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

[JR Richardson]

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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