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<DIV><FONT face=Arial size=2>I don't have any wait commands in my s
extention.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I don't use (or need to use since they don't work
in the UK) caller display on external calls, but I do want to keep it on intenal
calls, so is there any way to turn it off on exernal calls only ?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>One more point, I rebooted my server and when I
tried to resart Asterisk again, I got an error saying something about no card on
d0001 (or something similar) and it refused to start. I had to run
"</FONT><SPAN class=Userinput><FONT face=Arial size=2>modprobe wcfxo" before I
could start the server. Is that normal, or is there something I can do so
it automaticly decects the card when I turn the server on.</FONT></SPAN></DIV>
<DIV><SPAN class=Userinput><FONT face=Arial size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=Userinput><FONT face=Arial size=2>Thanks again for yor help
Sean.</FONT></SPAN></DIV>
<DIV><SPAN class=Userinput><FONT face=Arial size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=Userinput><FONT face=Arial size=2>Paul.</DIV>
<DIV><BR></DIV></FONT></SPAN>
<DIV><FONT face=Arial size=2></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 20:41<BR>Posted To:
Asterisk-Users<BR>Conversation: [Asterisk-Users] x100p config<BR>Subject: RE:
[Asterisk-Users] x100p config</FONT></DIV>
<DIV> </DIV><FONT face=Arial size=2>
<DIV><BR>It could be one of several things. The two things that come to
mind is Caller ID and a Wait() statement in your dialplan. Since the
Caller ID information is transmitted between the first and second ring, Asterisk
has to wait for it if Caller ID is enabled. Other than that, is there
a<BR>Wait() line in your S extension?</DIV>
<DIV> </DIV>
<DIV>Sean<BR>-----Original Message-----<BR>From: Paul Tyreman
[mailto:paul@tyreman.org.uk] <BR>Sent: Sunday, April 18, 2004 2:31 PM<BR>To: <A
href="mailto:Asterisk-Users@lists.digium.com">Asterisk-Users@lists.digium.com</A><BR>Subject:
RE: [Asterisk-Users] x100p config</DIV>
<DIV> </DIV>
<DIV><BR>Thanks for your help. I've got it working now.</DIV>
<DIV> </DIV>
<DIV>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 ?</DIV>
<DIV> </DIV>
<DIV>Thanks, Paul.</DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>-----Original Message-----<BR>From: <A
href="mailto:asterisk-users-admin@lists.digium.com">asterisk-users-admin@lists.digium.com</A><BR>[mailto:asterisk-users-admin@lists.digium.com]
On Behalf Of Sean Cheesman Posted At: 18 April 2004 19:48 Posted To:
Asterisk-Users<BR>Subject: RE: [Asterisk-Users] x100p config</DIV>
<DIV> </DIV>
<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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