Hi Mat, I am using asterisk 1.6.1.1 Thanks, Dave George Teletone Inc. -----Original Message----- From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx [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: 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
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