ast*CLI> show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - - - - - - - - - - - gsm - - 4 4 11 4 3 13 38 - 52 ulaw - 10 - 1 9 2 1 11 36 - 50 alaw - 10 1 - 9 2 1 11 36 - 50 g726 - 16 8 8 - 8 7 17 42 - 56 adpcm - 10 2 2 9 - 1 11 36 - 50 slin - 9 1 1 8 1 - 10 35 - 49 lpc10 - 17 9 9 16 9 8 - 43 - 57 g729 - 16 8 8 15 8 7 17 - - 56 speex - - - - - - - - - - - ilbc - 18 10 10 17 10 9 19 44 - - ast*CLI> show modules output codec_g729a.so Annex A/B (floating point) G.729 Codec ( 0 when i m calling from both lines of linksys pap2. one line works and other gives me error on console of asterisk. here is the output. Nov 9 16:35:03 NOTICE[1589]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Nov 9 16:38:03 NOTICE[1589]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! -- Executing Dial("SIP/ast.d2visp.com-0814b270", "SIP/18158340674@isphone|30") in new stack -- Called 18158340674@isphone -- SIP/isphone-ee39 is making progress passing it to SIP/ast.d2visp.com-0814b270 -- SIP/isphone-ee39 is ringing Nov 9 16:40:13 NOTICE[1589]: chan_sip.c:2792 process_sdp: No compatible codecs! Nov 9 16:40:15 NOTICE[1589]: rtp.c:298 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible -- SIP/isphone-ee39 answered SIP/ast.d2visp.com-0814b270 -- Attempting native bridge of SIP/ast.d2visp.com-0814b270 and SIP/isphone-ee39 == Spawn extension (ser-in, 18158340674, 1) exited non-zero on 'SIP/ast.d2visp.com-0814b270'