[asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks

[Jared Geiger]

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.

Rgds,
Jared

On Wed, Nov 9, 2011 at 7:03 PM, JR Richardson <jmr.richardson at gmail.com>wrote:

>  Hi All,****
>
> ** **
>
> 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.****
>
> ** **
>
> 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:****
>
> ** **
>
> Not working:****
>
> Customer IP PBX><sip trunk rfc2833><ast 1.4 rfc2833><sip trunk><call
> server ast 1.8 rfc2833><sip trunk><upstream carrier****
>
> ** **
>
> Customer PRI><cisco PRI gateway><sip trunk rfc2833><ast 1.4 rfc2833><sip
> trunk>< call server ast 1.8 rfc2833><sip trunk><upstream carrier****
>
> ** **
>
> I can see DTMF RTP events pass through call server to carrier but no
> response, nothing, nada, zip.****
>
> ** **
>
> Working:****
>
> Customer SIP Phone><sip rfc2833><ast 1.4 rfc2833><sip trunk>< call server
> ast 1.8 rfc2833><sip trunk><upstream carrier****
>
> ** **
>
> Customer SIP Phone><sip rfc2833><ast 1.4 rfc2833><sip trunk>< call server
> ast 1.2 rfc2833><sip trunk><upstream carrier****
>
> ** **
>
> Customer IP PBX><sip trunk rfc2833><ast 1.4 rfc2833><sip trunk>< call
> server ast 1.2 rfc2833><sip trunk><upstream carrier****
>
> ** **
>
> Customer PRI><cisco PRI gateway><sip trunk rfc2833><ast 1.4 rfc2833>< call
> server sip trunk><ast 1.2><sip trunk><upstream carrier****
>
> ** **
>
> I can see DTMF RTP events pass through to carrier, RTP stream looks the
> same as the 1.8 server with reliable responses.****
>
> ** **
>
> On both the 1.4 and 1.8 ast servers, these sip.conf parameters are active
> on peer and global settings:****
>
> relaxdtmf=yes****
>
> rfc2833compensate=yes****
>
> dtmfmode=rfc2833****
>
> ** **
>
> 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.****
>
> ** **
>
> 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.****
>
> ** **
>
> 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?****
>
> ** **
>
> Thanks.****
>
>
> JR****
>
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