Hi, check, if not exists add it! On Wed, 2010-04-21 at 08:36 -0400, Dave George wrote: > 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 at lists.digium.com [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: [asterisk-ss7] 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 > > > > -- > _____________________________________________________________________ > -- 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/20100421/efae6a57/attachment-0001.pgp
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