mvickers@real.com
2003-Sep-05 14:31 UTC
[Asterisk-Users] T1 - A little guidance needed to get started, What order to do zaptel, zapata...
I have about a dozen SIP phones up and working, now I want to connect the asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers conencted to the Fujitsu PBX that I built with mgetty/pppd and have the lines provisioned the same way as those dial-up server, ESF, B8ZS, and E&M wink start, so I have confidence in the guys who set up the PBX. I've built a loop back plug for T1 and looped it both directions, I can get the Digum T100P card to go green as well as the PBX port. So I have confidence in the wiring. I don't understand the relation ship between zaptel and zapata and wether I need to config a dail plan or anything else in asterisk before I get the T1 up. So do I start with /etc/zaptel.conf and the zaptel module. Then work with the zapata stuff (Does this have a module), then asterisk. There seems to be a lot of docs scattered around about asterisk is a good list where docs are, and some kind of overview. I'm a router/linux guy not a PBX guy and the architecture and nomenclature are unfamilure to me, is there a PBX book I should read for background info. Oh and how does one make the equivalent of a defaut route in the dial plan, ie any call not listed in the dial plan goes out the T1 Thanks in advance. Mark Vickers, RealNetworks Inc. Desk: (206) 674-2391 Fax: (206)674-3588
mvickers@real.com
2003-Sep-05 17:28 UTC
[Asterisk-Users] T1 - A little guidance needed to get started, What order to do zaptel, zapata...
More info: cat /etc/zaptel.conf |grep -v "^#" span=1,1,0,esf,b8zs e&m=1-24 loadzone = us defaultzone=us cat /etc/asterisk/zapata.conf |grep -v "^;" [channels] context=default signalling=em_w group=1 channel => 1-24 lsmod: Module Size Used by Not tainted wct1xxp 12320 24 zaptel 180416 50 [wct1xxp] ppp_generic 15776 0 [zaptel] slhc 4384 0 [ppp_generic] dmesg: Freed a Wildcard Zapata Telephony Interface Unloaded PPP generic driver version 2.4.2 Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 9 for device 01:09.0 PCI: Sharing IRQ 9 with 00:1f.3 Framer: DS21552, Revision: 3 (T1) Found a Wildcard: Digium Wildcard T100P T1/PRI Registered tone zone 0 (United States / North America) Using ESF/B8ZS coding/framing Calling startup (flags is 4099) Thanks again Mark Vickers, RealNetworks Inc. Desk: (206) 674-2391 Fax: (206)674-3588 On Fri, 5 Sep 2003 mvickers@real.com wrote:> > I have about a dozen SIP phones up and working, now I want to connect the > asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers > conencted to the Fujitsu PBX that I built with mgetty/pppd and have the > lines provisioned the same way as those dial-up server, ESF, B8ZS, and E&M > wink start, so I have confidence in the guys who set up the PBX. > > I've built a loop back plug for T1 and looped it both directions, I can > get the Digum T100P card to go green as well as the PBX port. So I have > confidence in the wiring. > > I don't understand the relation ship between zaptel and zapata and wether > I need to config a dail plan or anything else in asterisk before I get > the T1 up. So do I start with /etc/zaptel.conf and the zaptel module. > > Then work with the zapata stuff (Does this have a module), then asterisk. > > There seems to be a lot of docs scattered around about asterisk is a good > list where docs are, and some kind of overview. I'm a router/linux guy > not a PBX guy and the architecture and nomenclature are unfamilure to me, > is there a PBX book I should read for background info. > > Oh and how does one make the equivalent of a defaut route in the dial > plan, ie any call not listed in the dial plan goes out the T1 > > Thanks in advance. > > > Mark Vickers, RealNetworks Inc. Desk: (206) 674-2391 Fax: (206)674-3588 > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users >
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