Attila, I am using Asterisk 1.6.1.10, the p->dialing is indeed missing from where you are saying - the p->progress is however present. I will test - probably not today unfortunately - and let you know ! Many thanks for your help, J. On Fri, Feb 5, 2010 at 4:30 AM, Attila Domjan <adomjan at tvnet.hu> wrote: > Hi, check the existence of the > > p->dialing = 0; > > in chan_dahdi.c, static void *ss7_linkset(void *data) after the > case ISUP_EVENT_CON: > case ISUP_EVENT_ANM: > > and > case ISUP_EVENT_CPG: > > near p->progress = 1; > > > On Thu, 2010-02-04 at 19:02 -0500, Dave George wrote: > > To have audio after the call is answered I have to hit a key. See my > > chan_dahhi.conf below. Any suggestions welcome. > > > > > > > > > > > > [trunkgroups] > > > > > > > > [channels] > > > > context=in_dahdi > > > > > > > > switchtype=national > > > > > > > > usecallerid=yes > > > > callwaiting=yes > > > > usecallingpres=yes > > > > callwaitingcallerid=yes > > > > threewaycalling=yes > > > > transfer=yes > > > > canpark=yes > > > > cancallforward=yes > > > > callreturn=yes > > > > echocancel=yes > > > > echocancelwhenbridged=yes > > > > > > > > signalling = ss7 > > > > > > > > ss7type = ansi > > > > > > > > > > > > group=1 > > > > callgroup=1 > > > > pickupgroup=1 > > > > > > > > ss7_called_nai=dynamic > > > > ss7_calling_nai=dynamic > > > > ss7_internationalprefix = 00 > > > > ss7_nationalprefix = 0 > > > > > > > > ; All settings apply to linkset 1 > > > > linkset = 1 > > > > context=in_dahdi > > > > pointcode = 157 > > > > adjpointcode = 163 > > > > defaultdpc = 163 > > > > > > > > networkindicator=national > > > > > > > > cicbeginswith = 102 > > > > channel = 2-24 > > > > sigchan = 1 > > > > > > > > > > > > group = 2 > > > > linkset = 2 > > > > context=in_dahdi > > > > pointcode = 157 > > > > adjpointcode = 163 > > > > defaultdpc = 163 > > > > > > > > networkindicator=national > > > > > > > > cicbeginswith = 126 > > > > channel = 26-48 > > > > sigchan = 25 > > > > > > > > > > > > > > > > > > > > Thanks, > > > > Dave George > > > > Teletone Inc. > > > > 561 674 3838 > > > > > > > > > > -- > > _____________________________________________________________________ > > -- 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100205/1bf31d5b/attachment.htm
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