are you doing the trace on the FS box too? it says it's established RTP and bridging.
NO audio is 9.8/10 times a firewall issue. typing in a message is not the right way to call someone on jingle. That's the point. In component mode you add the sip ext to your buddy list and call them the normal way. This has nothing to do with your audio issue though so it's not a big deal. On Mon, Dec 22, 2008 at 9:42 AM, kriko <[email protected]> wrote: > There are absolutely no UDP packets going trough like when doing a call > from gtalk to gtalk. > > You mean this (component mode): > > http://wiki.freeswitch.org/wiki/Dingaling#What_is_Component_.28server_to_server.29_mode.3F > Is there more documentation that this? > > All I would like to do is to initiate a call between SIP telephone and > gtalk user who typed in the message. > > Thank you! > > > On Mon, 22 Dec 2008 16:19:08 +0100, Anthony Minessale > <[email protected]> wrote: > > > 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 <[email protected]> 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/[email protected] > >> &bridge(dingaling/gmail.com/[email protected]) > >> > >> 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 > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> [email protected] > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > > > -- > Porn - the reason you need a new hard drive. > > _______________________________________________ > Freeswitch-users mailing list > [email protected] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[email protected] <msn%[email protected]> GTALK/JABBER/PAYPAL:[email protected]<paypal%[email protected]> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[email protected] <sip%[email protected]> iax:[email protected]/888 googletalk:[email protected]<googletalk%3aconf%[email protected]> pstn:213-799-1400
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