Yes, I have IAX2 trunks on this server, I can change them to SIP trunks,<br>but when any CIC in SS7 link gets this "strange" state,<br>even looped calls SS7-SS7 through this CIC have one way audio - incoming,<br>
outgoing audio direction is silent ...<br><br>- Dawid<br><br><div><span class="gmail_quote">2007/11/8, Anton <<a href="mailto:anton.vazir@gmail.com">anton.vazir@gmail.com</a>>:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
Do you use IAX on this server? If so try SIP instead, let<br>know here if so...<br><br>But a some noticed this behavior before, including me, and<br>now I'm not sure what was the reason, IAX or chan_ss7<br><br>On Thursday 08 November 2007, Dawid Kerad wrote:
<br>> Helo,<br>><br>> I have a problem with one way audio using chan_ss7,<br>> this problem occures randomly after a few weeks of work<br>> and many calls, and appears in not transferring audio in<br>> outgoing direction on selected channel.
<br>><br>> When it happens all next calls through this channel has<br>> one way audio, meaningless from which side this call was<br>> initiated. there are no notices in logs, and helps only<br>> restart of chan_ss7 module.
<br>><br>> Does anyone noticed such problems and maybe solved it?<br>> Please send me some advices where to start debugging,<br>> but this problem is very hard to simulate ...<br>> I have asterisk 1.4, chan_ss7
0.9 and Digium card TE410P<br>><br>> - Dawid<br><br><br><br>_______________________________________________<br>--Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--">http://www.api-digital.com--
</a><br><br>asterisk-ss7 mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-ss7">http://lists.digium.com/mailman/listinfo/asterisk-ss7</a><br></blockquote>
</div><br>