[asterisk-users] Gtalk/Jabber Issue

William Stillwell william at stillwellsoft.com
Fri Feb 11 02:04:00 CST 2011


1:1 nat, I even turned off iptables.. same issue.

 

Guess I will try install wireshark when I get back next week, im done farting with this tonight, when I get back from fort Lauderdale next week I will play with it some more.

 

 

 

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 2:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gtalk/Jabber Issue

 

William,

Another thing to exclude is networking.  Can you verify that nothing blocks the specific traffic on your network?  Any chance of taking the packet trace on your gateway?

-Vladimir




On 2/11/2011 1:18 AM, William Stillwell wrote: 

I don’t’ appear to have an jabber [] OUTGOING packets?

 

I get just 1 incoming packet, and it just sits there, until it rings to voicemail.

 

 

 

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 1:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Gtalk/Jabber Issue

 

William,

I have gone through the similar frustration recently.  Everything works as of early morning yesterday. The big difference, I am on 1.8.2.3.

Have you seen this ticket on the tracker https://issues.asterisk.org/view.php?id=10512 ?   Anything applicable to your case?  The messages are identical to yours on the outgoing call.

-Vladimir




On 2/11/2011 12:32 AM, William Stillwell wrote: 

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.

 

 

 

 

--
_____________________________________________________________________
-- 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

 
--
_____________________________________________________________________
-- 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
 
 
--
_____________________________________________________________________
-- 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
 
 
--
_____________________________________________________________________
-- 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
 
 
--
_____________________________________________________________________
-- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110211/3c2a74ea/attachment.htm>


More information about the asterisk-users mailing list