---------- Forwarded message ---------- From: Rogelio Serrano rogelio.serrano@gmail.com Date: Oct 11, 2007 5:18 PM Subject: Re: [Serusers] need help implementing sip callback To: Atle Samuelsen clona@cyberhouse.no
On 10/11/07, Atle Samuelsen clona@cyberhouse.no wrote:
- Rogelio Serrano rogelio.serrano@gmail.com [071011 07:51]:
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
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.
how is nat hairpin possible when i dont know which udp port both parties are going to use?
-- Lay low and nourish in obscurity _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Lay low and nourish in obscurity
Rogelio Serrano wrote:
---------- Forwarded message ---------- From: Rogelio Serrano rogelio.serrano@gmail.com Date: Oct 11, 2007 5:18 PM Subject: Re: [Serusers] need help implementing sip callback To: Atle Samuelsen clona@cyberhouse.no
On 10/11/07, Atle Samuelsen clona@cyberhouse.no wrote:
- Rogelio Serrano rogelio.serrano@gmail.com [071011 07:51]:
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
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.
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
how is nat hairpin possible when i dont know which udp port both parties are going to use?
-- Lay low and nourish in obscurity _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Lay low and nourish in obscurity
-
Rogelio Serrano -
Stefan Sayer