Re: [asterisk-users] odd audio problem
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Peter wrote: When I upgraded the 1st asterisk box to 1.4.11. A call comes in, relays to the 2nd asterisk box. The AA answers the call and the audio is good. Once the call is forwarded to an agent. The agent hears everything no problem, but the audio returned to the callers is really bad. It sounds like it is missing 75% of the audio. Do you have enough bandwidth to do the same test with ulaw all the way through? It just seems similar to a problem I've been having. - -- Kind Regards, Matt Riddell Director ___ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News - html) http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.7 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFHAd6pDQNt8rg0Kp4RAt0BAKCwnlNlVwHthYRTRdfzfbzGZ6FHbACfQC/2 NXGPPkrxfu2BNqWXQaIB1r8= =2LoB -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] odd audio problem
I am having an odd audio problem. See setup diagram below. When a call comes in it get routed through the 1st asterisk box (currently running 1.2) through another asterisk box (running 1.4.11). All audio is good. When I upgraded the 1st asterisk box to 1.4.11. A call comes in, relays to the 2nd asterisk box. The AA answers the call and the audio is good. Once the call is forwarded to an agent. The agent hears everything no problem, but the audio returned to the callers is really bad. It sounds like it is missing 75% of the audio. There is no packet loss and 10 ms ping times between the two asterisk boxes. All audio streams are g729 and there is no trans coding anywhere. When I recorded the audio on both asterisk boxes using Mixmonitor, the recorded files sounded good. +-+ | TNT MAX | +-+ | | SIP G729 V ++ | Asterisk Box 1 | ++ | | IAX2 G729 V ++ | Asterisk 1.4 Box 2 | ++ | | IAX2 G729 V Agents Peter ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
