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.richard...@gmail.comwrote:
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 PBXsip trunk rfc2833ast 1.4 rfc2833sip trunkcall
server ast 1.8 rfc2833sip trunkupstream carrier
** **
Customer PRIcisco PRI gatewaysip trunk rfc2833ast 1.4 rfc2833sip
trunk call server ast 1.8 rfc2833sip trunkupstream carrier
** **
I can see DTMF RTP events pass through call server to carrier but no
response, nothing, nada, zip.
** **
Working:
Customer SIP Phonesip rfc2833ast 1.4 rfc2833sip trunk call server
ast 1.8 rfc2833sip trunkupstream carrier
** **
Customer SIP Phonesip rfc2833ast 1.4 rfc2833sip trunk call server
ast 1.2 rfc2833sip trunkupstream carrier
** **
Customer IP PBXsip trunk rfc2833ast 1.4 rfc2833sip trunk call
server ast 1.2 rfc2833sip trunkupstream carrier
** **
Customer PRIcisco PRI gatewaysip trunk rfc2833ast 1.4 rfc2833 call
server sip trunkast 1.2sip trunkupstream 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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