Dear list, We are running FreeBSD 6.2 with Asterisk 1.4.5 (that has been built with threading in the IAX2 channel removed due to the threading bug in it). Codec in use before today has been G.729, but we were not able to get chanspy to work without crashing Asterisk (persistently). So we switched to G.711, which does not crash asterisk when chanspy is used. We don't wish to stay with G.711 however, since our VoIP service provider only support G.729 and thus we want to run G.729 over the whole network eventually again (all links from the phone, Asterisk (PBX01), wan link, Asterisk (Vrouter) to VoIP SP).
The biggest problem, however, seems to be a timing issue. This is what happens: We record all calls in Asterisk, but when the recording of a poor call (breaking up, bouncing sound, etc) is listened to afterwards, the quality is good. This, in our minds, points to timing issues, not codec translation (which is from G.711 to G.729 and finally to GSM when the call is placed to mobile phones). Strangely enough, the calls to GSM users sound poor when in progress, but the recordings are of acceptable quality. Calls to landline numbers though, sound excellent and recordings are just as good. That seems to indicate that the timing issues are not just confined to our network, but it's may be a combination of our and the SP's network and GSM related (if doesn't make sense though, or does it - that the codec affect the timing?). Having said that, however, I'm must confess that I'm not the technical guru that can qualify this suspicion, and I may be wrong in my analysis of the scenario. How can we address these issues? Are they timing issues, or could it be something else? Are there timing issues in general in Asterisk/FreeBSD if we do not use any zaptel hardware? (We don't have any legacy PBX or Telco integration or hardware in our Asterisk installation, only ethernet) regards -- Roland Giesler Green Tree Systems cc, Stellenbosch, South Africa Mobile: 072-450-2817 http://www.thegreentree.za.net Shop online at http://www.digitalplanet.co.za/?AID=497 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- Asterisk-BSD mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-bsd
