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 >>>>>> >>>>>> _______________________________________________ >>>>>> --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 >
|