Is there any way you could get a cut-sheet from Verizon. I know they are difficult to work with, but it would help to see for sure if your circuit is indeed Loop-start. You could always try E&M_wink or E&M immediate and see if there is any change.
MATT--- On 7/8/08, Daniel Hazelbaker <[EMAIL PROTECTED]> wrote: > > Date: Mon, 7 Jul 2008 16:48:00 -0400 > > From: "Jason Aarons \(US\)" <[EMAIL PROTECTED]> > > > > > Digital ISDN used Q931 messages. You should get a disconnect message > > from telco on the d-channel 23. > > > I am pretty sure it is a T1 and not a PRI. I did try configuring it > as a PRI and it started spewing all kinds of errors and completely > stopped working. > > > > Date: Mon, 07 Jul 2008 16:55:27 -0400 > > From: Doug Lytle <[EMAIL PROTECTED]> > > > > > Daniel Hazelbaker wrote: > >> We are in the process of preparing to move our Asterisk server to a > >> Digital T1 interface card instead of a analog card (via an Adtran > >> which is now connected to the T1). I did a preliminary test the > >> other > >> > > > > A T1 or a PRI? Just make sure we're on the same page. > > Also, show us your zaptel and zapata.conf > > > > Again, I am pretty sure T1. It is a Verizon "Flex-Grow" package, > which they list as expandable up to 24 voice channels. That and I > tried configuring as a PRI and it harfed. The Adtran box we use now > is configured as: > > Timing Mode Network > Format ESF > Line Code B8ZS > Equalization 0 dB > CSU Lpbk Enable > Rx Sensitivity Auto > > Right now with Asterisk "mostly" working (it answers calls, dials out, > etc. just doesn't detect hangup) my /etc/zaptel.conf is: > # > # Span Configuration > # ~~~~~~~~~~~~~~~~~~ > span=1,1,0,esf,b8zs > span=2,0,0,esf,b8zs > > # > # Channel Configuration > # ~~~~~~~~~~~~~~~~~~~~~ > fxsks=1-24 > fxoks=25-48 > > loadzone = us > defaultzone=us > --CUT-- > > /etc/asterisk/zapata.conf: > [channels] > usecallerid=yes > callerid=asreceived > cidsignalling=bell > cidstart=ring > callprogress=yes # I have turned this off too > > ;------------------------------------------------- > ; > ; Define telco channels in rotary, these should be answered > ; like a normal incoming call. > ; > context=bridgeNEC > usecallerid=yes > signalling=fxs_ks > group=1 ; Part of ZAP group 1 > channel => 1-9 > > context=incoming > channel => 12 > > ;------------------------------------------------- > ; > ; Telco line, computer dialup, needs to be routed to output line. > ; > group=2 > usecallerid=no > channel => 10 ; PSTN attached to Span1:Port10 > > ;------------------------------------------------- > ; > ; Telco line, construction trailer fax, needs to be routed. > ; > group=3 > usecallerid=no > channel => 11 ; PSTN attached to Span1:Port11 > > > ;------------------------------------------------- > ; > ; ADTran lines, used for outgoing to analog devices > ; > context=incoming > group=4 > usecallerid=no > signalling=fxo_ks > channel => 25-36 > --CUT-- > > For context, the bridgeNEC context just dials out one of the ADTran > lines to our existing NEC system, but the incoming context starts our > menu-system, which was also not detecting hangups. > > I have also tried using loopstart and groundstart signalling, doesn't > seem to make a difference. I am pretty well stumped myself. I need > to call the telco about the caller id not working to verify that it is > still turned on, but I figure I might as well wait so that if I need > to ask them about the signalling I can know all the questions to ask > at the same time. > > > > Thanks, > > Daniel > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
