Jason Chan (jasonOfficial)
2005-Dec-14 22:12 UTC
[Asterisk-Users] I don't want ilbc, i just want G.711
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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@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@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@192.168.2.3:5060 > Caller-ID: elite > Need Destroy: 0 > Last Message: Rx: ACK > Promiscuous Redir: No > Route: sip:8888@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 > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/>-- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051215/5fe9e2d0/attachment.htm