[asterisk-ss7] VÁ: RE: VÁ: Understanding libss7 code

Fri Dec 19 02:12:41 CST 2008

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> > >>>> > >>>> > >>> _________________________________________________________________> > >>> Send e-mail anywhere. No map, no compass.> > >>>> > >> http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_anywhere_122008> > >>> _______________________________________________> > >>> --Bandwidth and Colocation Provided by http://www.api-digital.com--> > >>>> > >>> asterisk-ss7 mailing list> > >>> To UNSUBSCRIBE or update options visit:> > >>> http://lists.digium.com/mailman/listinfo/asterisk-ss7> > >>>> > >>>> > >>> _______________________________________________> > >>> --Bandwidth and Colocation Provided by http://www.api-digital.com--> > >>>> > >>> asterisk-ss7 mailing list> > >>> To UNSUBSCRIBE or update options visit:> > >>> http://lists.digium.com/mailman/listinfo/asterisk-ss7> > >>> > >>> > >> ______________________________________________________________________> > >> Send e-mail faster without improving your typing skills. 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