[Freeswitch-users] Call between gtalk and sip - no audio

[Anthony Minessale]

Your log shows rtp streams being allocated.
did you look at at the packets on the wire with a packet capture program?

You are better off using proper jingle and component mode.  What you are
describing sounds like
a workaround to avoid doing it right.



On Mon, Dec 22, 2008 at 8:42 AM, kriko <kristjan.ugrin at gmail.com> wrote:

> I modified mod_dingaling.c so I can intercept google talk chat messages
> via socket - nothing fancy.
> Then I wrote a java app that connects to freeswitch socket and in case of
> a proper message (trigger) it sends a command to freeswitch, in my case:
> api originate sofia/default/1001 at 10.99.8.221
> &bridge(dingaling/gmail.com/my_mail at gmail.com)
>
> Dingaling is logged in as another user which I have added as buddy, chat
> messages go trough, however when a call is started
> between SIP and Gtalk client, we cannot hear each other at all.
> Using freeswitch revision: 10866
>
> Here is the log:
> http://pastebin.com/m1eba2cb8
>
> What can be the problem? First I thought it is because running sip client
> + gtalk and freeswitch on one host, but then I
> moved SIP phone and Gtalk to 2 different workstations, using the third
> only for freeswitch. Also calls from "call" example program
>  from google lib works fine with same setup - something must be problematic
> with freeswitch, however cannot see what.
>
> Thank you!
>
> --
> kriko
>
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-- 
Anthony Minessale II

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