[asterisk-users] gtalk - no audio

demuel at thephinix.org demuel at thephinix.org
Fri Jun 22 11:50:47 CDT 2007


Hi Philippe,

In my /etc/asterisk/extensions.conf, I tested both Gtalk/asterisk/<googletalkbuddy> and
Jingle/asterisk/<googletalkbuddy>.

When using Gtalk/asterisk/<googletalkbuddy>, it is consistent with making a call to
googletalk buddy but it just ring once. After the ringing, it just displayed on the voip
phone that it is connected to the googletalk buddy and the timer clock in its lcd starts
incrementing. Though at the googletalk buddy, it just indicates that somebody is calling
and waiting to be answered. But when accepting the call, I still got audio both ways.

Now, when using Jingle/asterisk/<googletalkbuddy>, I got a ring until such that the
googletalk buddy accepts the incoming call then the ringing stops but I could not hear
any audio at all.

FYI, I don't have any problem with making a call from the googletalk client to asterisk.

What is the main distinction between Jingle and Gtalk here? How should I generate the
file streamed to the SIP phone by Asterisk?


Regards,
Demuel

> Hi Demuel,
>
> On 6/22/07, demuel at thephinix.org <demuel at thephinix.org> wrote:
>> Yeah, just the same as the sample configuration by mog. However, if I am using a gtalk
>> application in asterisk to dial googletalk buddy, my voip phone is suddenly connected
>> to
>> the googletalk buddy though at the googletalk client software it is still waiting to
>> be
>> accepted or not accepted. I mean from my voip phone perspective, there is just one
>> ring
>> to make a call to the googletalk buddy unlike in the jingle application wherein there
>> are successive ring before the googletalk buddy accepts the call.
>
> That's strange. I was not able to reproduce this problem, that is,
> when dialing an extension that points to a GoogleTalk client from a
> SIP phone, I *always* have to wait for the GoogleTalk client to accept
> the call.
>
> That's just like if you had Asterisk automatically answer GoogleTalk
> calls. Do you have any file streamed to the SIP phone by Asterisk?
>
> Philippe
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>





More information about the asterisk-users mailing list