For some reason a couple weeks ago users began experiencing garbled audio in one direction when dialing out via our VoIP provider. This happened at multiple sites simultaneously. The VoIP provider doesn't think it's their problem. If I switch to another codec so that Asterisk transcodes everything is fine. On conference calls (where Asterisk gets in the middle to relay ulaw to all channels) everything is fine. Calls between sites via ulaw are excellent.
We have plenty of bandwidth, and QoS in place. I've monitored and our QoS box never drops a VoIP packet (4569 UDP). Phones are Polycom IP500s. We're running 1_0_7 stable. Plenty of CPU horsepower (P4 northwood, 2x256 dual channel DDR, 3ware RAID for mirroring) to handle maybe 6 calls simultaneously. This problem occurs even when there's only 1 call on the line. We're now running GSM with decent quality, but I would love to get them back to ulaw. Any ideas? TIA, -Ron _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
