It looks like http://bugs.digium.com/view.php?id=5266 is the problem here. My CDR shows as not answered for the tool free number. The local number answers and call forwards.
Questions: It says it was committed on 10-04-05. How do I know which versions that was? I am currently running: asterisk stable 1.0.9 zaptel stable 1.2.1 libpri stable 1.0.9 I was told that zaptel and asterisk versions do not have to match. What about libpri? Can I go to libpri 1.2.1 and stay with asterisk 1.0.9? Should I just patch 1.0.9? (I would have to figure out which version the patch was for) -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. --- - --- - - - - - - - -- - - - --- - ------ - - --- - - -- - - - -- - - - "Steven" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED] > This is an outbound issue that affects SIP and Zap (T1 from another PBX) > channels going out our PRI to Telco. > > I have two AT&T conference number that will take the conference access codes. > (in theory) > (214) 622 4991 > (866) 340 2763 > > If we dial the toll free one, the menus time out because they are not > recieving any DTMF. > If I wait and connect to the conference receptionist/tech(?) they can do a > three way call back in and my DTMF works. (they then > tell me there is no problem) > > If I call the 214 number it works without issue. The odd thing here is that > I receive DTMF back from them when it first answers > the line. > ref: > Dec 6 10:28:21 VERBOSE[1448]: -- Called g0/12146224991 > Dec 6 10:28:21 DEBUG[1448]: Ooh, format changed from unknown to ulaw > Dec 6 10:28:24 DEBUG[1448]: DTMF digit: * on Zap/2-1 > Dec 6 10:28:24 DEBUG[1448]: DTMF digit: 8 on Zap/2-1 > Dec 6 10:28:24 DEBUG[1448]: Enabled echo cancellation on channel 2 > > Is this something that they are sending to test/set some DTMF setting on my > side, or might I just be hearing them call forward to > some other number? > > The thing that really confuses me is the 866 number. If there is something > wrong with my setup, then why does my DTMF work if > they 3 way back in. I am still on the same call and do not think any settings > on my side would change because of what they do on > the other side. > > But I still think the Issue IS on my side, because if the main toll free AT&T > Conference number has this problem, I think they > would know and would have addressed it. > > zaptel.conf: > span=1,1,0,esf,b8zs > bchan=1-23 > dchan=24 > > span=2,0,0,esf,b8zs > e&m=25-48 > > loadzone = us > defaultzone = us > > > zapata.conf: > > context=from-pstn > switchtype=national > priindication = outofband > signalling=pri_cpe > rxwink=300 ; Atlas seems to use long (250ms) winks > usecallerid=yes > hidecallerid=no > callwaiting=no > usecallingpres=yes > callwaitingcallerid=no > threewaycalling=no > transfer=no > cancallforward=yes > callreturn=yes > echocancel=yes > echocancelwhenbridged=no > echotraining=no > rxgain=0.0 > txgain=0.0 > faxdetect=no > group=0 > callgroup=1 > pickupgroup=1 > immediate=no > accountcode=I > musiconhold=default > channel => 1-23 > > > -- > -- > Steven > > May you have the peace and freedom that come from abandoning all hope of > having a better past. > --- - --- - - - - - - - -- - - - --- - ------ - > - --- - - -- - - - -- - - - > > > _______________________________________________ > --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
