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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 190 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100205/a195da57/attachment.pgp