if you see them leave FS and never reach dest. It implies a firewall somewhere in between is blocking them.
On Mon, Dec 22, 2008 at 10:19 AM, kriko <[email protected]> wrote: > But what I would like to achieve is something different (quite similar). > You type in a message like "call [email protected]" and you it would call a > SIP buddy with any local number. > > In component mode you need to add a buddy everytime for a different sip > nr.? > Which would mean a lot of numbers if you would like to call more than one > sip nr. in a lan. > > As for the first issue, there are RTP packets traveling on FS, but never > reach destination after they leave our internal network. > Do they have to go outside lan even when the call is placed in a lan > between gtalk and SIP? > Gtalk to gtalk is no problem on same machines... > > > On Mon, 22 Dec 2008 17:02:02 +0100, Anthony Minessale > <[email protected]> wrote: > > > 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 > >> > > > > > > > > > > -- > > _______________________________________________ > 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
_______________________________________________ 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
