He estado trabajando sin problemas con asterisk@home 1.0 con los codecs g726 y g729.

Pero hoy he instalado la version 1.0.5 y me pasa algo muy curioso.

Todos los telefonos que tienen el codec g729 funcionan sin problemas pero los telefonos que tienen el codec g726 no funcionan.

He cambiado el codec de uno de estos telefonos para que funcionen con el g729 y funcionan sin problemas....

Aparentemente el codec g726 esta cargado bien:

asterisk1*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 - 7 3 3 5 3 2 14 29 - 28
gsm 54 - 2 2 4 2 1 13 28 - 27
ulaw 54 6 - 1 4 2 1 13 28 - 27
alaw 54 6 1 - 4 2 1 13 28 - 27
g726 56 8 4 4 - 4 3 15 30 - 29
adpcm 54 6 2 2 4 - 1 13 28 - 27
slin 53 5 1 1 3 1 - 12 27 - 26
lpc10 58 10 6 6 8 6 5 - 32 - 31
g729 58 10 6 6 8 6 5 17 - - 31
speex - - - - - - - - - - -
ilbc 57 9 5 5 7 5 4 16 31 - -

Y esta es el log cuando intento llamar a estos telefonos:

Setting NAT on RTP to 0
Aug 22 17:09:47 DEBUG: Check for res for 351
Aug 22 17:09:47 DEBUG: Call from user '351' is 1 out of 0
Aug 22 17:09:47 DEBUG: build_route: Contact hop: <sip:351@192.168.30.54:5060>
Aug 22 17:09:47 VERBOSE: -- Executing Dial("SIP/351-85b7", "SIP/410|60|Ttr") in new stack
Aug 22 17:09:47 DEBUG: SIMPLE DIAL (NO URL)
Aug 22 17:09:47 DEBUG: Setting NAT on RTP to 0
Aug 22 17:09:47 DEBUG: Outgoing Call for 410
Aug 22 17:09:47 DEBUG: Call from user '410' is 1 out of 0
Aug 22 17:09:47 VERBOSE: -- Called 410
Aug 22 17:09:47 DEBUG: (Provisional) Stopping retransmission (but retaining packet) on '3ed1f29257be4c5f5933e4586471a091@192.168.29.2' Request 102: Found
Aug 22 17:09:47 DEBUG: Acked pending invite 102
Aug 22 17:09:47 DEBUG: Stopping retransmission on '3ed1f29257be4c5f5933e4586471a091@192.168.29.2' of Request 102: Found
Aug 22 17:09:47 VERBOSE: -- Got SIP response 488 "Not Acceptable Here" back from 192.168.2.21
Aug 22 17:09:47 DEBUG: update_user_counter(410) - decrement outUse counter
Aug 22 17:09:47 VERBOSE: == No one is available to answer at this time
Aug 22 17:09:47 DEBUG: Exiting with DIALSTATUS=NOANSWER.
Aug 22 17:09:47 VERBOSE: -- Executing Hangup("SIP/351-85b7", "") in new stack
Aug 22 17:09:47 VERBOSE: == Spawn extension (varios, 410, 2) exited non-zero on 'SIP/351-85b7'

Pero si cambio el codec a g729 funciona sin problemas...

A cambiado la forma de referirse al codec g726 en asterisk 1.0.9 ?

Gracias..