Hans Witvliet wrote:
Hi all,
Hola,
For one of my inverstigations it looks like i'm back to "square one"
I'm trying to accept an incoming xmpp call and forward it conditionally
to a sip, isdn, or voicemail.
No google is involved as i use a local xmpp server (ejabberd)
I personally am using Openfire but have this exact scenario working
fine, so it does indeed work.
I was experimenting on 1.8.15.1 (with jabber.conf, jingle.conf), but
some suggested me to have a look at asterisk11,so i did...
I downloaded and built 11-beta1.
Edited (according to the asterisk11 wiki-page) extensions.conf,
chan_motif.conf, jingle.conf and restarted.
The files used for res_xmpp and chan_motif are xmpp.conf and motif.conf
respectively.
Same behavior, except for minor details.
As soon as I start, ejabberd tells me that the defined user becomes
online.
From jitsi I can send a text-message, which I see as I enabled "debug"
in motif.conf
(This is actually progress, as in 1.8.15.1 I saw only empty strings
coming along ;-)
But when starting an audio or an AV-call, I only see the xmpp-debug
message (used to be jabber-debug-message).
Within de xmpp-message I see the capabilities (samplerate, codecs,
address, port) from the jitsi-client.
This means your res_xmpp is correctly configured.
Although I made a separate context in the dialplan, it seems never to
get there: hence no answer :(
Eventhough I explicitly point to them in xmpp.conf and jingle.conf....
So my client remains in "connecting" for ever....
chan_motif is what handles this and it sounds like you don't have it
correctly configured. If it's not correctly configured the incoming
messages go unhandled. What are your two configuration files like (minus
passwords)? Only the specific contexts are needed.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
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