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-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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