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

[Umair Bari]

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
>
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>
>
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