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 >
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