I am not sure whether ITU ANSI standrad calls for it. But in real life I am
having following probelm.
A SIP phone registered with Asterisk calls a Mobile subscriber
Asterisk ---------IAM------------>PSTN ( Mobile subscriber )
Asterisk <--------ACM--------------PSTN
The SIP phone hears the ring back tone
The Mobile subscriber rejects the call by pressing the release button. In this
part of the world, call does not get forwarded to Mobile subscriber's voice
mail. Probably incumbennt PLMN does not have voice mail service. Instead PSTN
sends CPG with user busy.
Asterisk <----CPG ( with user busy)----PSTN
The SIP phone keeps hearing the ring back tone for 60-90 seconds and finally
the PSTN sends RELEASE after 60-90 seconds.
Asterisk <------REL-------------------PSTN
Asterisk --------RLC ----------------->PSTN
My idea is to cut this 60-90 seconds to 0 by sending REL to PSTN immediately
after getting the CPG with user busy from PSTN. I have tried talking to PSTN
to send RELEASE to Asterisk right after they send CPG with user busy but has
been invain.
So any help with the code will be appreciated.
thanks,
> From: [email protected]> To: [email protected]> Date: Thu, 18 Dec
> 2008 09:02:08 +0100> Subject: [asterisk-ss7] VÁ: Understanding libss7 code> >
> The code is very readable, I red the all :)> where is in the itu/ansi
> standard that we have to do it?> > -- eredeti üzenet --> Tárgy:
> [asterisk-ss7] Understanding libss7 code> Feladó: Rana Dhekial
> <[email protected]>> Dátum: 2008.12.18. 01:27> > > Hi Matthew,> > > Can you
> point me to some documentations to understand the libss7 source code and how
> it is linked with Asterisk? I have been struggling to modify your code to
> send ISUP_RELEASE on getting CPG with user busy from PSTN but has been
> successful yet.> > thanks,> > Rana> > >
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