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

[asterisk at nicox.org]

Only a feature request, is it possible to make a "cluster" where the 
signalling is made on one server, and 2 or more other servers makes only 
"voice-channels"? or is it planned to do a "cluster"?




On Sat, 17 Nov 2007, Matthew Fredrickson wrote:

> 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
>>>>>>>>> _______________________________________________
>>>>>>>>> --Bandwidth and Colocation Provided by
>>>>>>>>> http://www.api-digital.com--
>>>>>>>>>
>>>>>>>>> asterisk-ss7 mailing list
>>>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>>>
>>>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss
>>>>>>>>> 7
>>>>>>> _______________________________________________
>>>>>>> --Bandwidth and Colocation Provided by
>>>>>>> http://www.api-digital.com--
>>>>>>>
>>>>>>> asterisk-ss7 mailing list
>>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>>
>>>>>>> http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>>>> _______________________________________________
>>>>>> --Bandwidth and Colocation Provided by
>>>>>> http://www.api-digital.com--
>>>>>>
>>>>>> asterisk-ss7 mailing list
>>>>>> To UNSUBSCRIBE or update options visit:
>>>>>>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>>> _______________________________________________
>>>>> --Bandwidth and Colocation Provided by
>>>>> http://www.api-digital.com--
>>>>>
>>>>> asterisk-ss7 mailing list
>>>>> To UNSUBSCRIBE or update options visit:
>>>>>    http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>>
>>>> _______________________________________________
>>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>>
>>>> asterisk-ss7 mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>
>>> asterisk-ss7 mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-ss7
>>
>>
>
>
> -- 
> Matthew Fredrickson
> Software/Firmware Engineer
> Digium, Inc.
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-ss7 mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-ss7
>