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
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