<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
hi!<br>
<br>
I'm having the same problem, I'm connecting through a Planet VIP-450
ITG, and when I send a DTMF code I get a:<br>
<br>
WARNING: codec_ilbc.c:141 ilbctolin_framein: Huh? An ilbc frame that
isn't a multiple of 50 bytes long from RTP (4)?<br>
<br>
I tried using different dtmf settings in sip.conf, but the message is
still there. I don't have problems using a softphone...<br>
<br>
any ideas???<br>
<br>
saludos! santiago.<br>
<br>
<br>
Lee Norvall wrote:
<blockquote cite="mid007001c448d1$9d83ecd0$6401a8c0@eth0s" type="cite">
<pre wrap="">Hi
Just tried that, and still the same with the same error! The spec for
the phones includes rfc2833, so I don't think that is it.
Rgds
-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-admin@lists.digium.com">asterisk-users-admin@lists.digium.com</a>
[<a class="moz-txt-link-freetext" href="mailto:asterisk-users-admin@lists.digium.com">mailto:asterisk-users-admin@lists.digium.com</a>] On Behalf Of Justin
Carlson
Sent: 02 June 2004 19:23
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Subject: Re: [Asterisk-Users] DTMF and SIP
have you tried commenting out the dtmf lines in your sip.conf we had
similar problems with our snom 200's and after commenting out the dtmf
lines in sip.conf & asterisk reload they worked great :-)
On Wed, 2004-06-02 at 11:36, Lee Norvall wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Hi
I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the
phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also
tried inband) and I get the following error:
june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein:
Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP
(4)?
This means that I cannot get access to voicemail from the handsets !!!
Any clues???
</pre>
</blockquote>
<pre wrap=""><!---->
_______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com</a>
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
_______________________________________________
Asterisk-Users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com</a>
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<br>
</body>
</html>