[asterisk-users] Gtalk/Jabber Issue

Vladimir Mikhelson vlad at mikhelson.com
Thu Feb 10 23:51:28 CST 2011 

William,

I have just noticed that you have several configuration statements
commented out.

I would suggest to un-comment the "status=" in jabber.conf.  I would
also suggest to un-comment the "timeout=", I am not that concerned of
the "keepalive=".

You can reload jabber, no need to restart the Asterisk.

-Vladimir



On 2/10/2011 11:40 PM, Vladimir Mikhelson wrote:
> William,
>
> Have you tried outgoing calls?  What happens there?
>
> Have you restarted the Asterisk after you fixed the typo?
>
> -Vladimir
>
>
>
> On 2/10/2011 10:44 PM, William Stillwell wrote:
>>
>> Yeah, that was a typo, but I fixed, still no dice.
>>
>>  
>>
>> The incoming jabber call doesn’t fire the gtalk connection.
>>
>>  
>>
>>  
>>
>> *From:*asterisk-users-bounces at lists.digium.com
>> [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of
>> *Warren Selby
>> *Sent:* Thursday, February 10, 2011 10:16 PM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] Gtalk/Jabber Issue
>>
>>  
>>
>> You've got connection=jp_jabber defined in one file, and [jb_jabber]
>> defined in the other. 
>>
>> Thanks,
>>
>> --Warren Selby, dCAP
>>
>>
>> On Feb 10, 2011, at 5:55 PM, "William Stillwell"
>> <william at stillwellsoft.com <mailto:william at stillwellsoft.com>> wrote:
>>
>>     Sorry, Asterisk Build 1.6.2.7
>>
>>      
>>
>>     *From:*asterisk-users-bounces at lists.digium.com
>>     <mailto:asterisk-users-bounces at lists.digium.com>
>>     [mailto:asterisk-users-bounces at lists.digium.com] *On Behalf Of
>>     *William Stillwell
>>     *Sent:* Thursday, February 10, 2011 6:50 PM
>>     *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
>>     *Subject:* [asterisk-users] Gtalk/Jabber Issue
>>
>>      
>>
>>     OK, im pulling my hair out, everything looks configured right,
>>     deleted, and started over, etc, etc. but can’t seem to get this
>>     to work
>>
>>      
>>
>>      
>>
>>     Gtalk.conf
>>
>>      
>>
>>     [general]
>>
>>     context=google-in
>>
>>     allowguest=yes
>>
>>     bindaddr=192.168.xxx.xxx
>>
>>     extenip=96.254.xxx.xxx
>>
>>      
>>
>>     [guest]
>>
>>     context=google-in
>>
>>     disallow=all
>>
>>     allow=ulaw
>>
>>     allow=g729
>>
>>     connection=jp_jabber
>>
>>      
>>
>>     jabber.conf
>>
>>      
>>
>>     [general]
>>
>>     debug=yes
>>
>>     ;autoprune=no
>>
>>     autoregister=yes
>>
>>      
>>
>>      
>>
>>     [jb_jabber]
>>
>>     type=client
>>
>>     serverhost=talk.google.com
>>
>>     username=XXXXXXXXX at gmail.com
>>     <mailto:username=XXXXXXXXX at gmail.com>/Talk
>>
>>     secret=XXXXXXX
>>
>>     port=5222
>>
>>     usetls=yes
>>
>>     usesasl=yes
>>
>>     ;status=Available
>>
>>     statusmessage="Connected via Asterisk"
>>
>>     ;timeout=100
>>
>>     ;keepalive=yes
>>
>>      
>>
>>      
>>
>>     Extensions.conf
>>
>>      
>>
>>     [google-in]
>>
>>     exten => s,1,NoOp(Call from GTalk)
>>
>>     exten => s,n,Set(CallerID(Name)="From GoogleTalk")
>>
>>     exten => s,n,Dial(SIP/1000)
>>
>>      
>>
>>     jabber show connected
>>
>>      
>>
>>     Jabber Users and their status:
>>
>>            User: xxxxxx at gmail.com <mailto:xxxxxx at gmail.com>/Talk    
>>     - Connected
>>
>>     ----
>>
>>        Number of users: 1
>>
>>      
>>
>>      
>>
>>     ---- CLI on incoming Call ----
>>
>>      
>>
>>     bannana*CLI>
>>
>>     JABBER: jb_jabber INCOMING: <iq
>>     from="+1*********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2 <mailto:+1*********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
>>     to="******@gmail.com/TalkD876FAA0
>>     <mailto:******@gmail.com/TalkD876FAA0>"
>>     id="jingle:10.218.14.137-17447266:1:03800E94"
>>     type="set"><ses:session type="initiate"
>>     id="SIP1007753261 at 10.218.122.83
>>     <mailto:SIP1007753261 at 10.218.122.83>"
>>     initiator="+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2
>>     <mailto:+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
>>     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>
>>
>>     bannana*CLI>
>>
>>     JABBER: jb_jabber INCOMING: <iq
>>     from="+1********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2
>>     <mailto:+1********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
>>     to="******@gmail.com/TalkD876FAA0
>>     <mailto:******@gmail.com/TalkD876FAA0>"
>>     id="jingle:10.218.14.137-17447266:1:03800EB9"
>>     type="set"><ses:session type="terminate"
>>     id="SIP1007753261 at 10.218.122.83
>>     <mailto:SIP1007753261 at 10.218.122.83>"
>>     initiator="+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2
>>     <mailto:+1*******@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2>"
>>     xmlns:ses="http://www.google.com/session"><pho:call-ended
>>     xmlns:pho="http://www.google.com/session/phone">Call
>>     cancelled</pho:call-ended></ses:session></iq>
>>
>>     bannana*CLI>
>>
>>      
>>
>>      
>>
>>     it doesn’t even try to fire the google-in context ?
>>
>>      
>>
>>     Lastest Version of iksemel Installed, asterisk was rebuild after
>>     installed, asterisk sees both jabber/gtalk commands.
>>
>>      
>>
>>     It just will NOT ring my dialplan.
>>
>>      
>>
>>      
>>
>>      
>>
>>      
>>
>>     --
>>     _____________________________________________________________________
>>     -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>> --
>> _____________________________________________________________________
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>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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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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