[asterisk-users] Problems with Codecs in Asterisk

[Chan Kwang Mien]

Hi,

My test-setup is as follows :

sip1 <--> Asterisk <--> sip2
              ^
              |-------> sip3    

In sip.conf, 

[sip1]
type=friend
host=dynamic
secret=pass
disallow=all
allow=g729
allow=ulaw

[sip2]
type=friend
host=dynamic
secret=pass
disallow=all
allow=g729

[sip3]
type=friend
host=dynamic
secret=pass
disallow=all
allow=ulaw


sip1 supports g.729 and g.711u only
sip2 supports g.729 only
sip3 supports g.711u only

sip1 is able to establish a call to sip2.
However, I have problem establishing a call from sip1 to sip3. sip3
rings but when I answered it, it hanged up.

The Logs are :

    -- Executing Dial("SIP/2006-389a", "SIP/2003") in new stack
    -- Called 2003
Aug  8 09:55:15 WARNING[6937]: channel.c:2725
ast_channel_make_compatible: No path to translate from SIP/2003-b5f8(4)
to SIP/2006-389a(256)

    -- SIP/2003-b5f8 is ringing
    -- SIP/2003-b5f8 answered SIP/2006-389a

Aug  8 09:55:16 WARNING[6937]: channel.c:2725
ast_channel_make_compatible: No path to translate from
SIP/2006-389a(256) to SIP/2003-b5f8(4) 
Aug  8 09:55:16 WARNING[6937]: app_dial.c:1608 dial_exec_full: Had to
drop call because I couldn't make SIP/2006-389a compatible with
SIP/2003-b5f8
  == Spawn extension (phones, 2003, 1) exited non-zero on
'SIP/2006-389a'


I think the codecs used by sip3 and sip1 are incompatible. Does anyone
know how I could make them compatible ?


Thank you.

Regards,
Kwang Mien