[asterisk-users] Google Voice receiving call problem

[Elliot Murdock]

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);
>>>>      };
>>>> }
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
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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
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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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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>