[asterisk-users] Google Voice receiving call problem

Vladimir Mikhelson vlad at mikhelson.com
Wed Jun 15 01:20:19 CDT 2011


Elliot,

I do not think Issue # 17993 is related.  As Terry Wilson says on the
Bug Tracker, "Google Voice inbound calls still work, it is just coming
from Google Talk that doesn't."

-Vladimir


On 6/14/2011 5:51 PM, Elliot Murdock wrote:
> Hello,
>
> Seems that it's been spotted and tracked at
> https://issues.asterisk.org/jira/browse/ASTERISK-17993
>
> --Elliot
>
>
> On Tue, Jun 14, 2011 at 7:03 PM, Vladimir Mikhelson <vlad at mikhelson.com> wrote:
>> Elliot,
>>
>> You need to execute "sendDTMF(1) "
>>
>> Articles are available with detailed setup description.
>>
>> -Vladimir
>>
>>
>>
>>
>> On 6/14/2011 1:26 AM, Elliot Murdock wrote:
>>> Hello,
>>>
>>> To help clarify, Jabber is receiving the incoming packets, but
>>> Asterisk does not seem to be associating it with the gtalk
>>> configuration and the call is not routed into any context.  The remote
>>> caller only hears continous ringing.  However, outgoing, gtalk and
>>> jabber work fine.
>>>
>>> What could be the problem?
>>>
>>> Elliot
>>>
>>> On Tue, Jun 14, 2011 at 2:02 AM, Elliot Murdock <murdocke at gmail.com> wrote:
>>>> Hello,
>>>>
>>>> I am using 1.8.4.2 and while outgoing seems to work, incoming still
>>>> does not route calls in to the appropriate context.
>>>>
>>>> Please advise.
>>>>
>>>> Thank you,
>>>> Elliot
>>>>
>>>> On Sat, Apr 16, 2011 at 4:24 PM, William Stillwell
>>>> <william at stillwellsoft.com> wrote:
>>>>> You must have 1.8+ its already been posted the 1.6 didn’t get a backport fix
>>>>> in the jabber protocol.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> From: asterisk-users-bounces at lists.digium.com
>>>>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Leandro
>>>>> Dardini
>>>>> Sent: Saturday, April 16, 2011 3:57 AM
>>>>> To: asterisk-users at lists.digium.com
>>>>> Subject: [asterisk-users] Google Voice receiving call problem
>>>>>
>>>>>
>>>>>
>>>>> Hello,
>>>>> I have a Google Voice phone number and want to connect it to my asterisk box
>>>>> to have calls handled to my SIP account.
>>>>>
>>>>> When I call the number I receive the correct INCOMING request on Jabber
>>>>> portion of asterisk, but the call is not connected to the gtalk part.
>>>>>
>>>>> JABBER: asterisk INCOMING: <iq
>>>>> from="+17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4"
>>>>> to="ldardini at gmail.com/asterisk438D86E0"
>>>>> id="jingle:10.176.108.16-15899749:1:457BCF36" type="set"><ses:session
>>>>> type="initiate" id="SIP784359174 at 10.177.37.1"
>>>>> initiator="+17174695631 at voice.google.com/srvres-MTAuMTc2LjEwOC4xNjo5ODQ4"
>>>>> xmlns:ses="http://www.google.com/session"><pho:description
>>>>> xmlns:pho="http://www.google.com/session/phone"><pho:payload-type id="0"
>>>>> name="PCMU" clockrate="8000"/><pho:payload-type id="101"
>>>>> name="telephone-event"/></pho:description><transport
>>>>> behind-symmetric-nat="false" can-receive-from-symmetric-nat="false"
>>>>> xmlns="http://www.google.com/transport/raw-udp"/><transport
>>>>> xmlns="http://www.google.com/transport/p2p"/></ses:session></iq>
>>>>>
>>>>> No other messages are logged. Where is my mistake?
>>>>>
>>>>> I am using asterisk 1.6.2.7 (Ubuntu packaging) and following there are the
>>>>> relevant files.
>>>>>
>>>>> Thank you
>>>>>
>>>>> Leandro
>>>>>
>>>>> ####### jabber.conf
>>>>>
>>>>> [general]
>>>>> autoregister=yes
>>>>>
>>>>> [asterisk]
>>>>> type=client
>>>>> serverhost=talk.google.com
>>>>> username=ldardini at gmail.com
>>>>> secret=**********
>>>>> priority=1
>>>>> port=5222
>>>>> usetls=yes
>>>>> usesasl=yes
>>>>> buddy=ldardini at gmail.com
>>>>> status=available
>>>>>
>>>>> ####### gtalk.conf
>>>>>
>>>>> [general]
>>>>> context=default
>>>>> bindaddr=0.0.0.0
>>>>> allowguest=yes
>>>>>
>>>>> [guest]
>>>>> disallow=all
>>>>> allow=ulaw
>>>>> context=google-in
>>>>>
>>>>> [ldardini]
>>>>> username=ldardini at gmail.com
>>>>> disallow=all
>>>>> allow=ulaw
>>>>> context=google-in
>>>>> connection=asterisk
>>>>>
>>>>> ######## extension.ael
>>>>>
>>>>> context google-in {
>>>>>     s => {
>>>>>       NoOp( Call from Gtalk );
>>>>>       Dial(SIP/************@************,60,r);
>>>>>      };
>>>>> }
>>>>>
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>>>
>>> --
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>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>



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