are you doing the trace on the FS box too?<br>it says it's established RTP and bridging.<br><br>NO audio is 9.8/10 times a firewall issue.<br><br>typing in a message is not the right way to call someone on jingle.<br>That's the point. In component mode you add the sip ext to your buddy list<br>
and call them the normal way. This has nothing to do with your audio issue though so it's<br>not a big deal.<br><br><div class="gmail_quote">On Mon, Dec 22, 2008 at 9:42 AM, kriko <span dir="ltr"><<a href="mailto:kristjan.ugrin@gmail.com">kristjan.ugrin@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">There are absolutely no UDP packets going trough like when doing a call<br>
from gtalk to gtalk.<br>
<br>
You mean this (component mode):<br>
<a href="http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F" target="_blank">http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F</a><br>
Is there more documentation that this?<br>
<br>
All I would like to do is to initiate a call between SIP telephone and<br>
gtalk user who typed in the message.<br>
<br>
Thank you!<br>
<div><div></div><div class="Wj3C7c"><br>
<br>
On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale<br>
<<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
<br>
> Your log shows rtp streams being allocated.<br>
> did you look at at the packets on the wire with a packet capture program?<br>
><br>
> You are better off using proper jingle and component mode. What you are<br>
> describing sounds like<br>
> a workaround to avoid doing it right.<br>
><br>
><br>
><br>
> On Mon, Dec 22, 2008 at 8:42 AM, kriko <<a href="mailto:kristjan.ugrin@gmail.com">kristjan.ugrin@gmail.com</a>> wrote:<br>
><br>
>> I modified mod_dingaling.c so I can intercept google talk chat messages<br>
>> via socket - nothing fancy.<br>
>> Then I wrote a java app that connects to freeswitch socket and in case<br>
>> of<br>
>> a proper message (trigger) it sends a command to freeswitch, in my case:<br>
>> api originate sofia/default/<a href="mailto:1001@10.99.8.221">1001@10.99.8.221</a><br>
>> &bridge(dingaling/<a href="http://gmail.com/my_mail@gmail.com" target="_blank">gmail.com/my_mail@gmail.com</a>)<br>
>><br>
>> Dingaling is logged in as another user which I have added as buddy, chat<br>
>> messages go trough, however when a call is started<br>
>> between SIP and Gtalk client, we cannot hear each other at all.<br>
>> Using freeswitch revision: 10866<br>
>><br>
>> Here is the log:<br>
>> <a href="http://pastebin.com/m1eba2cb8" target="_blank">http://pastebin.com/m1eba2cb8</a><br>
>><br>
>> What can be the problem? First I thought it is because running sip<br>
>> client<br>
>> + gtalk and freeswitch on one host, but then I<br>
>> moved SIP phone and Gtalk to 2 different workstations, using the third<br>
>> only for freeswitch. Also calls from "call" example program<br>
>> from google lib works fine with same setup - something must be<br>
>> problematic<br>
>> with freeswitch, however cannot see what.<br>
>><br>
>> Thank you!<br>
>><br>
>> --<br>
>> kriko<br>
>><br>
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>><br>
><br>
><br>
><br>
<br>
<br>
<br>
--<br>
</div></div>Porn - the reason you need a new hard drive.<br>
<div><div></div><div class="Wj3C7c"><br>
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