But what I would like to achieve is something different (quite similar). You type in a message like "call 1001@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@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@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_ser... 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@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@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@10.99.8.221 &bridge(dingaling/gmail.com/my_mail@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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