Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <61342> Message-ID: <[email protected]>
I thought I removed L16 codex, but funny as it seems, when you dial into a sipx 4.2.0 system from a verizon cell phone, the codex you have just agreed to is L16. from fs logs: (run fs_cli and dial into the AA, or dial a did from a cell phone and put caller on hold:) 011-07-07 11:49:48.116994 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated mailto:L16@8000hz 1 channels 20ms in system->servers [mysipx], sip-trunking, [sip] I only have pcmu and g722 (and when call first hits fs, it looks like it knows this: 2011-07-07 11:42:56.417796 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[G722:9:8000:20] 2011-07-07 11:42:56.417796 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv payload to 101 2011-07-07 11:42:56.417796 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [telephone-event:101:8000:20]/[G722:9:8000:20] 2011-07-07 11:42:56.417796 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20] 2011-07-07 11:42:56.417796 [DEBUG] sofia_glue.c:3596 Bah HUMBUG! Sticking with mailto:PCMU@8000h but when it gets to the AA, and the AA wants to play: L16: 2011-07-07 11:42:56.475125 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated mailto:L16@8000hz 1 channels 20ms (and warnings on L16 include warnings that L16 can overflow your mtu, cause fragmented packets, and if your firewall blocks fragmented packets, you can lose audio. SOUNDS like one of the problems with one certain version of polycom firmware, and may be unrelated. so, why, after sipx selected PCMU, does the AA outplay L16? -- -- Michael Scheidell, CTO SECNAP Network Security Corp _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
