps baseband Ethernet specification using two pairs of twisted-pair cabling (Categories 3, 4, or
5): one pair for transmitting data and the other for receiving data. 10BASET, which is part of the
IEEE 802.3 specification, has a distance limit of approximately 328 feet (100 meters) per segment.
A
A-law
ITU-T companding standard used in the conversion between analog and digital signals in PCM
systems. A-law is used primarily in European telephone networks and is similar to the North American
µ-law standard. See also companding and µ-law.
AVT tones
Out-of-bound signaling as defined in RFC 2833.
C
category-3 cable
One of five grades of UTP cabling described in the EIA/TIA-586 standard. Category 3 cabling is used
in 10BaseT networks and can transmit data at speeds up to 10 Mbps.
CED tone detection
Called station identification. A three-second, 2100 Hz tone generated by a fax machine answering a
call, which is used in the hand-shaking used to set the call; the response from a called fax machine to
a CNG tone.
CELP
code excited linear prediction compression. Compression algorithm used in low bit-rate voice
encoding. Used in ITU-T Recommendations G.728, G.729, G.723.1.
CLIP
Calling Line Identification Presentation. Shows your identity to callers with Caller ID.
CLIR
Calling Line Identification Restriction. Hides your identity from callers with Caller ID.
CNG
Comfort Noise Generation.
codec
coder decoder. In Voice over IP, Voice over Frame Relay, and Voice over ATM, a DSP software
algorithm used to compress/decompress speech or audio signals.
companding
Contraction derived from the opposite processes of compression and expansion. Part of the PCM
process whereby analog signal values are rounded logically to discrete scale-step values on a nonlinear
scale. The decimal step number then is coded in its binary equivalent prior to transmission. The process
is reversed at the receiving terminal using the same nonlinear scale. Compare with compression and
expansion. See also a-law and µ-law.
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GL-1
Glossary
compression
The running of a data set through an algorithm that reduces the space required to store or the bandwidth
required to transmit the data set. Compare with companding and expansion.
CoS
Class of service. An indication of how an upper-layer protocol requires a lower-layer protocol to treat
its messages. In SNA subarea routing, CoS definitions are used by subarea nodes to determine the
optimal route to establish a given session. A CoS definition comprises a virtual route number and a
transmission priority field.
D
DHCP
Dynamic Host Configuration Protocol. Provides a mechanism for allocating IP addresses dynamically
so that addresses can be reused when hosts no longer need them.
dial peer
An addressable call endpoint. In Voice over IP (VoIP), there are two types of dial peers: POTS and
VoIP.
DNS
Domain Name System. System used on the Internet for translating names of network nodes into
addresses.
DSP
digital signal processor. A DSP segments the voice signal into frames and stores them in voice packets.
DTMF
dual tone multifrequency. Tones generated when a button is pressed on a telephone, primarily used in
the U.S. and Canada.
E
E.164
The international public telecommunications numbering plan. A standard set by the ITU-T which
addresses telephone numbers.
endpoint
A SIP terminal or gateway. An endpoint can call and be called. It generates and/or terminates the
information stream.
expansion
The process of running a compressed data set through an algorithm that restores the data set to its
original size. Compare with companding and compression.
F
firewall
Router or access server, or several routers or access servers, designated as a buffer between any
connected public networks and a private network. A firewall router uses access lists and other methods
to ensure the security of the private network.
FoIP
Fax over IP
FQDN
Fully Qualified Domain (FQDN) format “mydomain.com” or “company.mydomain.com.”
FSK
Frequency shift key.
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Glossary
FXO
Foreign Exchange Office. An FXO interface connects to the public switched telephone network
(PSTN) central office and is the interface offered on a standard telephone. Cisco FXO interface is an
RJ-11 connector that allows an analog connection at the PSTN central office or to a station interface
on a PBX.
FXS
Foreign Exchange Station. An FXS interface connects directly to a standard telephone and supplies
ring, voltage, and dial tone. Cisco's FXS interface is an RJ-11 connector that allows connections to
basic telephone service equipment, keysets, and PBXs.
G
G.711
Describes the 64-kbps PCM voice coding technique. In G.711, encoded voice is already in the correct
format for digital voice delivery in the PSTN or through PBXs. Described in the ITU-T standard in its
G-series recommendations.
G.723.1
Describes a compression technique that can be used for compressing speech or audio signal
components at a very low bit rate as part of the H.324 family of standards. This Codec has two bit
rates associated with it: 5.3 and 6.3 kbps. The higher bit rate is based on ML-MLQ technology and
provides a somewhat higher quality of sound. The lower bit rate is based on CELP and provides
system designers with additional flexibility. Described in the ITU-T standard in its G-series
recommendations.
G.729A
Describes CELP compression where voice is coded into 8-kbps streams. There are two variations of
this standard (G.729 and G.729 Annex A) that differ mainly in computational complexity; both
provide speech quality similar to 32-kbps ADPCM. Described in the ITU-T standard in its G-series
recommendations.
gateway
A gateway allows SIP or H.323 terminals to communicate with terminals configured to other protocols
by converting protocols. A gateway is the point where a circuit-switched call is encoded and
repackaged into IP packets.
H
H.245
An ITU standard that governs H.245 endpoint control.
H.323
H.323 allows dissimilar communication devices to communicate with each other by using a standard
communication protocol. H.323 defines a common set of CODECs, call setup and negotiating
procedures, and basic data transport methods.
I
ICMP
Internet Control Message Protocol
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GL-3
Glossary
IP
Internet Protocol. Network layer protocol in the TCP/IP stack offering a connectionless internetwork
service. IP provides features for addressing, type-of-service specification, fragmentation and
reassembly, and security. Defined in RFC 791.
IVR
Interactive voice response. Term used to describe systems that provide information in the form of
recorded messages over telephone lines in response to user input in the form of spoken words or, more
commonly, DTMF signaling.
L
LDAP
Lightweight DirectoryAccess Protocol
LEC
local exchange carrier.
Location Server
A SIP redirect or proxy server uses a location server to get information about a caller’s location.
Location services are offered by location servers.
M
MGCP
Media Gateway Control Protocol.
MWI
message waiting indication.
µ-law
North American companding standard used in conversion between analog and digital signals in PCM
systems. Similar to the European a-law. See also a-law and companding.
N
NAT
Network Address Translation. Mechanism for reducing the need for globally unique IP addresses.
NAT allows an organization with addresses that are not globally unique to connect to the Internet by
translating those addresses into globally routable address spaces. Also known as Network Address
Translator.
NSE packets
Real-Time Transport Protocol (RTP) digit events are encoded using the Named Signaling Event (NSE)
format specified in RFC 2833, Section 3.0.
NAT Server
Network Address Translation. an Internet standard that enables a local-area network (LAN) to use one
set of IP addresses for internal traffic and a second set of addresses for external traffic.
NTP
Network Time Protocol. Protocol built on top of TCP that assures accurate local time-keeping with
reference to radio and atomic clocks located on the Internet. This protocol is capable of synchronizing
distributed clocks within milliseconds over long time periods.
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Glossary
P
POTS
Plain old telephone service. Basic telephone service supplying standard single-line telephones,
telephone lines, and access to the PSTN.
Proxy Server
An intermediary program that acts as both a server and a client for the purpose of making requests on
behalf of other clients. Requests are serviced internally or by passing them on, possibly after
translation, to other servers. A proxy interprets, and, if necessary, rewrites a request message before
forwarding it.
PSTN
Public switched telephone network.
Q
QoS
Quality of Service. The capability of a network to provide better service to selected network traffic
over various technologies, including Frame Relay, Asynchronous Transfer Mode (ATM), Ethernet and
802.1 networks, SONET, and IP-routed networks that may use any or all of these underlying
technologies. The primary goal of QoS is to provide priority including dedicated bandwidth,
controlled jitter and latency (required by some real-time and interactive traffic), and improved loss
characteristics.
R
Redirect Server
A redirect server is a server that accepts a SIP request, maps the address into zero or more new
addresses, and returns these addresses to the client. It does not initiate its own SIP request nor accept
calls.
Registrar Server
A registrar server is a server that accepts Register requests. A registrar is typically co-located with a
proxy or redirect server and may offer location services.
router
Network layer device that uses one or more metrics to determine the optimal path along which
network traffic should be forwarded. Routers forward packets from one network to another based on
network layer information. Occasionally called a gateway (although this definition of gateway is
becoming increasingly outdated). Compare with gateway.
RTP
Real-Time Transport Protocol. One of the IPv6 protocols. RTP is designed to provide end-to-end
network transport functions for applications transmitting real-time data, such as audio, video, or
simulation data, over multicast or unicast network services. RTP provides services such as payload
type identification, sequence numbering, timestamping, and delivery monitoring to real-time
applications.
S
SCCP
Signaling connection control part.
SDP
Session Definition Protocol. An IETF protocol for the definition of Multimedia Services. SDP
messages can be part of SGCP and MGCP messages.
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GL-5
Glossary
SIP
Session Initiation Protocol. Protocol developed by the IETF MMUSIC Working Group as an
alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999. SIP
equips platforms to signal the setup of voice and multimedia calls over IP networks.
SIP endpoint
A terminal or gateway that acts as a source or sink of Session Initiation Protocol (SIP) voice data. An
endpoint can call or be called, and it generates or terminates the information stream.
SLIC
Subscriber Line Interface Circuit. An integrated circuit (IC) providing central office-like telephone
interface functionality.
SOHO
Small office, home office. Networking solutions and access technologies for offices that are not
directly connected to large corporate networks.
T
TCP
Transmission Control Protocol. Connection-oriented transport layer protocol that provides reliable
full-duplex data transmission. TCP is part of the TCP/IP protocol stack.
TFTP
Trivial File Transfer Protocol. Simplified version of FTP that allows files to be transferred from one
computer to another over a network, usually without the use of client authentication (for example,
username and password).
TN power systems
A TN power system is a power distribution system with one point connected directly to earth (ground).
The exposed conductive parts of the installation are connected to that point by protective earth
conductors.
TOS
Type of service. See CoS.
U
UAC
User agent client. A client application that initiates the SIP request.
UAS
User agent server (or user agent). A server application that contacts the user when a SIP request is
received, and then returns a response on behalf of the user. The response accepts, rejects, or redirects
the request.
UDP
User Datagram Protocol. Connectionless transport layer protocol in the TCP/IP protocol stack. UDP
is a simple protocol that exchanges datagrams without acknowledgments or guaranteed delivery,
requiring that error processing and retransmission be handled by other protocols. UDP is defined in
RFC 768.
user agent
See UAS.
V
VAD
Voice activity detection. When enabled on a voice port or a dial peer, silence is not transmitted over
the network, only audible speech. When VAD is enabled, the sound quality is slightly degraded but
the connection monopolizes much less bandwidth.
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Glossary
voice packet
gateway
Gateway platforms that enable Internet telephony service providers to offer residential and
business-class services for Internet telephony.
VoIP
Voice over IP. The capability to carry normal telephony-style voice over an IP-based Internet with
POTS-like functionality, reliability, and voice quality. VoIP enables a router to carry voice traffic (for
example, telephone calls and faxes) over an IP network. In VoIP, the DSP segments the voice signal
into frames, which then are coupled in groups of two and stored in voice packets. VoIP is a blanket
term, which generally refers to Cisco’s standard-based (for example H.323) approach to IP voice
traffic.
X
XML
eXtensible Markup Language. Designed to enable the use of SGML on the World-Wide Web. XML
allow you to define your own customized markup language.
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GL-7
Glossary
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
GL-8
OL-3410-01
I N D EX
backup-server parameters
Numerics
Bellcore (FSK) method
486 Busy Here response-timeout configuration
802.1Q VLAN ID
3-3, 5-35
5-26
5-13
5-23
Bellcore method for called ID
billable features
5-25
4-4
billable features configuration
binary configuration file creation
A
Action Identifiers
5-23, 5-24
bootload
5-9
process
6-4
advanced audio configuration
3-2
BusyTone
4-4
5-45
5-47
alphanumeric values
3-17
alternate gatekeeper requirement configuration
setting 5-28
alternate NTP IP address configuration
AltGk parameter
5-31
4-10, 5-13
AltNTPIP parameter
ata.txt file
3-9
blind-transfer configuration setting
account information parameters
AlertTone
4-4
CallCmd parameter
Call Command behavior
changing
caller ID
atadefault.cfg configuration file
atadefault.cfg file
atapname.exe Tool
3-12
3-11
3-10
ata unique configuration file
audio and call-waiting tone mixing
5-29
5-23
caller-ID-configuration setting
5-24
5-25
5-25
5-23
CallFeatures parameter
5-23
call forwarding
5-20, 7-2, 7-6
in Sweden
5-20
A-6
in United States
4-2, 5-12
authentication ID
caller ID configuration setting
call features
4-4
audio operating mode
A-2
CallerIdMethod parameter
4-5
AudioMode parameter
6-7
A-1
caller ID format
3-3, 3-12
audio and call waiting tone mixing
audio configuration
5-47
call commands
5-31
3-9
authentication
C
types
5-11
A-5
A-5
call forwarding configuration setting
5-29
calling line identification presentation (CLIP)
A-6
calling line identification restriction
B
definition
backup proxy configuration parameter
5-13
in Sweden
A-6
A-7
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IN-1
Index
in United States
advanced audio
A-7
call-progress tone parameters
Call-progress tones
alphabetical listing of features with related
parameters 4-12
5-42, 5-44
4-8
atadefault.cfg file
call return
in Sweden
audio
A-6
in United States
5-23
call-return configuration setting
5-24
call waiting
hook flash timing
A-2
call-waiting-caller ID configuration setting
call-waiting caller ID setting
5-23, 5-24
5-33
3-18, 3-20
Wed-based
3-21
NAT/PAT translation
4-5
Cisco ATA-specific configuration file
5-7
3-3
Cisco CallManager environment configuration
setting 5-28
Cisco Discovery Protocol (CDP)
3-2
4-5
5-23, 5-24
5-28
AltGk
4-10, 5-13
AltNTPIP
5-31
AudioMode
CallCmd
5-20, 7-2, 7-6
5-47
5-25
CallFeatures
5-23
CfgInterval
3-2, 5-5
ConnectMode
4-3
codec setting
DHCP
4-3
4-5
common parameters
configuration
3-8
conference call
in Sweden
in United States
configuration
5-28, 7-3
5-7
DialPlan
5-38
DNS1IP
5-31
DNS2IP
5-31
EncryptKey
3-11, 5-6, 9-5
FeatureTimer
A-4
A-4
4-8
parameters
CallerIdMethod
5-23
codec negotiation in sending fax
4-7
OutBoundProxy support
3-2, 5-5
4-5
4-8
on-hook delay
Cisco ATA route static configuration
CLIP_CLIR setting
3-7, 3-8
4-7
network timing
5-34
3-9, 3-11
CfgInterval parameter
Comfort
Web-based
NAT gateway
CDP discovery setting
CLIP_CLIR
methods
mixing of call waiting tone and audio
4-5
5-46
cfgfmt.exe tool
4-12
4-5
using TFTP and DHCP servers
call waiting tone and audio mixing
CallWaitTone
5-24
5-23
call-waiting configuration setting
call waiting tone
4-6
features and related parameters
A-5
call-waiting period
3-9, 3-11
4-3
dial plan
A-5
call-waiting caller ID
4-4
cfgfmt.exe tool
codec
in United States
4-2
billable features
4-10
in Sweden
3-11
4-4
authentication
A-6
call return configuration setting
codecs
4-4
Gateway
Gateway2
5-26
5-11
5-11
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Index
GkId
UID1
5-14
GkOrProxy
4-2, 4-10, 5-10
GkTimeToLive
5-14
5-9
UIPassword
5-3, 9-5
UseLoginID
5-11
IPDialPlan
5-38
UseSIP
LBRCodec
5-20
VLAN Setting
5-14
5-35
LoginID0
5-12
refresh interval
LoginID1
5-12
repeat dialing on busy signal
MAXRedirect
MediaPort
4-2, 5-15
required parameters
4-9
4-1
services and related parameters
4-7, 5-17
MediaPort parameter
NATIP
4-3
4-2
4-2, 4-7, 5-16, 5-29
silence suppression
4-12
4-5
SIP proxy server redundancy
4-10
NatServer
5-18
timeout on no answer for call forwarding
NatTimer
5-19
timing
network timing
NPrintF
NumTxFrames
5-22
configuration-complete parameter
OutBoundProxy
4-2, 4-8, 5-17
configuration in a TFTP server environment
configuration-method parameters
5-27
5-4
5-9
configuration update interval
PWD1
5-10
configuring seconds between registration renewal
5-37
ConnectMode parameter
5-5
5-21
Context-Identifiers
SigTimer
5-32
converting MAC address to hexadecimal format
SIPPort parameter
SIPRegInterval
SIPRegOn
StaticIp
5-7
StaticRoute
TftpURL
TimeZone
ToConfig
TraceFlags
5-8
4-7, 5-7
5-5
5-30
5-4
5-36
4-2
creating a text configuration file
datagram precedence
3-8
debugging
5-32
9-4
debugging, preserv.exe program
9-4
debugging diagnostics
TraceFlags
4-5
default call waiting state
5-29
3-12
5-22
default configuration file
UDPTOS
5-32
default values
5-9
3-5
D
TxCodec
UID0
3-10
4-2, 5-15
4-2, 5-16
StaticNetMask
6-3, 6-4
convert unique configuration file to binary format
4-7, 5-17
4-2
5-28, 7-3
RxCodec
SIPPort
3-6
3-5
PWD0
RingOnOffTime
4-2
5-4
configuration in a non-TFTP server environment
5-34
Polarity
3-6
configuration-complete configuration parameter
5-30
OpFlags
4-9
configuration changes taking effect
5-36
NTPIP
4-8
VIA header
4-8
4-10
3-8, 5-2
device hostname
5-35
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Index
always out-of-band setting
DHCP
disabling
5-7
by negotiation
5-21
enabling
5-7
in band setting
5-21
DHCP discovery message
DHCP option 12
DTMF method for caller ID
dummy packets for NAT
3-13, 5-34
DHCP option 66
DHCP options
DtmfMethod configuration settings
5-34
5-35
DHCP option 150
dynamic payload type
3-13
5-7
configuration without using DHCP
3-15
3-14
DHCP server-supplied TFTP configuration filename used
configuration setting
DHCP server usage
5-34
5-28
5-6
3-11
EncryptKey parameter
Dial Plan Commands
5-39
dial plan configuration
4-6
5-6, 9-5
entering alphanumeric values
3-17
environmental specifications
C-2
establishing IP connectivity
3-15
dial plan examples
5-40
Ethernet ports
DialPlan parameter
5-38
example_uprofile.txt file
5-38
dial plan syntax
1-6
3-9
example_uprofile.txt sample configuration file
5-39
example_uprofile.txt text file
5-44
direct IP-to-IP calls
3-8
5-2
example configuration text file
5-2
4-11
disabling access to Web interface
disabling CDP discovery
4-6
F
3-3
disabling use of DHCP server
disabling VLAN encapsulation
3-16
display of hardware information
DNS1IP parameter
5-31
DNS2IP parameter
5-31
factory default reset
3-3
disabling VLAN IP encapsulation
DNS lookup
3-3
3-11
EncryptKey
5-41
4-2
enabling user-specified voice VLAN ID
encryption key
4-6
C-2
enabling SIP registration
encryption
3-13
4-5
dial plan blocking
DialTone
5-18
electrical specifications
if not under control of Cisco ATA administrator
dial plans
5-25
E
DHCP server
DialPlan
5-21
3-14
DHCP parameter
diagnostics
5-21
3-18
failback to primary proxy
3-3
5-13
fax CED tone detection configuration setting
4-7
fax mode
7-1, 7-6
configuration
per-call basis
5-29
7-7
G.711 codec only setting
DNS SRV lookup
4-6
DNS SRV support
4-6
downloading Cisco ATA software from CCO
5-21
5-23, 5-24
no fax tone detection setting
no silence suppression setting
3-6, 3-7
fax mode configuration
5-23, 5-24
5-23, 5-24
7-6, 7-7
DTMF Method
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Index
fax mode configuration settings
5-23
fax-mode configuration settings
5-24
fax pass-through
G.711µ-law
H
h245 tunneling configuration setting
5-28
hardware information display
5-28
G.711 A-law
hook flash time
5-28
fax pass-through mode
fax pass-through mode, enabling
fax Pass-through mode configuration
Hotline/Warmline
7-2
fax pass-through redundancy configuration setting
fax relay, disabling
fax services
4-5
hook flash timing configuration
7-4, 7-5
5-28
HTTP refresh
HTTP reset
7-5
4-7
5-33
hook-flash timing
7-1
5-28
4-5
5-41
5-34
5-34
7-1
fax services, debugging
7-7, 7-9
features and related parameters
FeatureTimer parameter
forgotten password
I
4-12
immunity specifications
5-26
include command
5-3
forward-on-busy setting
3-9
installation
5-23, 5-24
forward-on-no-answer setting
C-2
procedure
5-23, 5-24
2-3
forward-unconditionally setting
5-23, 5-24
IP address static configuration
frames per packet configuration
5-22
IP dial plan configuration
frequently asked questions
IPDialPlan parameter
9-5
5-7
5-38
5-38
Function button
8-1, 9-1
IP precedence (ToS bit) of UDP packets
function button
1-6, 3-3
ITU G.711
FXS ports
5-32
4-5
1-6
L
G
LBRCodec parameter
G.711 silence suppression configuration
G.723.1
G.729
Lightweight Directory Access Protocol (LDAP)
line polarity
5-20
5-20
Gateway2 parameter
Gateway parameter
5-11
5-12
LoginID1 parameter
5-12
D-3, D-5, D-8, D-14
low-bit-rate codec
low-bit-rate codecs
5-14
GkOrProxy parameter
3-9
1-3
5-27
LoginID0 parameter
log listings
5-11
generating binary configuration file
GkId parameter
5-21
5-20
5-21
5-20
4-2, 4-10, 5-10
GkTimeToLive parameter
5-14
Greenwich Mean Time offset
5-30
M
MAC address
3-10
making configuration changes after boot up
3-10
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Index
maximum number of digits in phone number
5-25
on-hook delay configuration
maximum number of times to try redirection
4-2
operating parameters
maximum redirects to another number
MAXRedirect parameter
MediaPort
4-2, 5-15
4-7
MediaPort parameter
5-34
optional feature parameters
5-35
OutBoundProxy parameter
4-2, 4-8, 5-17
outbound proxy server
4-2, 5-17
message-waiting-indication setting
methods supported
5-19
OpFlags parameter
5-15
4-5
P
paid features
5-24
parameters and defaults
NAT/PAT translation
NAT gateway configuration
parameter types
4-7
password
4-7
4-7
NATIP parameter
4-2, 5-16, 5-29
5-3
password for Phone 1 port
5-9
password for Phone 2 port
5-10
5-18
physical interfaces
NatTimer parameter
5-19
physical specifications
Network Address Translation (NAT)
polarity
5-6
network requirements
4-8
4-8
normal start connection mode setting
3-9
5-28
5-25
polarity reversal before and after caller ID signal
5-25
incoming SIP requests
5-17
outgoing SIP requests
5-17
configuration
5-30
5-36
port for incoming SIP requests
5-30
NumTxFrames parameter
5-17
port for debug messages
5-36
NTP IP address configuration
NTPIP parameter
5-27
polarity reversal before and after Caller ID signal
RTP media
4-5
NPrintF parameter
4-11
port
5-33
non-dotted hexadecimal MAC address location
NPrintf
C-1
5-27
Polarity parameter
2-2
network timing configuration
C-3
placing call without using SIP proxy
4-7, 5-18
5-4
5-1
NatServer parameter
network parameters
xii, 5-1
parameters for configuration method
4-7
NAT/PAT translation configuration
no-answer timeout
4-8
1-9
N
network timing
5-17
OutBoundProxy support configuration
5-23, 5-24
mixing of call waiting tone and audio
NATIP
4-5
4-2
primary domain name server IP address
5-22
5-31
Private Line Automatic Ringdown (PLAR)
Progress Indicator configuration setting
O
progress tones
obtaining non-dotted hexadecimal MAC address of Cisco
ATA 3-9
proxy server
on-hook delay
prserv
4-5
on-hook delay before call is disconnected
5-26
5-41
5-28
4-8
1-4
proxy server for all outbound SIP requests
4-2
7-9
PWD0 parameter
5-9
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PWD1 parameter
ringing characteristics
5-10
ring loads
9-5
RingOnOffTime
R
C-3
5-37
RingOnOffTime parameter
received= tag in VIA header
receiving-audio codec settings
redialing timeout
4-2
redundancy
Cisco ATA 188
1-6
7-12, 7-13, 7-15, 7-16, 7-17, 7-19
7-12
RTP media port
refreshing the Cisco ATA
RTP packets
3-22
refresh-interval configuration
refresh procedure
5-28
7-12
RxCodec parameter
REGISTER message configuration
5-21
5-29
4-2
S
1-4
registration-renewal configuration setting
registration renewal increment
registration server
7-12
RTP statistics
3-5
3-10
registrar address
4-2, 5-17
RTP payload type
4-3, 5-5
registrar server
1-6
RTP frames
5-15
5-31
refresh interval
Cisco ATA 186
rtpcatch
1-4
redirects to another number
5-33
RJ-45 LED
5-21
5-26
redirection attempts
redirect server
ring timeout
5-29
safety recommendations
5-28
scaling factor calculation
5-10
D-2
5-25
5-25
5-33
send special character P
Reorder Tone
5-45
services
request high reliability
request high throughput
basic
4-9
setting a password
setting the codec
4-1
resetting the Cisco ATA
3-18
5-26
reviewing default router for Cisco ATA to use
reviewing subnet mask of Cisco ATA
review IP address of Cisco ATA
Ringback Tone
3-17
4-3
5-32
silence suppression
4-5
SIP
5-29
5-37
3-22
1-2, 1-3
advanced features
clients
5-46
ringer cadence pattern configuration
SigTimer parameter
call flow scenarios
3-16
ringback tone configuration setting
3-17
4-12
5-3
signaling image upgrade
3-22
resetting the Cisco ATA to factory defaults
retry timeout
1-10
services and related parameters
5-32
5-31
1-8
supplemental
5-32
5-32
required parameters
5-43
send special character O
reorder delay
repeat dialing on busy signal
3-8
secondary domain name server IP address
D-3
registration without authentication
2-2
sample configuration text file
5-15
registration with authentication
request low delay
5-37
4-3
D-2
1-4
parameters
5-14
request methods
D-1
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Index
servers
1-4
T
SIP call return configuration setting
SIP mode configuration parameter
SIP mode parameter
SIPPort
5-29
TFTP server IP address
4-2
TFTP server IP address or URL specification
5-14
4-7
SIPPort parameter
4-2, 5-17
SIP proxy not used for call
SIP proxy server address
4-2
SIP Proxy Server redundancy
SIPRegInterval parameter
TFTP server name
3-13
TFTP server usage
5-4
TftpURL parameter
4-11
SIP proxy server configuration parameter
SIP registration
3-13
5-10
5-5
three-way calling configuration setting
5-23
three-way-calling configuration setting
5-24
timeout configuration for failback
4-10
SIPRegOn parameter
redialing
5-16
retry
4-2, 5-16
SIP to SIP call with authentication
software download from CCO
5-26
5-26
time-stamping incoming calls
D-12
SIP to SIP call without authentication
time zone configuration
D-6
3-6, 3-7
TimeZone parameter
5-30
C-3
timing configuration
4-8
specifying a preconfigured VLAN ID
3-3
ToConfig parameter
5-4
specifying VLAN CoS bit value (802.1P priority) for UDP
packets 3-3
tones
specifying VLAN CoS bit value for TCP packets
TraceFlags parameter
standard payload type
3-3
5-28
3-2
5-7
5-36
5-23
3-15
5-8
static network-router-probing configuration setting
3-15, 4-7
StaticRoute parameter
stuttering dial tone
5-34
general tips
9-1
installation
9-3
upgrade issues
9-3
TxCodec parameter
5-28
5-22
4-10
supplementary services
A-1
5-22
two-way cut-through configuration setting
5-7
subnet mask static configuration
A-1
5-24
troubleshooting
StaticNetMask parameter
common
5-36
transmitting-audio codec settings
StaticNetMask
cancelling
trace-features configuration
transfer-with-consultation configuration setting
3-15
StaticRoute
4-8
transfer with consultation
statically configuring various IP addresses
5-30
5-30
software specifications (all protocols)
StaticIP parameter
4-10
timeouts
4-2
SIP registration enabled
StaticIP
5-13
timeout on no answer for call forwarding
4-2, 5-15
5-5
5-8
U
UDPTOS parameter
UID0 parameter
5-9
UID1 parameter
5-9
5-32
UIPassword parameter
5-3, 9-5
UIPassword promt configuration setting
5-35
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Index
unconditional call forwarding
unique configuration file
circuit breaker (15A)
4-10
installation
3-9, 3-12
upgrading firmware from TFTP server
8-1
upgrading software
2-2
lightning activity
2-2
main disconnecting device
using executable file
No. 26 AWG
8-2
upgrading software from TFTP server
upgrading the signaling image
UseLoginID parameter
8-1
User agent server (UAS)
1-3
user configurable timeout
product disposal
2-2
Web interface access disabled
1-3
2-2
2-5
web configuration disabling
3-22
5-11
User agent client (UAC)
User ID
2-5
5-34
4-6
4-10
5-9
User Interface (UI) Parameters
UseSIP parameter
5-3
5-14
UseTFTP parameter
5-4
V
VIA header
4-9
VLAN CoS bit value (802.1P priority) for TCP
packets 5-35
VLAN CoS bit value (802.1P priority) for UDP
packets 5-35
VLAN ID
5-35
VLAN ID example
3-4
VLAN IP encapsulation setting
VLAN-related parameters
3-3
VLAN Setting parameter
5-35
VLAN tagging
5-34
3-2
voice configuration menu
3-16
Voice Menu Codes
configuration parameters
information options
software upgrade
B-2
B-1
B-4
W
WAN address of attached router/NAT
4-2, 5-16
warnings
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
78-3410-01
IN-9
Index
Cisco ATA 186 and Cisco ATA 188 Analog Telephone Adaptor Administrator’s Guide (SIP)
IN-10
78-3410-01
Source Exif Data:
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