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

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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] V?: RE: V?: Understanding libss7 code
> > 
> > In my version, in my svn
> > 
> > -- eredeti ?zenet --
> > T?rgy: [asterisk-ss7] RE: V?: Understanding libss7 code
> > Felad?: Rana Dhekial <dhekial at msn.com>
> > D?tum: 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] V?:
> 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] V?: 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 ?zenet --> >
> >>> T?rgy: [asterisk-ss7] Understanding libss7 code> > >>> Felad?:
> Rana Dhekial <dhekial at msn.com>> > >>> D?tum: 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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