[asterisk-users] Cisco 7961G

[Pavel Jezek]

here is my minimalistic .cnf.xml, that works for my 7941

<device>
  <deviceProtocol>SIP</deviceProtocol>
  <sshUserId>admin</sshUserId>
  <sshPassword>***</sshPassword>
  <devicePool>
     <dateTimeSetting>
        <dateTemplate>D-M-Y</dateTemplate>
        <timeZone>Central Europe Standard/Daylight Time</timeZone>
      </dateTimeSetting>
     <callManagerGroup>
        <members>
           <member priority="0">
              <callManager>
                 <ports>
                    <ethernetPhonePort>2000</ethernetPhonePort>
                    <sipPort>5060</sipPort>
                    <securedSipPort>5061</securedSipPort>
                 </ports>
                 <processNodeName>192.168.0.100</processNodeName>
              </callManager>
           </member>
        </members>
     </callManagerGroup>
  </devicePool>

  <sipProfile>
     <sipProxies>
        <registerWithProxy>true</registerWithProxy>
     </sipProxies>
     <enableVad>false</enableVad>
     <preferredCodec>g711a</preferredCodec>
     <natEnabled>0</natEnabled>
     <phoneLabel>Asterisk</phoneLabel>
     <sipLines>
        <line button="1">
           <featureID>9</featureID>
           <featureLabel>SIP 961</featureLabel>
           <proxy>192.168.0.100</proxy>
           <name>961</name>
           <displayName>PJ7961</displayName>
           <authName>961</authName>
           <authPassword>***</authPassword>
           <messagesNumber>8299</messagesNumber>
        </line>
        <line button="2">
           <featureID>21</featureID>
           <featureLabel>Echo test</featureLabel>
           <speedDialNumber>959</speedDialNumber>
        </line>
     </sipLines>
     <dialTemplate>DRdialplan.xml</dialTemplate>
  </sipProfile>

  <commonProfile>
     <phonePassword>***</phonePassword>
  </commonProfile>

  <loadInformation>SIP41.8-2-1S</loadInformation>
  
<versionStamp>1143565489-a3cbf294-7526-4c29-8791-c4fce4ce4c37</versionStamp> 

</device>




Eric Lubow wrote:
> It sounds like you are telling me that it is likely a firmware issue and
> not an Asterisk issue.  Would it be possible for someone to provide me
> with a copy of your SEP<MAC>.cnf.xml file and whatever other files the
> phone uses so I can ensure that its not something else?  Thanks.
>
> Eric
>
> On Fri, 2007-06-01 at 15:07 -0500, Greg Oliver wrote:
>   
>> On Fri, 2007-06-01 at 21:28 +0200, Pavel Jezek wrote:
>>     
>>> we are using 7941 with sip v8.2(2)SR3, it working quite well  ;-)
>>>
>>>
>>> Eric Lubow wrote:
>>>       
>>>> All,
>>>>
>>>>    I am having a lot of trouble with the Cisco 7961G phones.  I have
>>>> managed to get them up and running with Asterisk to the point where I
>>>> can get incoming calls and make outgoing calls.  The problem is when I
>>>> make outgoing calls or extension to extension calls, the calls die after
>>>> 20 seconds.  I have google'd around and came up with little that is of
>>>> help.  The firmware version I am using on the phone is 8.0.4SR1.
>>>>
>>>>    I have tried tcpdumping the conversation and I see that the phone
>>>> doesn't send the SIP/SDP ACK packet back to the remote end.  Sometimes
>>>> it does, but that's a rarity.  There doesn't seem to be any rhyme or
>>>> reason as to when it will send the SIP/SDP ACK.  All I see is the
>>>> following before the phone hangs up at 20 seconds (201 is the phone and
>>>> 205 is the Asterisk Box):
>>>>
>>>> 10.230103 192.168.0.205 -> 192.168.0.201 SIP/SDP Status: 200 OK, with
>>>> session description
>>>>
>>>>    Is there a newer version of the firmware that fixes this?  Is there a
>>>> setting in Asterisk that can fix this?  Any help is greatly appreciated.
>>>> Thanks.
>>>>
>>>> Eric
>>>>
>>>>         
>> Anything older than 8.0.4SR2 is asking for grief.  You cannot even
>> download older from Cisco's website anymore.  Those were their
>> CallManager "transitional" loads from SCCP -> SIP that were riddled with
>> bugs.
>>
>> -Greg
>>
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