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 >>>>>>> _______________________________________________ >>>>>>> --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.
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