Hi,
p->do_hangup = SS7_HANGUP_DO_NOTHING;
do_hangup is not a member of the struct dahdi_pvt. Also where is the definition
of the "SS7_HANGUP_DO_NOTHING"
> From: [email protected]> To: [email protected]> Date: Thu, 18 Dec
> 2008 23:22:26 +0100> Subject: Re: [asterisk-ss7] VÁ: Understanding libss7
> code> > On Thu, 2008-12-18 at 15:16 -0600, Matthew Fredrickson wrote:> >
> Domjan Attila wrote:> > > should put in chan_dahdi after ISUP_EVENT_CPG and I
> think have to parse> > > and pass this busy attribute to chan_dahdi via
> event_cpg.> > > How looks like this kind of CPG?> > > > I would dare say that
> it would probably be best to not even explicitly > > send an REL at that
> point, just set the SOFTHANGUP flag on the > > ast_channel so that Asterisk
> will initiate the hangup at that point.> > > but in this case we will send
> rel with cause code busy (17), but we are> not busy, I vote in this situation
> sending rel with normal call> clearing.> > p->owner->hangupcause =
> e->cpg.cause;> p->owner->_softhangup |= AST_SOFTHANGUP_DEV;> p->do_hangup =
> SS7_HANGUP_DO_NOTHING;> isup_rel(ss7, p->ss7call,
> AST_CAUSE_NORMAL_CLEARING);> > > > > That is how it is done in libpri in a
> similar scenario, if you look at > > PRI_EVENT_PROGRESS handling code in
> chan_dahdi.c. (IIRC)> > > > Matthew Fredrickson> > Digium, Inc.> > > > > > >
> > On Thu, 2008-12-18 at 11:56 -0800, Rana Dhekial wrote:> > >> 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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