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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