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.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**** > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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