[asterisk-ss7] chan_ss7 one way audio through random CIC

[asterisk at nicox.org]

I used chan_ss7 for months, and i writed a script which is dialing every 
30 minutes and send dtmf-tones in each direction to restart automatically 
if this error happens.

Now i'm using libss7 which in the subversion revision 125 is working much 
more stable than chan_ss7

Please let me know if you find something where the error could happen.

My things i seen was:
 	IAX is not the Problem.
 	SIP is not the Problem.
 	chan_ss7 gets the audio data from asterisk, it seems but i see no 
audio data in zaptel, so i think chan_ss7 -> zaptel is the problem, but i 
could not find where exactly, i'm sorry.


Nico


On Tue, 13 Nov 2007, Anton wrote:

> BTW, I just had a one-way audio situation on one ss7 link
> while using SIP.
>
> On Friday 09 November 2007, Anton wrote:
>> I could speculate that IAX in conjunction with chan_ss7 -
>> leads to that behavior - breaks something or so. - Try
>> SIP... And please let know if behavior reappear.
>>
>> On Friday 09 November 2007, Dawid Kerad wrote:
>>> Yes, I have IAX2 trunks on this server, I can change
>>> them to SIP trunks, but when any CIC in SS7 link gets
>>> this "strange" state, even looped calls SS7-SS7 through
>>> this CIC have one way audio - incoming, outgoing audio
>>> direction is silent ...
>>>
>>> - Dawid
>>>
>>> 2007/11/8, Anton <anton.vazir at gmail.com>:
>>>> Do you use IAX on this server? If so try SIP instead,
>>>> let know here if so...
>>>>
>>>> But a some noticed this behavior before, including
>>>> me, and now I'm not sure what was the reason, IAX or
>>>> chan_ss7
>>>>
>>>> On Thursday 08 November 2007, Dawid Kerad wrote:
>>>>> Helo,
>>>>>
>>>>> I have a problem with one way audio using chan_ss7,
>>>>> this problem occures randomly after a few weeks of
>>>>> work and many calls, and appears in not
>>>>> transferring audio in outgoing direction on
>>>>> selected channel.
>>>>>
>>>>> When it happens all next calls through this channel
>>>>> has one way audio, meaningless from which side this
>>>>> call was initiated. there are no notices in logs,
>>>>> and helps only restart of chan_ss7 module.
>>>>>
>>>>> Does anyone noticed such problems and maybe solved
>>>>> it? Please send me some advices where to start
>>>>> debugging, but this problem is very hard to
>>>>> simulate ... I have asterisk 1.4, chan_ss7 0.9 and
>>>>> Digium card TE410P
>>>>>
>>>>> - Dawid
>>>>
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