Ok, I've been messing with asterisk for the last 3 weeks and I just can't seem to figure this out. Everything I've read seems to state that asterisk will work "out of the box" with only minor config changes when being used only for SIP to SIP calls. The problem I am having is I cannot make outbound calls or receive incoming calls over my sip-provider. Asterisk registers properly, and internal communications seem to work fine. I have, at one time or another, had either outgoing only, or incoming only, but never both at once. Unfortunately, I didn't know what I did to make either of those work since I had made multiple adjustments and had done a reload after each change. For some reason, the incoming calls only started working after restarting the computer so it could have been any of 50 things I had changed. I am back to the sample config files. Is there any kind of walkthrough for a sip only setup? I have seen SIP only touched on briefly, with most of the documentation leaning torwards IAX communications. Here is what I am trying to accomplish: 1. Asterisk server registers with our sip-provider for sip to pstn local and international calls 2. Internal extensions 0, 200-210 can call eachother (of course) 3. Extensions 200-205 are in a Tech Support Queue 4. Extensions 206-210 are in a Customer Support Queue 5. Extension 0 is the operator or menu system (I guess this would be s?) 6. All phones (for now) are x-ten soft phones 7. Each extension has voice mail 8. When a customer calls during office hours, they are presented with a menu, press 1 for CS, press 2 for TS, or dial the extension you wish to reach, etc... 9. Calls can be forwarded to other extensions 10. On-hold music is implemented I can handle doing everything on the list except for #1. If anyone can offer any suggestions, it would make me, and my boss, very happy. Thanks in Advance Aaron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051026/991e25b9/attachment.htm
Carlos Alperin
2005-Oct-27 06:22 UTC
[Asterisk-Users] Simple SIP only Asterisk Configuration
That shouldn't be complicate, but it looks like you 're not registering with your provider. However, without the configuration files, it is not much to do for help you. Carlos Alperin _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Pikoro Sent: Wednesday, October 26, 2005 11:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Simple SIP only Asterisk Configuration Ok, I've been messing with asterisk for the last 3 weeks and I just can't seem to figure this out. Everything I've read seems to state that asterisk will work "out of the box" with only minor config changes when being used only for SIP to SIP calls. The problem I am having is I cannot make outbound calls or receive incoming calls over my sip-provider. Asterisk registers properly, and internal communications seem to work fine. I have, at one time or another, had either outgoing only, or incoming only, but never both at once. Unfortunately, I didn't know what I did to make either of those work since I had made multiple adjustments and had done a reload after each change. For some reason, the incoming calls only started working after restarting the computer so it could have been any of 50 things I had changed. I am back to the sample config files. Is there any kind of walkthrough for a sip only setup? I have seen SIP only touched on briefly, with most of the documentation leaning torwards IAX communications. Here is what I am trying to accomplish: 1. Asterisk server registers with our sip-provider for sip to pstn local and international calls 2. Internal extensions 0, 200-210 can call eachother (of course) 3. Extensions 200-205 are in a Tech Support Queue 4. Extensions 206-210 are in a Customer Support Queue 5. Extension 0 is the operator or menu system (I guess this would be s?) 6. All phones (for now) are x-ten soft phones 7. Each extension has voice mail 8. When a customer calls during office hours, they are presented with a menu, press 1 for CS, press 2 for TS, or dial the extension you wish to reach, etc... 9. Calls can be forwarded to other extensions 10. On-hold music is implemented I can handle doing everything on the list except for #1. If anyone can offer any suggestions, it would make me, and my boss, very happy. Thanks in Advance Aaron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051027/fe44a46d/attachment.htm