Hi Atilla, I had this issue during setup and I got this from the list: " 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; " Now that I have traffic on the issue is back on 30% of the calls. Do I have to add it after case ISUP_EVENT_ACM:, or should the above solve the problem. Thanks, Dave George Teletone Inc. 561 674 3838 -----Original Message----- From: asterisk-ss7-bounces@xxxxxxxxxxxxxxxx [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Attila Domjan Sent: Wednesday, April 21, 2010 3:58 AM To: asterisk-ss7 at lists.digium.com Subject: Re: libss7 Audio after DTMF I think it is the bug what I wrote it many times to this list, the missing p->proceeding = 1; p->dialing = 0; after the case ISUP_EVENT_ACM:, in static void *ss7_linkset(void *data) A On Wed, 2010-04-21 at 01:47 -0300, Rafael Prado Rocchi wrote: > What asterisk version are you using? > There's a bug like it in some asterisk versions where you have to > press a key before hearing the audio. > > > > > > -----Original Message----- > > From: asterisk-ss7-bounces at lists.digium.com [mailto:asterisk-ss7- > > bounces at lists.digium.com] On Behalf Of Dave George > > Sent: ter?a-feira, 20 de abril de 2010 10:34 > > To: asterisk-ss7 at lists.digium.com > > Subject: [asterisk-ss7] libss7 Audio after DTMF > > > > 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
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