[asterisk-users] Gtalk/Jabber Issue

Fri Feb 11 00:32:56 CST 2011

Still no dice..

This make no since.. ive gone over the config a million times now..

The windows gtalk /voice client works just fine.  (incoming and outgoing calls)

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vladimir Mikhelson
Sent: Friday, February 11, 2011 12:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gtalk/Jabber Issue

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> wrote:

Sorry, Asterisk Build 1.6.2.7

From: 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/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/Talk     - Connected

----

   Number of users: 1

---- CLI on incoming Call ----

bannana*CLI> 

JABBER: jb_jabber INCOMING: <iq from="+1*********@voice.google.com/srvres-MTAuMjE4LjE0LjEzNzo5ODU2" to="******@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" initiator="+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" to="******@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" initiator="+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.

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