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<DIV><FONT face=Arial size=2>Thanks for your help. I've got it working
now.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Only one problem. When users from the public
network call my server, they hear three rings before the phones on my server
start ringing. Is that usual, or is it a setting that can be changed
?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks, Paul.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>-----Original Message-----<BR>From: <A
href="mailto:asterisk-users-admin@lists.digium.com">asterisk-users-admin@lists.digium.com</A>
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Sean
Cheesman<BR>Posted At: 18 April 2004 19:48<BR>Posted To:
Asterisk-Users<BR>Subject: RE: [Asterisk-Users] x100p config</FONT></DIV>
<DIV> </DIV><FONT face=Arial size=2>
<DIV><BR>Welcome to the wonderful world of Asterisk! In the future, you
might want to make sure that you post in plain text mode instead of HTML. There
are quite a few people here who are great assets that won't even read if you
post in HTML.</DIV>
<DIV> </DIV>
<DIV>Your problem has to do with the contexts. In your zapata.conf file,
you will see reference to a context for your X100P. That is the context
into which calls on that card will be dumped. If you check your
extensions.conf, you should find a matching context that will have all of the
demo stuff in it. You can either change the demo context to meet your
needs, or change your zapata.conf to point to a more useful context that has
just what you want in it.</DIV>
<DIV> </DIV>
<DIV>You might want to read over the info at <A
href="http://www.voip-info.org">http://www.voip-info.org</A>. There's a lot of
good reading there that will help you make the most of Asterisk.</DIV>
<DIV> </DIV>
<DIV>Sean</DIV>
<DIV> </DIV>
<DIV>-----Original Message-----<BR>From: Paul Tyreman
[mailto:paul@tyreman.org.uk] <BR>Sent: Sunday, April 18, 2004 1:31 PM<BR>To: <A
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com</A><BR>Subject:
[Asterisk-Users] x100p config</DIV>
<DIV> </DIV>
<DIV><BR>Hi,</DIV>
<DIV> </DIV>
<DIV>I have just installed my first X100P card, and seams to be half
working.</DIV>
<DIV> </DIV>
<DIV>You can call the public telephone number which the card is attached to and
hear some lady telling you about asterisk. If I dial the extention number
of the phone I want to call, it connects and it's all good.</DIV>
<DIV> </DIV>
<DIV>However, I have put this line in my extensions.conf:<BR>[incoming]<BR>exten
=> s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr)</DIV>
<DIV> </DIV>
<DIV>So it should ring phone one and phone two rather then give that that girls
voice ! Can anyone tell me what I'm doing wrong ?</DIV>
<DIV> </DIV>
<DIV><BR>Also, I have put this in the same extensions.conf file: [outgoing]
exten => _0X.,1,Dial,Zap/1/${EXTEN:1}</DIV>
<DIV> </DIV>
<DIV>[sip]<BR>include => outgoing</DIV>
<DIV> </DIV>
<DIV>Yet I still cannot make outgoing calls, when I dial 0 and the number I want
to call on the public network.</DIV>
<DIV> </DIV>
<DIV>Any help would be great as I'm starting to pull my hair out !</DIV>
<DIV> </DIV>
<DIV>Thanks,
Paul.<BR>_______________________________________________<BR>Asterisk-Users
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