I had similar problems with 1.8.6 and polycom phones intermittently having DTMF issues. I updated to 1.8.7 and things cleared up. I went through the release notes at the time, but don't recall which commit made me decide to give it a try.<br>
<br>Rgds,<br>Jared<br><br><div class="gmail_quote">On Wed, Nov 9, 2011 at 7:03 PM, JR Richardson <span dir="ltr"><<a href="mailto:jmr.richardson@gmail.com">jmr.richardson@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">
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<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Hi All,<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">I recently turned up some 1.8.6.0 call servers in
productions, SIP trunks in routing calls to upstream carrier via SIP trunks
out. I spent a lot of time in the lab testing 1.8 which included heavily
testing DTMF with no issues that came up. It all just seemed to work fine.
But then again you can’t reproduce every real work scenario in the lab.<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">I’m using rfc2833 inbound and outbound for the new 1.8
call servers. Here is a quick diagram of what is working and what is not:<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Not working:<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Customer IP PBX><sip trunk rfc2833><ast 1.4
rfc2833><sip trunk><call server ast 1.8 rfc2833><sip
trunk><upstream carrier<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Customer PRI><cisco PRI gateway><sip trunk
rfc2833><ast 1.4 rfc2833><sip trunk>< call server ast 1.8
rfc2833><sip trunk><upstream carrier<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">I can see DTMF RTP events pass through call server to
carrier but no response, nothing, nada, zip.<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Working:<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Customer SIP Phone><sip rfc2833><ast 1.4
rfc2833><sip trunk>< call server ast 1.8 rfc2833><sip
trunk><upstream carrier<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Customer SIP Phone><sip rfc2833><ast 1.4
rfc2833><sip trunk>< call server ast 1.2 rfc2833><sip
trunk><upstream carrier<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Customer IP PBX><sip trunk rfc2833><ast 1.4
rfc2833><sip trunk>< call server ast 1.2 rfc2833><sip
trunk><upstream carrier<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Customer PRI><cisco PRI gateway><sip trunk
rfc2833><ast 1.4 rfc2833>< call server sip trunk><ast 1.2><sip
trunk><upstream carrier<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">I can see DTMF RTP events pass through to carrier, RTP
stream looks the same as the 1.8 server with reliable responses.<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">On both the 1.4 and 1.8 ast servers, these sip.conf
parameters are active on peer and global settings:<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">relaxdtmf=yes<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">rfc2833compensate=yes<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">dtmfmode=rfc2833<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Now it quickly appears like a problem between the customer
PBX and Customer PRI with the SIP trunks to the ast 1.4 servers but it all
worked fine before with the 1.2 call servers. After the upgrade of the
call servers to 1.8 DTMF is not recognized by the carrier on calls from the
customer IP PBX or PRI but is fine with the SIP phones directly registered to the
ast 1.4 servers.<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">I found the bug issues with the SRCC change/update issues
with DTMF events. It looks like 1.8.6.0 implemented the ‘update’
and as I read it, should have fixed the issue with the changing SRCC effecting
DTMF. But this may not be the case.<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Specifically, how would I debug RTP/DTMF on the new ast 1.8
server and see if the SRCC is changing between my scenarios described above.
Am I on the right track or is there something else I should be looking at?<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial">Thanks.<u></u><u></u></span></font></p>
<p class="MsoNormal"><font face="Arial" size="2"><span style="font-size:10.0pt;font-family:Arial"><br><font color="#888888">
JR<u></u><u></u></font></span></font></p>
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