[asterisk-ss7] libss7 Audio after DTMF

[TCB]

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
> >
> >
> >
> >
>
>
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-- 
TC
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