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

[Matthew Fredrickson]

Matthew Fredrickson wrote:
> asterisk at nicox.org wrote:
>> 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*
> 
> *Jumps for Joy*
> 
> That is awesome!  Thanks for sharing that.  It's a always good to hear 
> positive feedback :-)

Oh yeah, negative feedback is good too.  Let me know if there's anything 
I can fix as well.

Matthew Fredrickson

> 
> Matthew Fredrickson
> 
>>
>> 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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> 


-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.