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Hi,<BR>
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p->do_hangup = SS7_HANGUP_DO_NOTHING;<BR>
do_hangup is not a member of the struct dahdi_pvt. Also where is the definition of the "SS7_HANGUP_DO_NOTHING"<BR></FONT>
<BR><BR>> From: adomjan@tvnet.hu<BR>> To: asterisk-ss7@lists.digium.com<BR>> Date: Thu, 18 Dec 2008 23:22:26 +0100<BR>> Subject: Re: [asterisk-ss7] VÁ: Understanding libss7 code<BR>> <BR>> On Thu, 2008-12-18 at 15:16 -0600, Matthew Fredrickson wrote:<BR>> > Domjan Attila wrote:<BR>> > > should put in chan_dahdi after ISUP_EVENT_CPG and I think have to parse<BR>> > > and pass this busy attribute to chan_dahdi via event_cpg.<BR>> > > How looks like this kind of CPG?<BR>> > <BR>> > I would dare say that it would probably be best to not even explicitly <BR>> > send an REL at that point, just set the SOFTHANGUP flag on the <BR>> > ast_channel so that Asterisk will initiate the hangup at that point.<BR>> > <BR>> but in this case we will send rel with cause code busy (17), but we are<BR>> not busy, I vote in this situation sending rel with normal call<BR>> clearing.<BR>> <BR>> p->owner->hangupcause = e->cpg.cause;<BR>> p->owner->_softhangup |= AST_SOFTHANGUP_DEV;<BR>> p->do_hangup = SS7_HANGUP_DO_NOTHING;<BR>> isup_rel(ss7, p->ss7call, AST_CAUSE_NORMAL_CLEARING);<BR>> <BR>> <BR>> <BR>> > That is how it is done in libpri in a similar scenario, if you look at <BR>> > PRI_EVENT_PROGRESS handling code in chan_dahdi.c. (IIRC)<BR>> > <BR>> > Matthew Fredrickson<BR>> > Digium, Inc.<BR>> > <BR>> > > <BR>> > > On Thu, 2008-12-18 at 11:56 -0800, Rana Dhekial wrote:<BR>> > >> I am not sure whether ITU ANSI standrad calls for it. But in real life<BR>> > >> I am having following probelm.<BR>> > >> <BR>> > >> <BR>> > >> A SIP phone registered with Asterisk calls a Mobile subscriber <BR>> > >> <BR>> > >> Asterisk ---------IAM------------>PSTN ( Mobile subscriber )<BR>> > >> <BR>> > >> Asterisk <--------ACM--------------PSTN <BR>> > >> <BR>> > >> The SIP phone hears the ring back tone<BR>> > >> <BR>> > >> The Mobile subscriber rejects the call by pressing the release button.<BR>> > >> In this part of the world, call does not get forwarded to Mobile<BR>> > >> subscriber's voice mail. Probably incumbennt PLMN does not have voice<BR>> > >> mail service. Instead PSTN sends CPG with user busy.<BR>> > >> <BR>> > >> Asterisk <----CPG ( with user busy)----PSTN<BR>> > >> <BR>> > >> <BR>> > >> The SIP phone keeps hearing the ring back tone for 60-90 seconds and<BR>> > >> finally the PSTN sends RELEASE after 60-90 seconds. <BR>> > >> <BR>> > >> <BR>> > >> Asterisk <------REL-------------------PSTN<BR>> > >> <BR>> > >> Asterisk --------RLC ----------------->PSTN<BR>> > >> <BR>> > >> <BR>> > >> My idea is to cut this 60-90 seconds to 0 by sending REL to PSTN<BR>> > >> immediately after getting the CPG with user busy from PSTN. I have<BR>> > >> tried talking to PSTN to send RELEASE to Asterisk right after they<BR>> > >> send CPG with user busy but has been invain. <BR>> > >> <BR>> > >> So any help with the code will be appreciated.<BR>> > >> <BR>> > >> thanks,<BR>> > >> <BR>> > >><BR>> > >><BR>> > >>> From: adomjan@tvnet.hu<BR>> > >>> To: asterisk-ss7@lists.digium.com<BR>> > >>> Date: Thu, 18 Dec 2008 09:02:08 +0100<BR>> > >>> Subject: [asterisk-ss7] VÁ: Understanding libss7 code<BR>> > >>><BR>> > >>> The code is very readable, I red the all :)<BR>> > >>> where is in the itu/ansi standard that we have to do it?<BR>> > >>><BR>> > >>> -- eredeti üzenet --<BR>> > >>> Tárgy: [asterisk-ss7] Understanding libss7 code<BR>> > >>> Feladó: Rana Dhekial <dhekial@msn.com><BR>> > >>> Dátum: 2008.12.18. 01:27<BR>> > >>><BR>> > >>><BR>> > >>> Hi Matthew,<BR>> > >>><BR>> > >>><BR>> > >>> Can you point me to some documentations to understand the libss7<BR>> > >> source code and how it is linked with Asterisk? I have been struggling<BR>> > >> to modify your code to send ISUP_RELEASE on getting CPG with user busy<BR>> > >> from PSTN but has been successful yet.<BR>> > >>> thanks,<BR>> > >>><BR>> > >>> Rana<BR>> > >>><BR>> > >>><BR>> > >>> _________________________________________________________________<BR>> > >>> Send e-mail anywhere. No map, no compass.<BR>> > >>><BR>> > >> http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_anywhere_122008<BR>> > >>> _______________________________________________<BR>> > >>> --Bandwidth and Colocation Provided by http://www.api-digital.com--<BR>> > >>><BR>> > >>> asterisk-ss7 mailing list<BR>> > >>> To UNSUBSCRIBE or update options visit:<BR>> > >>> http://lists.digium.com/mailman/listinfo/asterisk-ss7<BR>> > >>><BR>> > >>><BR>> > >>> _______________________________________________<BR>> > >>> --Bandwidth and Colocation Provided by http://www.api-digital.com--<BR>> > >>><BR>> > >>> asterisk-ss7 mailing list<BR>> > >>> To UNSUBSCRIBE or update options visit:<BR>> > >>> http://lists.digium.com/mailman/listinfo/asterisk-ss7<BR>> > >><BR>> > >><BR>> > >> ______________________________________________________________________<BR>> > >> Send e-mail faster without improving your typing skills. Get your<BR>> > >> Hotmail® account.<BR>> > >><BR>> > >> ------------------------------------------------------------------------<BR>> > >><BR>> > >> _______________________________________________<BR>> > >> --Bandwidth and Colocation Provided by http://www.api-digital.com--<BR>> > >><BR>> > >> asterisk-ss7 mailing list<BR>> > >> To UNSUBSCRIBE or update options visit:<BR>> > >> http://lists.digium.com/mailman/listinfo/asterisk-ss7<BR>> > <BR>> > <BR>> > _______________________________________________<BR>> > --Bandwidth and Colocation Provided by http://www.api-digital.com--<BR>> > <BR>> > asterisk-ss7 mailing list<BR>> > To UNSUBSCRIBE or update options visit:<BR>> > http://lists.digium.com/mailman/listinfo/asterisk-ss7<BR><BR><br /><hr />Suspicious message? There’s an alert for that. <a href='http://windowslive.com/Explore/hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_broad2_122008' target='_new'>Get your Hotmail® account now.</a></body>
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