[asterisk-ss7] VÁ: RE: VÁ: Understanding libss7 code
Domjan Attila
adomjan at tvnet.hu
Sat Dec 20 18:30:31 CST 2008
I was not a complete code, have to implement cause code in event_cpg in
isup.c, and executue the rel/hangup only if cause code exists in cpg.
On Sat, 2008-12-20 at 16:19 -0800, Rana Dhekial wrote:
> Get following error
>
>
> menuselect/menuselect --check-deps menuselect.makeopts
> Generating embedded module rules ...
> [CC] chan_dahdi.c -> chan_dahdi.o
> chan_dahdi.c: In function âss7_linksetâ:
> chan_dahdi.c:9999: error: âss7_event_cpgâ has no member named âcauseâ
> make[1]: *** [chan_dahdi.o] Error 1
> make: *** [channels] Error 2
>
>
> snippets of chan_dahdi.c
>
> case CPG_EVENT_INBANDINFO:
> {
> struct ast_frame f =
> { AST_FRAME_CONTROL, AST_CONTROL_PROGRESS, };
> ast_debug(1, "Queuing
> frame PROGRESS on CIC %d\n", p->cic);
> dahdi_queue_frame(p,
> &f, linkset);
> p->progress = 1;
> if (p->dsp &&
> p->dsp_features) {
>
> ast_dsp_set_features(p->dsp, p->dsp_features);
>
> p->dsp_features = 0;
> }
> }
> break;
> default:
> ast_debug(1, "Do not handle
> CPG with event type 0x%x\n", e->cpg.event);
> }
> p->owner->hangupcause = e->cpg.cause;
> <-------------------line 9999
> p->owner->_softhangup |=
> AST_SOFTHANGUP_DEV;
> p->do_hangup = SS7_HANGUP_DO_NOTHING;
> isup_rel(ss7, p->ss7call,
> AST_CAUSE_NORMAL_CLEARING);
> ast_mutex_unlock(&p->lock);
> break;
>
>
>
> Do you think I should change line 9999 to
>
> p->owner->hangupcause = e->17;
>
> > Subject: RE: [asterisk-ss7] VÁ: RE: VÁ: Understanding libss7 code
> > From: adomjan at tvnet.hu
> > To: dhekial at msn.com
> > CC: asterisk-ss7 at lists.digium.com
> > Date: Fri, 19 Dec 2008 21:31:16 +0100
> >
> > http://87.242.0.27/repos/trunk/libss7/
> > http://87.242.0.27/repos/trunk/chan_dahdi/
> >
> > chan_dahdi is from 1.6.0
> >
> > On Fri, 2008-12-19 at 10:42 -0800, Rana Dhekial wrote:
> > >
> > > Under which team ?
> > >
> > > http://svn.digium.com/svn/asterisk/team/
> > >
> > >
> > > Also do you know whether your SVN code works with Digium's g.729
> > > software codec? As I could make Digium's sw g729codec working with
> > > Asterisk version 1.6, only if it is 1.6.0.1
> > >
> > > > From: adomjan at tvnet.hu
> > > > To: asterisk-ss7 at lists.digium.com
> > > > Date: Fri, 19 Dec 2008 09:12:41 +0100
> > > > Subject: [asterisk-ss7] VÁ: RE: VÁ: Understanding libss7 code
> > > >
> > > > In my version, in my svn
> > > >
> > > > -- eredeti üzenet --
> > > > Tárgy: [asterisk-ss7] RE: VÁ: Understanding libss7 code
> > > > Feladó: Rana Dhekial <dhekial at msn.com>
> > > > Dátum: 2008.12.19. 02:29
> > > >
> > > >
> > > > 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: adomjan at tvnet.hu> To: asterisk-ss7 at lists.digium.com>
> 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: adomjan at tvnet.hu> > >>> To:
> asterisk-ss7 at lists.digium.com> >
> > > >>> 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 <dhekial at msn.com>> > >>> 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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