[asterisk-ss7] No audio with CON message

[Attila Domjan]

Hi
check in chand_dahdi.c

exists or not:
 p->proceeding = 1;
p->dialing = 0;

in ss7_linkset()

after:
case ISUP_EVENT_CON:
case ISUP_EVENT_ANM:

if (chanpos < 0) { /* Never will be true */
                                        ast_log(LOG_WARNING, "ANM/CON on
unconfigured CIC %d PC %d\n", cic, (e->e == ISUP_EVENT_ANM) ? e->anm.opc :
e->con.opc);
                                        isup_free_call(ss7, (e->e ==
ISUP_EVENT_ANM) ? e->anm.call : e->con.call);
                                        break;
                                } else {
                                        p = linkset->pvts[chanpos];
                                        ast_mutex_lock(&p->lock);
                                        p->proceeding = 1;
                                        p->dialing = 0;

Regards,
Attila


2013/2/21 bipin singh <bipinraghuvanshi at gmail.com>

> Hi try without SIP user or IVR calls.
>
>
> On Thu, Feb 14, 2013 at 7:48 PM, Attila Domjan <attila.domjan.hu at gmail.com
> > wrote:
>
>> Just curious. Moved all of our SS7 interconnects to SIP.
>>
>> On Thu, 2013-02-14 at 16:15 +0200, Kaloyan Kovachev wrote:
>> > I am so glad to see you back Domjan
>> >
>> > On Thu, 14 Feb 2013 14:39:37 +0100, Attila Domjan
>> > <attila.domjan.hu at gmail.com> wrote:
>> > > Its an very old and fixed bug.
>> > > We talked about it many times in this list.
>> > >
>> >
>> >
>> > --
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>> >
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>>
>>
>> --
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>
>
>
> --
> BIPIN RAGHUVANSHI
> OPERATION HEAD
> ASTERISK (DEVELOPMENT AND RESEARCH)
> WWW.EHORIZONS.IN
> bipinraghuvanshi at gmail.com
> bipin.singh at ehorizons.in
>
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