[asterisk-ss7] libss7 Audio after DTMF
Dave George
dgeorge at teletoneinc.com
Wed Apr 21 07:36:08 CDT 2010
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: terça-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
More information about the asterisk-ss7
mailing list