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Newbee to Asterisk questionTechnical support, how-to guides, troubleshooting, and general assistance, from beginner to seasoned pro, this is where to discuss Asterisk, the most powerful open source PBX. |
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| It is the belief of most Asterisk users (that I've read about) that your Asterisk box should only be doing Asterisk and nothing else... Particularly for a business. Call quality is typically a big deal when in a business environment. If your * box is peforming double duties as web server, file server, whatever, that increases your chances of poor call quality. I highly recommend you read the Asterisk@Home Handbook found at http://asteriskathome.sourceforge.net/ It is best to at least assign it an internal static IP address, or if available, assign it a public IP address. Yes, you may have to have your router forward certain ports (i.e. 5060) to your Asterisk IP address. If all your calls come into your * box, then you won't need the SBC voicemail service. Unless, you exceed your call capacity. i.e. If you have 2 lines that are being used and 3rd call tries to come in, the 3rd call will never reach your * box, but it should go to the SBC voicemail. I'm a little confused about your existing setup (3 PSTN lines 1 DSL 2 business lines) so I can't comment on possible alternative solutions. |
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| these do not do much after the computer turns on it is expecting an asterisk username and password. I really like to get something working so I can learn more. I tried setting up a trunk and extension for FWD and configuring x-ten but I am not having much luck. the x-ten is not registering says contact administrator which is me. I do not even know what is a trunk. I would hardly say that asterisk could be setup by a newbee in a few hours. It's 2 days already and I am still experimenting. The instructions are not very clear in the handbook perhaps the handbook I have is for a different version I do not know it came with the package. Also the instructions on the FWD site are different and require editing various files. I found the place where I can get access to these filed but I am not finding the sections that need editing. Also it is not clear which set of instructions I should follow. I have no idea how some people here know so much about asterisk, its almost like you need to take a class in it to get it running. |
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| You have a lot going on there. First processing voice for phone calls is meant for more of a real time processing environment where operations complete in a set amount of time. In a multithreaded environment such as linux you can use it for much more than just asterisk. However, using an * box for anything other than * is not recommended as it slows the machine down adding what can amount to significant lag to the calls. You should dedicate a box to your asterisk setup for this reason as even fast machines can suffer from various device io slowdowns that will effect the overall performance. This is the methodology adopted by AAH as well. You would be better off using a static ip address as it will eliminate many issues that I'm not going into here. A dynamic IP will work but is not proffered. A good place to start would be www.voip-info.org they have a lot of resources dedicated to asterisk configurations and a lot of sample config files. Asterisk at home greatly simplifies the process using a nice graphical interface but having at the least a rudimentary understanding of what is under the hood will help you out a lot. If all else fails you can always outsource the planning and configuration of your setup and the investment will pay for itself by reducing office expenses in a very short period of time. |
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| OK I got the x-ten to register and it shows logged in but when I dial it says call not accepted. (I switched via netconfig to local static IP and entered my dsl provider's name server IP. ) Looks like I'm getting closer and the rest of the config problem is in asterisk. Also on the x-ten do I need to have anything in the outbound proxy or is it supposed to be blank? |
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| You're already starting to make progress. You still may not believe it, but the handbooks, forums, and lists are the best paths to success. First, about ports. As a rule, if you have a router that is not Symmetric NAT the only reason to ever manually forward ports through that router is to properly router unsolicited inbound packets. Registrations come in unsolicited from your various extensions to your Asterisk, so this must be provided for. Your Sipura, on the other hand, only has outbound sessions, registering and interacting with the service provider(s) it is configured for. Incoming calls to your Sipura are, in the eyes of your router, responses from your conversation with your service provider, and are negotiated automatically by the vast majority of NAT routers. You are smart to get your X-Lite working first. Get it registering reliably and get it making calls to the demo IVR. Then, get a second computer working with X-Lite and start making calls between the two computers. Then, a free service like FWD or SIPphone, and then on to the real world. As for outbound proxy on your X-Lite, it should not make a difference if it is present or absent. Asterisk is flexible, if nothing else. Good luck.
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