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

[Anthony Minessale]

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 <kristjan.ugrin at gmail.com> 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
> <anthony.minessale at gmail.com> 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 <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
> >>
> >> _______________________________________________
> >> Freeswitch-users mailing list
> >> Freeswitch-users at lists.freeswitch.org
> >> 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
> Freeswitch-users at lists.freeswitch.org
> 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:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20081222/6d7db863/attachment-0002.html