It said 'include zapata-channels.conf', where this line wasn't commented bij the ';'... Could you post me a working example of such a config (or a part of it, for the X100P cards...? Thanks guys! Message: 9 Date: Sat, 19 Mar 2005 18:04:26 -0500 From: "Jeff Glassman" <jrglass@columbus.rr.com> Subject: [Asterisk-Users] newbie question To: <asterisk-users@lists.digium.com> Message-ID: <000501c52cd8$02c10750$0200a8c0@newgems> Content-Type: text/plain; charset="us-ascii" bram kortleven Wrote>"Message: 6 >Date: Sat, 19 Mar 2005 22:16:39 +0100 >From: bram kortleven <bram@antwerpen.be> >Subject: [Asterisk-Users] newbie question >To: asterisk-users@lists.digium.com >Message-ID: <1111266999.7391.0.camel@athlon> >Content-Type: text/plain>I guess the first time it didn't get through... I didn't see it appearin >the list, that is...>I installed an Asterisk@home machineand configured a few SIP accountson >it. They seem to run fine inside my network, so that's OK. Now, I want to >start using a X100P to connect it to my phone line, to make call routing >between internal SIP phones/softphones, my local phoneline and an external >SIP server. How do I enable and configure the X100P?>I ran the configuration tool locally on the machine (the genzaptelconf >thing) and it added a line to the config. >Now using the number it gave me, in the trunk config in AMP, I stillcannot >get an outside line (connected it to a simple analogue pbx>system) and call outside the *-server.. >Could anyone help me with this? >Thanks guys"You need to go into the Zapta.conf and remove the semi colon ; channel => 1 Jeff
Hi, You said you are hooked to an analog PBX. Do you have to dial a 9 to gain access to an outside line on the PBX. If you do the easiest thing to do is go to the outbound-local context which is [outbound-local] exten => _NXXXXXX,1,Macro(dialout-default,${EXTEN}) exten => _NXXNXXXXXX,1,Macro(dialout-default,${EXTEN}) by default and change it to [outbound-local] exten => _9NXXXXXX,1,Macro(dialout-default,${EXTEN}) exten => _9NXXNXXXXXX,1,Macro(dialout-default,${EXTEN}) then reload asterisk from there if you dial 9,9 +the number you are trying to dial it should let you dial out ok. If you are having problems post what your CLI is showing. ----- Original Message ----- From: "bram" <bram@antwerpen.be> To: <asterisk-users@lists.digium.com> Sent: Saturday, March 19, 2005 8:55 PM Subject: [Asterisk-Users] RE:Newbie question> It said 'include zapata-channels.conf', where this line wasn't commented > bij the ';'... > > Could you post me a working example of such a config (or a part of it, > for the X100P cards...? > > Thanks guys! > > > Message: 9 > Date: Sat, 19 Mar 2005 18:04:26 -0500 > From: "Jeff Glassman" <jrglass@columbus.rr.com> > Subject: [Asterisk-Users] newbie question > To: <asterisk-users@lists.digium.com> > Message-ID: <000501c52cd8$02c10750$0200a8c0@newgems> > Content-Type: text/plain; charset="us-ascii" > > bram kortleven Wrote > >>"Message: 6 >>Date: Sat, 19 Mar 2005 22:16:39 +0100 >>From: bram kortleven <bram@antwerpen.be> >>Subject: [Asterisk-Users] newbie question >>To: asterisk-users@lists.digium.com >>Message-ID: <1111266999.7391.0.camel@athlon> >>Content-Type: text/plain > >>I guess the first time it didn't get through... I didn't see it appear > in >the list, that is... > > >>I installed an Asterisk@home machineand configured a few SIP accounts > on >it. They seem to run fine inside my network, so that's OK. Now, I > want to >start using a X100P to connect it to my phone line, to make > call routing >between internal SIP phones/softphones, my local phoneline > and an external >SIP server. How do I enable and configure the X100P? > >>I ran the configuration tool locally on the machine (the genzaptelconf >>thing) and it added a line to the config. >>Now using the number it gave me, in the trunk config in AMP, I still > cannot >get an outside line (connected it to a simple analogue pbx >>system) and call outside the *-server.. >>Could anyone help me with this? >>Thanks guys" > > You need to go into the Zapta.conf and remove the semi colon > > ; channel => 1 > > Jeff > > _______________________________________________ > 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
>Message: 5 >Date: Sun, 20 Mar 2005 03:55:50 +0100 >From: bram <bram@antwerpen.be> >Subject: [Asterisk-Users] RE:Newbie question >To: asterisk-users@lists.digium.com >Message-ID: <1111287350.7391.4.camel@athlon> >Content-Type: text/plain>It said 'include zapata-channels.conf', where this line wasn'tcommented >bij the ';'...>Could you post me a working example of such a config (or a part of it,for >the X100P cards...?>Thanks guys!Here is my Zapata,conf Do not worry about zapata-channels.conf' It is generated automaticly DO NOT CHANGE IT All I changed was ; channel => 1 to this channel => 1 Jeff ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=0.0 txgain=0.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no channel => 1 #include zapata-channels.conf