Recheck your zaptel.conf. That's not the correct setup for a T1 trunk. You need to know the signalling the channel bank uses, and specify the voice channels (bchannel=1-24), and the signalling channel (dchannel=25). Those numbers are bogus, as I've never worked with T1 ;)
BTW, why are you using such setup (1 channel bank to connect to 24 analog lines) instead of asking your Telco to install a T1 trunk in your office? Julian. On 7/5/05, Mehran Mozaffari <[EMAIL PROTECTED]> wrote: > Hi, > > I have some problem to get this setup working. I have a CAC Channel > Banl I, with FXO and an Asterisk box ( I am using [EMAIL PROTECTED] 1.2) > and I have a TE110p installed in this box. > > What I want to do is, just to be able to dial one of those lines that > already are connected to the channel bank, and transfer that call > through TE110p and Asterisk to a user agent somewhere through > Internet. > > <PSTN>-----<CAC Channel Bank I>-----<TE110p>-----<Asterisk>-----<SIP User > Agent> > > At this time the SIP UAs can communicate with each other and > everything works properly, but I can't dial through channel bank. When > I dial one of those numbers, I will get no answer ring, and I can't > see anything coming to Asterisk through CLI. and when I tried to dial > through SIP UA to the PSTN end, I will get all circuits are busy now > from asterisk. > > Any Idea what should I do? at this time all lights are green and it > looks like that everything is working properly, but I am not sure > where is the problem, here are my settings: > > /etc/zaptel.conf: > ---------------------------------------- > span=1,1,0,esf,b8zs > fxols=1-24 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users