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

[Matthew Fredrickson]

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 :-)

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