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.
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