if you see them leave FS and never reach dest.<br>It implies a firewall somewhere in between is blocking them.<br><br><br><div class="gmail_quote">On Mon, Dec 22, 2008 at 10:19 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;">But what I would like to achieve is something different (quite similar).<br>
You type in a message like "call <a href="mailto:1001@10.99.8.20">1001@10.99.8.20</a>" and you it would call a<br>
SIP buddy with any local number.<br>
<br>
In component mode you need to add a buddy everytime for a different sip<br>
nr.?<br>
Which would mean a lot of numbers if you would like to call more than one<br>
sip nr. in a lan.<br>
<br>
As for the first issue, there are RTP packets traveling on FS, but never<br>
reach destination after they leave our internal network.<br>
Do they have to go outside lan even when the call is placed in a lan<br>
between gtalk and SIP?<br>
Gtalk to gtalk is no problem on same machines...<br>
<br>
<br>
On Mon, 22 Dec 2008 17:02:02 +0100, Anthony Minessale<br>
<div><div></div><div class="Wj3C7c"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>> wrote:<br>
<br>
> 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<br>
> list<br>
> and call them the normal way. This has nothing to do with your audio<br>
> issue<br>
> though so it's<br>
> not a big deal.<br>
><br>
> On Mon, Dec 22, 2008 at 9:42 AM, kriko <<a href="mailto:kristjan.ugrin@gmail.com">kristjan.ugrin@gmail.com</a>> wrote:<br>
><br>
>> 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>
>><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>
>><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<br>
>> program?<br>
>> ><br>
>> > You are better off using proper jingle and component mode. What you<br>
>> 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>><br>
>> wrote:<br>
>> ><br>
>> >> I modified mod_dingaling.c so I can intercept google talk chat<br>
>> messages<br>
>> >> via socket - nothing fancy.<br>
>> >> Then I wrote a java app that connects to freeswitch socket and in<br>
>> case<br>
>> >> of<br>
>> >> a proper message (trigger) it sends a command to freeswitch, in my<br>
>> 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,<br>
>> 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<br>
>> 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>
>> >> _______________________________________________<br>
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>> >><br>
>> ><br>
>> ><br>
>> ><br>
>><br>
>><br>
>><br>
>> --<br>
>> Porn - the reason you need a new hard drive.<br>
>><br>
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>><br>
><br>
><br>
><br>
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<br>
<br>
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<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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