testing domjan's sources as well. picked up the same issue above. I'm testing with Asterisk SVN-branch-1.6.0-r258434M with atilia's chan_dahdi.c regards TC On Thu, Apr 22, 2010 at 8:20 PM, Dave George <dgeorge at teletoneinc.com>wrote: > Hi Mat, > > I am using asterisk 1.6.1.1 > > Thanks, > Dave George > Teletone Inc. > > -----Original Message----- > From: asterisk-ss7-bounces at lists.digium.com > [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Matthew > Fredrickson > Sent: Thursday, April 22, 2010 11:16 AM > To: asterisk-ss7 at lists.digium.com > Subject: Re: [asterisk-ss7] libss7 Audio after DTMF > > What version of Asterisk are you using? This looks like the p->dialing > bug that some spoke of earlier, where it was not cleared properly. I > had thought that the fix got committed to all the relevant Asterisk > branches, but it's possible that maybe I missed one. > > Matthew Fredrickson > Digium, Inc. > > Dave George wrote: > > I am using libss7 on an ansi ss7 interconnect. I have two T1s on a > Digium > > TE410P card. On many of the calls I have to hit a key before hearing any > > audio. Any suggestions welcome. Happens about 20 % of the calls. > > > > > > System.conf > > span=1,1,0,esf,b8zs > > span=2,0,0,esf,b8zs > > span=3,0,0,esf,b8zs > > span=4,2,0,esf,b8zs > > mtp2=1 > > bchan=2-24 > > mtp2=73 > > bchan=74-96 > > > > > > > > chan_dahdi.conf > > > > ; All settings apply to linkset 1 > > linkset = 1 > > pointcode = x-x-x > > adjpointcode = x-x-x > > defaultdpc = x-x-x > > > > slc=0 > > sigchan = 1 > > cicbeginswith = 102 > > channel = 2-24 > > > > slc=1 > > sigchan = 73 > > cicbeginswith = 126 > > channel = 74-96 > > > > > > > > Thanks, > > Dave George > > 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 > -- TC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20100423/aa3d1f36/attachment.htm
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