JR Richardson
2011-Nov-10 00:03 UTC
[asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111109/a622f35f/attachment.htm>
Jared Geiger
2011-Nov-10 00:55 UTC
[asterisk-users] DTMF issue with 1.8.6.0 and SIP Trunks
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**** > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111109/cc914fb0/attachment.htm>