<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=Content-Type content="text/html; charset=big5">
<META content="MSHTML 6.00.2900.2180" name=GENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=#ffffff>
<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>192.168.2.3
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 href=""><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: 192.168.2.3:5060<BR>
Received Address: 192.168.2.3:5060<BR>
NAT Support:
Always<BR> Audio
IP:
192.168.2.1 (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></BODY></HTML>