First of all, Alex, sorry for not seeing your reply. Nearly two weeks ago now :(
Honestly, with canreinvite=yes, I'm not sure what is meant by "the signalling still travels through asterisk"... I would ASSUME that includes out-of-band dtmf as well. Sorry! Moj Alex Crow wrote: > Moj, > > Does this mean that even out-of-band DTMF still gets sent > SIP-phone<-->SIP-phone without Asterisk hearing them? (eg RFCxxxx DTMF, > can't remember the number right now) > > Forgive me for butting into this thread but this is interesting... > > Cheers > > Alex > > > On Tue, 2007-06-12 at 09:21 -0800, Mojo with Horan & Company, LLC wrote: >> theoretically, with canreinvite=yes, it's phone <-> phone. with >> canreinvite=no, it's phone <-> asterisk <-> phone. BUT there are a few >> reasons which canreinvite=yes will not be this way. If for example you >> have a T or a t in the Dial string, asterisk will _remain_ in the media >> path so it can still detect the DTMF requests for transfer. >> >> Moj >> >> Deepak Naidu wrote: >>> Sounds crazy right? even was I, more over support guy logged in unloaded >>> the zap modules to test them, still an echo. >>> >>> Ya, I was clear saying that we have SIP--- SIP issue ie internal >>> extension echo problem. It seems the echo with SIP--SIP has many >>> factors. I am just curios to eliminate any possibility of Asterisk >>> failing to cancel the echo. >>> >>> OK, one question here howz the call flow when a SIP---SIP call is >>> established ie. is the connection between 2 phones when an Internal >>> call is made or does the SIP call goes via Asterisk once the SIP--SIP >>> call is establised. >>> >>> -- >>> Deepak >>> >>> */Matthew Fredrickson <[EMAIL PROTECTED]>/* wrote: >>> >>> >>> On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote: >>> >>> > Hi, >>> > We have a PRI connection & when its was on test >>> networks we >>> > had echo problems withoutside line. >>> > >>> > So I bought a TE212P card resolve the echo problem. Which did to an >>> > extent. Its using asterisk 1.2.18 & RHEL4-Update 4. >>> > >>> > >>> > But now when we are live, there is a terrible echo between 2 SIP >>> > calls. If I call the same extension from outside the voice is clear. >>> > >>> > I am not sure whats the problem. Also there's slight echo when >>> > calling Digium support. >>> > >>> > Totally lost Digium says we need to remove the echo module to >>> resolve >>> > SIP echo problems. Then ? the heck we pay for.. >>> >>> Are you sure that they understood that you were having this problem >>> between 2 SIP endpoints? That advice only makes sense to test if one >>> side is Zap and the other side is SIP. >>> >>> >>> --- >>> Matthew Fredrickson >>> Software Engineer >>> Digium, Inc. >>> >>> _______________________________________________ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >>> >>> ------------------------------------------------------------------------ >>> Yahoo! Answers - Get better answers from someone who knows. Try it now >>> <http://uk.answers.yahoo.com/;_ylc=X3oDMTEydmViNG02BF9TAzIxMTQ3MTcxOTAEc2VjA21haWwEc2xrA3RhZ2xpbmU>. >>> >>> >>> >>> >>> ------------------------------------------------------------------------ >>> >>> _______________________________________________ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> asterisk-users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users