[asterisk-users] Problems with Codecs in Asterisk

[Rosli Sukri]

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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