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

[Anthony Minessale]

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 <kristjan.ugrin at gmail.com> wrote:

> But what I would like to achieve is something different (quite similar).
> You type in a message like "call 1001 at 10.99.8.20" 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
> <anthony.minessale at gmail.com> 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 <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
> >> >>
> >> >> _______________________________________________
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> >> >>
> >> >
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Porn - the reason you need a new hard drive.
> >>
> >> _______________________________________________
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> >>
> >
> >
> >
>
>
>
> --
>
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-- 
Anthony Minessale II

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