[Serusers] Fwd: need help implementing sip callback

[Stefan Sayer]

Rogelio Serrano wrote:
> ---------- Forwarded message ----------
> From: Rogelio Serrano <rogelio.serrano at gmail.com>
> Date: Oct 11, 2007 5:18 PM
> Subject: Re: [Serusers] need help implementing sip callback
> To: Atle Samuelsen <clona at cyberhouse.no>
> 
> 
> On 10/11/07, Atle Samuelsen <clona at cyberhouse.no> wrote:
>> * Rogelio Serrano <rogelio.serrano at 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?
> 
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> 
> 
> --
> Lay low and nourish in obscurity
> 
> 

-- 
Stefan Sayer
VoIP Services

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