<div>in your sip.cong [general] contexts</div>
<div> </div>
<div>put </div>
<div>disallow=all</div>
<div>allow=ulaw</div>
<div>allow=alaw</div>
<div> </div>
<div>and in your sip user, use disallow only ONCE, that is </div>
<div>disallow=all</div>
<div>allow=ulaw</div>
<div>allow=alaw<br><br>hope this helps.</div>
<div> </div>
<div>regards,</div>
<div> </div>
<div>Umair bari</div>
<div> </div>
<div><span class="gmail_quote">On 12/15/05, <b class="gmail_sendername">Jason Chan (jasonOfficial)</b> <<a href="mailto:jason@jasonofficial.net">jason@jasonofficial.net</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">
<div><font size="2"><font size="3"> Hi there,<br>I am writing to ask about how to fix the codec to G.711 ONLY.<br>Actually what I am doing is, try to use DTMF when the POTS phone call has<br>directed to Asterisk via Planet VIP-450 FXO Port, but this gateway just
<br>simply doesn't support RFC2833 nor SIP-INFO. The only method I can use is<br>Inband DTMF. I know it only support G.711, but I DID disallow others and<br>make it work only with G.711. But the problem is, although I disallow all
<br>other codecs, ilbc still itching me...<br>[extensions.conf]<br>[852]<br>username=HKGW<br>serect=blah<br>type=friend<br>host=dynamic<br>nat =yes<br>canreinvite=no<br>disallow=all<br>disallow=ilbc<br>allow=ulaw<br>dtmfmode=inband
<br><br>(P.S. I don't use REINVITE simply because I need the asterisk to be a<br>media gateway cause the gateway is inside NAT behind the Asterisk)<br>Whenever I try to pass DTMF from phone to Asterisk via that gateway, I got
<br>such messages:<br><br>Dec 14 23:35:32 WARNING[10958]: dsp.c:1422 ast_dsp_process: Inband DTMF is<br>not supported on codec ilbc. Use RFC2833<br>Dec 14 23:35:32 WARNING[10958]: codec_ilbc.c:175 ilbctolin_framein: Huh?
<br>An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?<br><br>How come!? I DID DISALLOW them, but it keeps bugging me....<br><br>=====<br><a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.2.3/" target="_blank">
192.168.2.3</a> 852 79f9e0-c0a8 00101/00001 ulaw No Rx:<br>ACK<br>1 active SIP channel<br>*CLI> sip show channel 79<br><br> * SIP Call<br> Direction: Incoming<br> Call-ID:
<br></font><a onclick="return top.js.OpenExtLink(window,event,this)" href="http:///" target="_blank"><font size="3">79f9e0-c0a80203-13c4-3a53f3e1-bbfcaf8-3fcf@192.168.2.3</font></a><br></font><font size="3"> Our Codec Capability: 4
<br> Non-Codec Capability: 0<br> Their Codec Capability: 261<br> Joint Codec Capability: 4<br> Format ulaw<br> Theoretical Address: <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.2.3:5060/" target="_blank">
192.168.2.3:5060</a><br> Received Address: <a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.2.3:5060/" target="_blank">192.168.2.3:5060</a><br> NAT Support: Always<br> Audio IP:
<a onclick="return top.js.OpenExtLink(window,event,this)" href="http://192.168.2.1/" target="_blank">192.168.2.1</a> (local)<br> Our Tag: as737358ce<br> Their Tag: 3a53f3e1-bbfcafe6d5c<br> SIP User agent:
<br> Username: 852<br> Peername: 852<br> Original uri: sip:8888@192.168.2.3:5060<br> Caller-ID: elite<br> Need Destroy: 0<br> Last Message: Rx: ACK
<br> Promiscuous Redir: No<br> Route: sip:8888@192.168.2.3:5060<br> DTMF Mode: inband<br> SIP Options: (none)<br><br>======<br>Previously I installed 1.0.3 in same machine, but I overwrite all files
<br>with 1.2.1.. does it cause a trouble?<br><br><br>Can anyone figure out what is the problem? <br><br>======================================================================<br>Thanks very much for your help!<br><br>Best regards,
<br>Jason Chan, Hong Kong</font></div><br>No virus found in this outgoing message.<br>Checked by AVG Free Edition.<br>Version: 7.1.371 / Virus Database: 267.13.13/197 - Release Date: 9/12/2005<br><br><br>_______________________________________________
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