[Asterisk-Users] I don't want ilbc, i just want G.711

[Elton Machado]

And if, for some very strange reason, it doesn't work, use noload at
modules.conf ;)

Regards,





2005/12/15, Umair Bari <umairbari at gmail.com>:
>
> in your sip.cong [general] contexts
>
> put
> disallow=all
> allow=ulaw
> allow=alaw
>
> and in your sip user, use disallow only ONCE, that is
> disallow=all
> allow=ulaw
> allow=alaw
>
> hope this helps.
>
> regards,
>
> Umair bari
>
>  On 12/15/05, Jason Chan (jasonOfficial) <jason at jasonofficial.net> wrote:
>
> >     Hi there,
> > I am writing to ask about how to fix the codec to G.711 ONLY.
> > Actually what I am doing is, try to use DTMF when the POTS phone call
> > has
> > directed to Asterisk via Planet VIP-450 FXO Port, but this gateway just
> > simply doesn't support RFC2833 nor SIP-INFO. The only method I can use
> > is
> > Inband DTMF. I know it only support G.711, but I DID disallow others and
> > make it work only with G.711. But the problem is, although I disallow
> > all
> > other codecs, ilbc still itching me...
> > [extensions.conf]
> > [852]
> > username=HKGW
> > serect=blah
> > type=friend
> > host=dynamic
> > nat =yes
> > canreinvite=no
> > disallow=all
> > disallow=ilbc
> > allow=ulaw
> > dtmfmode=inband
> >
> > (P.S. I don't use REINVITE simply because I need the asterisk to be a
> > media gateway cause the gateway is inside NAT behind the Asterisk)
> > Whenever I try to pass DTMF from phone to Asterisk via that gateway, I
> > got
> > such messages:
> >
> > Dec 14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF
> > is
> > not supported on codec ilbc. Use RFC2833
> > Dec 14 23:35:32 WARNING[10958]: codec_ilbc.c:175 ilbctolin_framein: Huh?
> >
> > An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
> >
> > How come!? I DID DISALLOW them, but it keeps bugging me....
> >
> > =====
> > 192.168.2.3      852         79f9e0-c0a8  00101/00001  ulaw  No
> > Rx:
> > ACK
> > 1 active SIP channel
> > *CLI> sip show channel 79
> >
> >   * SIP Call
> >   Direction:              Incoming
> >   Call-ID:
> > 79f9e0-c0a80203-13c4-3a53f3e1-bbfcaf8-3fcf at 192.168.2.3 <http:///>
> >   Our Codec Capability:   4
> >   Non-Codec Capability:   0
> >   Their Codec Capability:   261
> >   Joint Codec Capability:   4
> >   Format                  ulaw
> >   Theoretical Address:    192.168.2.3:5060
> >   Received Address:       192.168.2.3:5060
> >   NAT Support:            Always
> >   Audio IP:               192.168.2.1 (local)
> >   Our Tag:                as737358ce
> >   Their Tag:              3a53f3e1-bbfcafe6d5c
> >   SIP User agent:
> >   Username:               852
> >   Peername:               852
> >   Original uri:           sip:8888 at 192.168.2.3:5060
> >   Caller-ID:              elite
> >   Need Destroy:           0
> >   Last Message:           Rx: ACK
> >   Promiscuous Redir:      No
> >   Route:                  sip:8888 at 192.168.2.3:5060
> >   DTMF Mode:              inband
> >   SIP Options:            (none)
> >
> > ======
> > Previously I installed 1.0.3 in same machine, but I overwrite all files
> > with 1.2.1.. does it cause a trouble?
> >
> >
> > Can anyone figure out what is the problem?
> >
> > ======================================================================
> > Thanks very much for your help!
> >
> > Best regards,
> > Jason Chan, Hong Kong
> >
> > No virus found in this outgoing message.
> > Checked by AVG Free Edition.
> > Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date:
> > 9/12/2005
> >
> >
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> >
>
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