[Asterisk-Users] RE:Newbie question

bram bram at antwerpen.be
Sat Mar 19 19:55:50 MST 2005


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 at columbus.rr.com>
Subject: [Asterisk-Users] newbie question
To: <asterisk-users at lists.digium.com>
Message-ID: <000501c52cd8$02c10750$0200a8c0 at 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 at antwerpen.be>
>Subject: [Asterisk-Users] newbie question
>To: asterisk-users at lists.digium.com
>Message-ID: <1111266999.7391.0.camel at 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 at 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




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