<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 TRANSITIONAL//EN">
<HTML>
<HEAD>
<META HTTP-EQUIV="Content-Type" CONTENT="text/html; CHARSET=UTF-8">
<META NAME="GENERATOR" CONTENT="GtkHTML/3.16.1">
</HEAD>
<BODY>
Hello again,<BR>
<BR>
Just to close this I have found the problem to be related to 1.4.10. For some unknown reason the sip debug showed<BR>
<BR>
Found description format PCMU for ID 0
<PRE>
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): <FONT COLOR="#ff0000"><B>us - 0x0 (nothing)</B></FONT>, peer - 0x0 (nothing), combined - 0x0 (nothing)
</PRE>
after upgrading to 1.4.17 everything worked ok again with the same configuration files:<BR>
<BR>
Found description format PCMU for ID 0
<PRE>
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
</PRE>
All here: <A HREF="http://www.len.ro/work/tools/gutsy-on-a-ubuntu-server/asterisk/view">http://www.len.ro/work/tools/gutsy-on-a-ubuntu-server/asterisk/view</A><BR>
<BR>
Best regards,<BR>
Len<BR>
<A HREF="http://www.len.ro">http://www.len.ro</A>
<PRE>
</PRE>
<BR>
On Mon, 2008-01-07 at 13:57 +0200, Len wrote:<BR>
<BLOCKQUOTE TYPE=CITE>
<FONT COLOR="#000000">Hello,</FONT><BR>
<BR>
<FONT COLOR="#000000">I have the following problem. I am migrating my asterisk infrastructure to a new server and I encounter a strange problem. The configuration is as followin: IAX clients connect to asterisk which forward calls to a sip box connected to a phone line. On the old server everything works ok but on the new server, even if the logs are identical it seems like the dtmf number does not get passed correctly to the sip box as the phone does not dial the proper number. The log shows something similar to:</FONT><BR>
<BR>
<FONT COLOR="#000000">[Jan 7 13:33:11] VERBOSE[7785] logger.c: -- Called 1002</FONT><BR>
<FONT COLOR="#000000">[Jan 7 13:33:11] VERBOSE[7785] logger.c: -- SIP/1002-081b4a80 answered IAX2/ioper00-1</FONT><BR>
<FONT COLOR="#000000">[Jan 7 13:33:11] VERBOSE[7785] logger.c: -- Sending DTMF 'w0214108658' to the called party.</FONT><BR>
<BR>
<FONT COLOR="#000000">where 1002 is the sip box</FONT><BR>
<BR>
<FONT COLOR="#000000">[1002]</FONT><BR>
<FONT COLOR="#000000">type=friend</FONT><BR>
<FONT COLOR="#000000">username=<A HREF="mailto:1002@10.0.0.1">1002@10.0.0.1</A></FONT><BR>
<FONT COLOR="#000000">callerid="1002"</FONT><BR>
<FONT COLOR="#000000">secret=xxxxxxx</FONT><BR>
<FONT COLOR="#000000">host=dynamic</FONT><BR>
<FONT COLOR="#000000">dtmfmode=inband</FONT><BR>
<FONT COLOR="#000000">deny=0.0.0.0/0.0.0.0</FONT><BR>
<FONT COLOR="#000000">permit=10.0.0.121/255.255.255.255</FONT><BR>
<BR>
<FONT COLOR="#000000">The only problem I can think of is dtmf related. Did something change from asterisk 1.2.13 to 1.4.10 which could cause this problem? Can it be related to the computer speed (very unlikely in my mind).</FONT><BR>
<BR>
<FONT COLOR="#000000">Thank you very much for any ideeas as I am bumping my head for a hole day trying various combination.</FONT><BR>
<BR>
<FONT COLOR="#000000">Best regards,</FONT><BR>
<FONT COLOR="#000000">Len</FONT><BR>
<FONT COLOR="#000000"><A HREF="http://www.len.ro">http://www.len.ro</A></FONT><BR>
<BR>
<BR>
<PRE>
<FONT COLOR="#000000">_______________________________________________</FONT>
<FONT COLOR="#000000">--Bandwidth and Colocation Provided by <A HREF="http://www.api-digital.com">http://www.api-digital.com</A>--</FONT>
<FONT COLOR="#000000">asterisk-users mailing list</FONT>
<FONT COLOR="#000000">To UNSUBSCRIBE or update options visit:</FONT>
<FONT COLOR="#000000"> <A HREF="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</A></FONT>
</PRE>
</BLOCKQUOTE>
</BODY>
</HTML>