We spent a considerable amount of time getting an A101 up and running. Try to find out what type of switch you are connecting to. In our case, we were working against a Nortel. For some reason, if we used ni2, it would not work. Finally setting the switchtype to 5ess or DMS100 would work and now everything sings.
Hope that helps. Jason -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, August 01, 2007 4:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] pri "call by call" trunking? Call Sangoma On 8/1/07, Erik Anderson <[EMAIL PROTECTED]> wrote: > On 8/1/07, John covici <[EMAIL PROTECTED]> wrote: > > I had some troubles -- try setting the timing parameter to 0 (second > > one in your span) and see if that helps. > > If I'm reading the docs correctly, this param should only be set to 0 > if you *never* want to use the T1 connected to this port for timing. > That's not the case in my situation, as I need to be syncing with the > telco's clock. > > That said, in the interest of troubleshooting, I did try setting it to > zero - this didn't fix the problem. > > -erik > > _______________________________________________ > --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 > _______________________________________________ --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 _______________________________________________ --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
