[asterisk-users] Problems with Codecs in Asterisk

[Rosli Sukri]

On 8/8/06, Dean @ INKnBITs <dean.bath at inknbits.co.uk> wrote:
>
>  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?
>

true, it might be a problem on da sip phones  itself (order of codec
preference/precedence maybe) - can you confirm what codec is sip1 passing it
to asterisk?..

currently for me i am using a pa1688 based sip phone and when setting the
codec you have to set the precedence order. i.e try ulaw, gsm then as a last
option use 729.

i am speculating in this particular scenario during the initialisation of
sip1 <-> asterisk wants bof of them probably agreed to do 729 as a result of
the precedence setting on the phone

maybe as an experiment, get sip3 to call sip2?

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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