12 Oct
2007
12 Oct
'07
7:55 a.m.
Rogelio Serrano wrote:
---------- Forwarded message ----------
From: Rogelio Serrano <rogelio.serrano(a)gmail.com>
Date: Oct 11, 2007 5:18 PM
Subject: Re: [Serusers] need help implementing sip callback
To: Atle Samuelsen <clona(a)cyberhouse.no>
On 10/11/07, Atle Samuelsen <clona(a)cyberhouse.no> wrote:
you need a dialog stateful element,
a media server - use SEMS, asterisk,
freeswitch, callweaver, or the like.
For SEMS, the closes available at the moment is jukecall + di_dial +
xmlrpc2di
http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_jukecall.html
http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_di_dialer.html
http://ftp.iptel.org/pub/sems/doc/current/ModuleDoc_xmlrpc2di.html
see:
http://ftp.iptel.org/pub/sems/doc/current/AppDocExample.html
and especially this thread:
http://lists.iptel.org/pipermail/sems/2007-September/002025.html
Regards
Stefan
* Rogelio Serrano
<rogelio.serrano(a)gmail.com> [071011 07:51]:
i want to ring phone a. then when phone a picks up i would play
recording that phone a has n minutes for this call.
then i ring phone b and when he picks up connect to phone a.
so whats the best way to do this? when the two parties are pstn phones?
the most obvious method is to use an rtp proxy. any pointers?
You could proberbly get some
pointers if you would describe what you
wanted properly. If you want a function like *393939# to make it call
back the last caller tell us... I do not know..
- ATle
how is nat hairpin possible when i dont know which udp port both
parties are going to use?
--
Stefan Sayer
VoIP Services
stefan.sayer(a)iptego.de
www.iptego.de
iptego GmbH
Am Borsigturm 40
13507 Berlin
Germany
Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann
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