[asterisk-ss7] ss7 no audio after the call is answered
Dave George
dgeorge at teletoneinc.com
Fri Feb 5 08:52:25 CST 2010
Hi,
We added p->dialing = 0; and the audio is working well now.
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: Friday, February 05, 2010 3:31 AM
To: asterisk-ss7 at lists.digium.com
Subject: Re: [asterisk-ss7] ss7 no audio after the call is answered
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;
On Thu, 2010-02-04 at 19:02 -0500, Dave George wrote:
> To have audio after the call is answered I have to hit a key. See my
> chan_dahhi.conf below. Any suggestions welcome.
>
>
>
>
>
> [trunkgroups]
>
>
>
> [channels]
>
> context=in_dahdi
>
>
>
> switchtype=national
>
>
>
> usecallerid=yes
>
> callwaiting=yes
>
> usecallingpres=yes
>
> callwaitingcallerid=yes
>
> threewaycalling=yes
>
> transfer=yes
>
> canpark=yes
>
> cancallforward=yes
>
> callreturn=yes
>
> echocancel=yes
>
> echocancelwhenbridged=yes
>
>
>
> signalling = ss7
>
>
>
> ss7type = ansi
>
>
>
>
>
> group=1
>
> callgroup=1
>
> pickupgroup=1
>
>
>
> ss7_called_nai=dynamic
>
> ss7_calling_nai=dynamic
>
> ss7_internationalprefix = 00
>
> ss7_nationalprefix = 0
>
>
>
> ; All settings apply to linkset 1
>
> linkset = 1
>
> context=in_dahdi
>
> pointcode = 157
>
> adjpointcode = 163
>
> defaultdpc = 163
>
>
>
> networkindicator=national
>
>
>
> cicbeginswith = 102
>
> channel = 2-24
>
> sigchan = 1
>
>
>
>
>
> group = 2
>
> linkset = 2
>
> context=in_dahdi
>
> pointcode = 157
>
> adjpointcode = 163
>
> defaultdpc = 163
>
>
>
> networkindicator=national
>
>
>
> cicbeginswith = 126
>
> channel = 26-48
>
> sigchan = 25
>
>
>
>
>
>
>
>
>
> Thanks,
>
> Dave George
>
> Teletone Inc.
>
> 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
More information about the asterisk-ss7
mailing list