Get following error
menuselect/menuselect --check-deps menuselect.makeoptsGenerating embedded
module rules ... [CC] chan_dahdi.c -> chan_dahdi.ochan_dahdi.c: In function
âss7_linksetâ:chan_dahdi.c:9999: error: âss7_event_cpgâ has no member named
âcauseâmake[1]: *** [chan_dahdi.o] Error 1make: *** [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:
> [email protected]> To: [email protected]> CC: [email protected]>
> 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: [email protected]> > > To:
> [email protected]> > > 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
> <[email protected]>> > > 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: [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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