[Asterisk-Users] codec translation

[dkwok]

The route of my call is:

gs101-->asterisk-->iaxtel-->asterisk-->gs101

I have 2 g729 from Digium and calls to iaxtel can only be in gsm format. 
The GS101 phones are set to use g729, then 711ulaw.

However when the called GS phone is picked up the connection is 
terminated. These are the console messages:

     -- SIP/1003-f8e1 is ringing
     -- SIP/1003-f8e1 answered IAX2[iaxtel at 69.73.19.178:4569]/6

Error Opening channel:2  not available, see va_g729_init_global(..)Feb 
24 08:47:55 WARNING[1242768320]: codec_g729b.c:179 lintog729_new: No 
available g729 resources for channel 2
Feb 24 08:47:55 WARNING[1242768320]: translate.c:111 
ast_translator_build_path: Failed to build translator step from 6 to 8
Feb 24 08:47:55 WARNING[1242768320]: chan_sip.c:1322 sip_write: Asked to 
transmit frame type 2, while native formats is 256 (read/write = 2/256)
   == Spawn extension (macro-stdexten, s, 4) exited non-zero on 
'IAX2[iaxtel at 69.73.19.178:4569]/6' in macro 'stdexten'
   == Spawn extension (incoming, 1003, 1) exited non-zero on 
'IAX2[iaxtel at 69.73.19.178:4569]/6'
     -- Executing Hangup("IAX2[iaxtel at 69.73.19.178:4569]/6", "") in new 
stack
   == Spawn extension (incoming, h, 1) exited non-zero on 
'IAX2[iaxtel at 69.73.19.178:4569]/6'
     -- Hungup 'IAX2[iaxtel at 69.73.19.178:4569]/6'
     -- Hungup 'IAX2[69.73.19.178:4569]/5'
   == Spawn extension (local, 17001813482, 1) exited non-zero on 
'SIP/1002-4360'
     -- Executing Hangup("SIP/1002-4360", "") in new stack

My questions are:

Although both gs101 are set to use g729 is the actual communication from 
gs to asterisk using g729 and asterisk to iaxtel using gsm and asterisk 
to the called gs using g729.

Do anyone make sense out of the console messages since I have 2 g729 
licence. It should be able to handle 2 g729 channel one receive and one 
send.

-- 
David Kwok

Iaxtel/FWD # 17001813482 ext 1002
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