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

[asterisk at nicox.org]

One of our International Interconnection is:

System uptime: 7 weeks, 1 day, 3 hours, 31 minutes, 51 seconds
Last reload: 4 weeks, 2 hours, 20 minutes, 48 seconds

without any problem.

i had a problem with one of our national IC,
1 signalling link 3 E1's and 2 trunk groups, one of the trunk groups with 
2 E1's gave up and no call was working, but thats a really small problem 
if you have a LCR running.
A restart solved the problem, and since this the link is up for "System 
uptime: 3 days, 2 hours, 14 minutes, 6 seconds"

i will report if this happens again.

(on the machine with 3 E1's there are about 800.000 minutes a month, so i 
think its working as asterisk can *g*


Nico

On Wed, 14 Nov 2007, Anton wrote:

> Nico,
>
> Do you think it's time to give libss7 another try? My last
> test (3-4 month ago) gave terrible results - links did not
> restart automatically , channel was dying accidently and
> unexpectedly and so on.
>
> Anton.
>
> On Wednesday 14 November 2007, asterisk at nicox.org wrote:
>> 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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