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Hi, I have just installed my first X100P card, and seams to be half working. 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. However, I have put this line in my extensions.conf: [incoming] exten => s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr) 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 ? Also, I have put this in the same extensions.conf file: [outgoing] exten => _0X.,1,Dial,Zap/1/${EXTEN:1} [sip] include => outgoing Yet I still cannot make outgoing calls, when I dial 0 and the number I want to call on the public network. Any help would be great as I'm starting to pull my hair out ! Thanks, Paul. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040418/66e262c1/attachment.htm
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. 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. You might want to read over the info at http://www.voip-info.org. There's a lot of good reading there that will help you make the most of Asterisk. Sean -----Original Message----- From: Paul Tyreman [mailto:paul@tyreman.org.uk] Sent: Sunday, April 18, 2004 1:31 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] x100p config Hi, I have just installed my first X100P card, and seams to be half working. 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. However, I have put this line in my extensions.conf: [incoming] exten => s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr) 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 ? Also, I have put this in the same extensions.conf file: [outgoing] exten => _0X.,1,Dial,Zap/1/${EXTEN:1} [sip] include => outgoing Yet I still cannot make outgoing calls, when I dial 0 and the number I want to call on the public network. Any help would be great as I'm starting to pull my hair out ! Thanks, Paul.
Thanks for your help. I've got it working now. 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 ? Thanks, Paul. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Sean Cheesman Posted At: 18 April 2004 19:48 Posted To: Asterisk-Users Subject: RE: [Asterisk-Users] x100p config 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. 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. You might want to read over the info at http://www.voip-info.org. There's a lot of good reading there that will help you make the most of Asterisk. Sean -----Original Message----- From: Paul Tyreman [mailto:paul@tyreman.org.uk] Sent: Sunday, April 18, 2004 1:31 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] x100p config Hi, I have just installed my first X100P card, and seams to be half working. 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. However, I have put this line in my extensions.conf: [incoming] exten => s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr) 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 ? Also, I have put this in the same extensions.conf file: [outgoing] exten => _0X.,1,Dial,Zap/1/${EXTEN:1} [sip] include => outgoing Yet I still cannot make outgoing calls, when I dial 0 and the number I want to call on the public network. Any help would be great as I'm starting to pull my hair out ! Thanks, Paul. _______________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040418/adf1d0c8/attachment.htm
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 Wait() line in your S extension? Sean -----Original Message----- From: Paul Tyreman [mailto:paul@tyreman.org.uk] Sent: Sunday, April 18, 2004 2:31 PM To: Asterisk-Users@lists.digium.com Subject: RE: [Asterisk-Users] x100p config Thanks for your help. I've got it working now. 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 ? Thanks, Paul. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Sean Cheesman Posted At: 18 April 2004 19:48 Posted To: Asterisk-Users Subject: RE: [Asterisk-Users] x100p config 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. 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. You might want to read over the info at http://www.voip-info.org. There's a lot of good reading there that will help you make the most of Asterisk. Sean -----Original Message----- From: Paul Tyreman [mailto:paul@tyreman.org.uk] Sent: Sunday, April 18, 2004 1:31 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] x100p config Hi, I have just installed my first X100P card, and seams to be half working. 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. However, I have put this line in my extensions.conf: [incoming] exten => s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr) 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 ? Also, I have put this in the same extensions.conf file: [outgoing] exten => _0X.,1,Dial,Zap/1/${EXTEN:1} [sip] include => outgoing Yet I still cannot make outgoing calls, when I dial 0 and the number I want to call on the public network. Any help would be great as I'm starting to pull my hair out ! Thanks, Paul. _______________________________________________ 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
I don't have any wait commands in my s extention. 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 ? 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 "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. Thanks again for yor help Sean. Paul. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Sean Cheesman Posted At: 18 April 2004 20:41 Posted To: Asterisk-Users Conversation: [Asterisk-Users] x100p config Subject: RE: [Asterisk-Users] x100p config 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 Wait() line in your S extension? Sean -----Original Message----- From: Paul Tyreman [mailto:paul@tyreman.org.uk] Sent: Sunday, April 18, 2004 2:31 PM To: Asterisk-Users@lists.digium.com Subject: RE: [Asterisk-Users] x100p config Thanks for your help. I've got it working now. 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 ? Thanks, Paul. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Sean Cheesman Posted At: 18 April 2004 19:48 Posted To: Asterisk-Users Subject: RE: [Asterisk-Users] x100p config 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. 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. You might want to read over the info at http://www.voip-info.org. There's a lot of good reading there that will help you make the most of Asterisk. Sean -----Original Message----- From: Paul Tyreman [mailto:paul@tyreman.org.uk] Sent: Sunday, April 18, 2004 1:31 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] x100p config Hi, I have just installed my first X100P card, and seams to be half working. 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. However, I have put this line in my extensions.conf: [incoming] exten => s,1,Dial(SIP/Phone1&SIP/Phone2,20,tr) 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 ? Also, I have put this in the same extensions.conf file: [outgoing] exten => _0X.,1,Dial,Zap/1/${EXTEN:1} [sip] include => outgoing Yet I still cannot make outgoing calls, when I dial 0 and the number I want to call on the public network. Any help would be great as I'm starting to pull my hair out ! Thanks, Paul. _______________________________________________ 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 _______________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040418/edec6a99/attachment.htm