rendered paste bodyast*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'