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Cisco 7942 DTMF issue on outgoing calls

I just bought a Cisco 7942 phone to connect to my asterisk PBX. I have installed the latest SIP firmware, and put together a SEP<MAC>.cnf.xml. I'm able to make phone calls at this point. When I try to use the keypad to select options on a remote call system, it doesn't work. On most phones, I'm able to set the DTMF mode to be rfc2833 compliant and it works fine, however there seems to be a lack of documentation on the possible settings that can go in the SEP<MAC>.cnf.xml files.

My SEP<MAC>.cnf.xml file and sip.conf file for asterisk are below. Two questions:

1) Where can I go for details on SEP<MAC>.cnf.xml files (and the other needed files). The asterisk and web pages I've seen don't seem to fully understand what all the settings do, so a good document to configure the phone would be great.

2) What are the correct DTMF settings that I should be using for this?

3) Any other suggestions for my configuration? Security wise, sanity wise, etc.

Thanks,

Scott

Relevent sip.conf:

[general]

context=default_context

allowguest=no

allowoverlap=no                

udpbindaddr=0.0.0.0            

tcpenable=yes

tcpbindaddr=0.0.0.0            

transport=udp,tcp                  

srvlookup=yes                  

dynamic_exclude_static = yes

buggymwi=yes

contactpermit=192.168.1.0/24

[laura_office]

type=friend

host=dynamic

context=home

username=laura_office

secret=************

mailbox=100@home

subscribemwi=yes

dtmfmode=info

disallow=all

allow=ulaw

insecure=invite

; deny=0.0.0.0/0.0.0.0

; permit=192.168.1.0/255.255.255.0

SEP<MAC>.cnf.xml

<device>

  <deviceProtocol>SIP</deviceProtocol>

  <sshUserId>username</sshUserId>

  <sshPassword>************</sshPassword>

  <devicePool>

    <dateTimeSetting>

      <dateTemplate>M-D-YA</dateTemplate>

      <timeZone>Eastern Standard/Daylight Time</timeZone>

      <ntps>

        <ntp>

          <name>192.168.1.1</name>

          <ntpMode>Unicast</ntpMode>

        </ntp>

      </ntps>

    </dateTimeSetting>

    <callManagerGroup>

      <members>

        <member priority="0">

          <callManager>

            <ports>

              <ethernetPhonePort>2000</ethernetPhonePort>

              <sipPort>5060</sipPort>

              <securedSipPort>5061</securedSipPort>

            </ports>

            <processNodeName>192.168.1.15</processNodeName>

          </callManager>

        </member>

      </members>

    </callManagerGroup>

  </devicePool>

  <sipProfile>

    <sipProxies>

      <registerWithProxy>true</registerWithProxy>

    </sipProxies>

    <sipCallFeatures>

      <cnfJoinEnabled>true</cnfJoinEnabled>

      <callForwardURI>x--serviceuri-cfwdall</callForwardURI>

      <callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>

      <callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>

      <callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>

      <meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>

      <abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>

      <rfc2543Hold>false</rfc2543Hold>

      <callHoldRingback>2</callHoldRingback>

      <localCfwdEnable>true</localCfwdEnable>

      <semiAttendedTransfer>true</semiAttendedTransfer>

      <anonymousCallBlock>2</anonymousCallBlock>

      <callerIdBlocking>2</callerIdBlocking>

      <dndControl>0</dndControl>

      <remoteCcEnable>true</remoteCcEnable>

    </sipCallFeatures>

    <sipStack>

      <sipInviteRetx>6</sipInviteRetx>

      <sipRetx>10</sipRetx>

      <timerInviteExpires>180</timerInviteExpires>

      <timerRegisterExpires>3600</timerRegisterExpires>

      <timerRegisterDelta>5</timerRegisterDelta>

      <timerKeepAliveExpires>120</timerKeepAliveExpires>

      <timerSubscribeExpires>120</timerSubscribeExpires>

      <timerSubscribeDelta>5</timerSubscribeDelta>

      <timerT1>500</timerT1>

      <timerT2>4000</timerT2>

      <maxRedirects>70</maxRedirects>

      <remotePartyID>false</remotePartyID>

      <userInfo>None</userInfo>

    </sipStack>

    <autoAnswerTimer>1</autoAnswerTimer>

    <autoAnswerAltBehavior>false</autoAnswerAltBehavior>

    <autoAnswerOverride>true</autoAnswerOverride>

    <transferOnhookEnabled>false</transferOnhookEnabled>

    <enableVad>false</enableVad>

    <dtmfAvtPayload>101</dtmfAvtPayload>

    <dtmfDbLevel>3</dtmfDbLevel>

    <dtmfOutofBand>avt</dtmfOutofBand>

    <alwaysUsePrimeLine>false</alwaysUsePrimeLine>

    <alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>

    <kpml>3</kpml>

    <phoneLabel>Laura Office</phoneLabel>

    <callStats>false</callStats>

    <silentPeriodBetweenCallWaitingBursts>3</silentPeriodBetweenCallWaitingBursts>

    <disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>

    <startMediaPort>16384</startMediaPort>

    <stopMediaPort>32766</stopMediaPort>

    <sipLines>

      <line button="1">

        <featureID>9</featureID>

        <featureLabel>Laura Office</featureLabel>

        <proxy>USECALLMANAGER</proxy>

        <port>5060</port>

        <name>laura_office</name>

        <displayName>Laura Office</displayName>

        <autoAnswer>

          <autoAnswerEnabled>2</autoAnswerEnabled>

        </autoAnswer>

        <callWaiting>3</callWaiting>

        <authName>laura_office</authName>

        <authPassword>***************</authPassword>

        <sharedLine>false</sharedLine>

        <messageWaitingLampPolicy>1</messageWaitingLampPolicy>

        <messagesNumber>9999</messagesNumber>

        <ringSettingIdle>4</ringSettingIdle>

        <ringSettingActive>5</ringSettingActive>

        <contact>laura_office</contact>

        <forwardCallInfoDisplay>

          <callerName>true</callerName>

          <callerNumber>false</callerNumber>

          <redirectedNumber>false</redirectedNumber>

          <dialedNumber>true</dialedNumber>

        </forwardCallInfoDisplay>

      </line>

    </sipLines>

    <dialTemplate>dialplan.xml</dialTemplate>

  </sipProfile>

  <vendorConfig>

    <disableSpeaker>false</disableSpeaker>

    <disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>

    <pcPort>1</pcPort>

    <settingsAccess>1</settingsAccess>

    <garp>0</garp>

    <voiceVlanAccess>0</voiceVlanAccess>

    <videoCapability>0</videoCapability>

    <autoSelectLineEnable>0</autoSelectLineEnable>

    <sshAccess>0</sshAccess>

    <sshPort>22</sshPort>

    <webAccess>0</webAccess>

    <spanToPCPort>1</spanToPCPort>

    <loggingDisplay>1</loggingDisplay>

    <loadServer></loadServer>

  </vendorConfig>

  <userLocale>

    <name></name>

    <uid></uid>

    <langCode>en_US</langCode>

    <version></version>

    <winCharSet></winCharSet>

  </userLocale>

  <networkLocale></networkLocale>

  <networkLocaleInfo>

    <name></name>

    <uid></uid>

    <version>1.0.0.0-1</version>

  </networkLocaleInfo>

  <deviceSecurityMode>1</deviceSecurityMode>

  <transportLayerProtocol>1</transportLayerProtocol>

  <loadInformation>SIP42.9-3-1-1S</loadInformation>

</device>

5 REPLIES
Blue

Cisco 7942 DTMF issue on outgoing calls

Hi

SIP firmware are designed and tested to interoperate with Cisco call control only. Although SIP firmware is IETF RFC 3261 compliant, it is not supported  by Cisco TAC or Engineering for use with non-Cisco call control systems.

That is why you will not find any document which will have details of integration of Cisco IP phone with 3rd party PBX.

Regards

Ronak patel

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Regards Ronak Patel Rate all helpful post by clicking stars below the answer.

Cisco 7942 DTMF issue on outgoing calls

There are bugs in the 9.1 firmware which cause problems with DTMF. Make sure you're on a later firmware.

GTG

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Community Member

Cisco 7942 DTMF issue on outgoing calls

Gordon, thanks for the info. I'm on the SIP42.9-3-1-1S firmware. Does this version have DTMF issues? Is there a better firmware version to use (downgrade?)

Cisco 7942 DTMF issue on outgoing calls

9.3.(1) doesn't (AFAIK) have the DTMF issue.

GTG

Please rate all helpful posts.
Community Member

Cisco 7942 DTMF issue on outgoing calls

I wound up just using the SCCP firmware instead of the SIP firmware. Asterisk has a SCCP driver that works well with this firmware.

Scott

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