[asterisk-users] Problems with Codecs in Asterisk

[Dean at INKnBITs]

I have the same problem here, why does asterisk not use ulaw with Sip1 -> Sip3 ?  As it has allow=g729 and allow=ulaw in Sip1, should it not fallback onto ulaw when the g729 fails?

Thanks,
Dean.
  -----Original Message-----
  From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com]On Behalf Of Rosli Sukri
  Sent: 08 August 2006 13:38
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: Re: [asterisk-users] Problems with Codecs in Asterisk


  either
  1)pay digium for g.729 license or
  2)allow g.729 for sip3

  - sip 1 -> sip2 work cause it will pass thru, 
  - sip 2 -> sip3 fails because since asterisk wants to do transcoding to 729<->711 and no license 
  if bandwidth is a concern just use GSM (if available as a codec on the phone)


  On 8/8/06, Chan Kwang Mien < kwangmien at asgent-tech.com> wrote:
    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



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