Re: [asterisk-users] Realtime: Should I say or should I go (now) ?
I'm working on a 500 seats Asterisk project. I'm wondering whether or not I should consider using Asterisk Realtime and a database to manage phones registrations. As far as stability goes, I've had no problem with realtime. In fact I've run a nationwide VoIP provider with asterisk using realtime. The main consideration I would make for using/not using it is how dynamic the system is. If you have 500 seats that aren't likely to change and aren't designing an interface to actually manage those users, then go static. Less complexity, less chance for failure (you don't have to worry about db servers, etc.). On the other hand, if it is for a situation where you are constantly adding phones, changing caller-ids, swapping phones and their extensions, etc. Realtime is the only way to go. And a free hint: if you are going to have to do anything that resembles number porting, swapping extensions, etc.--don't use extensions/phone numbers as SIP usernames. You have to regenerate config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. Terry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime: Should I say or should I go (now) ?
Terry Wilson wrote: And a free hint: if you are going to have to do anything that resembles number porting, swapping extensions, etc.--don't use extensions/phone numbers as SIP usernames. You have to regenerate config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. I second that emotion. I consult with a bunch of people who rolled their own Asterisk systems long ago, and when the try to virtualize their system in various ways they find their hands are tied. It is indescribably confusing once the number in sip.conf gets disengaged from the extensions in the dialplan. I wouldn't say to make the names meaningless, though; there are different ways to use those names so that they have useful meaning. Just don't make them extension numbers; it's like the TCP/IP boundary between layers. See SIP for an example of the problems such a thing can cause :-) B. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime: Should I say or should I go (now) ?
On Dec 21, 2007, at 1:29 PM, Brian Capouch wrote: Terry Wilson wrote: config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. I wouldn't say to make the names meaningless, though; there are different ways to use those names so that they have useful meaning. Just don't make them extension numbers; it's like the TCP/IP boundary between layers. See SIP for an example of the problems such a thing can cause :-) Since we were a nationwide VoIP provider and provisioned phones and had to deal with returns, etc. The SIP username really couldn't be used to identify anything. Since they were multi-line phones, etc. the only way we could have had a meaningful username would be possibly MACaddress-LineNumber or something. But if you have a normalized database, looking up user from SIP account becomes trivial. :-) ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime: Should I say or should I go (now) ?
And a free hint: if you are going to have to do anything that resembles number porting, swapping extensions, etc.--don't use extensions/phone numbers as SIP usernames. You have to regenerate config files, etc. Make your SIP usernames meaningless and use func_odbc to look up what extension is tied to which device. I second that emotion. I consult with a bunch of people who rolled their own Asterisk systems long ago, and when the try to virtualize their system in various ways they find their hands are tied. It is indescribably confusing once the number in sip.conf gets disengaged from the extensions in the dialplan. I wouldn't say to make the names meaningless, though; there are different ways to use those names so that they have useful meaning. Just don't make them extension numbers; it's like the TCP/IP boundary between layers. See SIP for an example of the problems such a thing can cause :-) B. Within my Realtime Asterisk Cluster, I use Directory Numbers (DN) for all sip/iax devices. These are a 5 or 6 digit number that don't mean a whole lot until I assign an extension to it in the dial plan. So DN 22331 could be exten 101 or exten 1001 and can be updated or changed to a different extension in the dial plan without having to update the device itself, unless the CID needs to be changed. You need very good record keeping to be successful. Also on the phone device, the auth name or account name may be 22331 but the display name will be 1001. To make this change, you need a central provisioning server, update the config file and reboot the phone to update the display name. Hope this helps and doesn't confuse things. JR -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Realtime: Should I say or should I go (now) ?
Hi, I'm working on a 500 seats Asterisk project. I'm wondering whether or not I should consider using Asterisk Realtime and a database to manage phones registrations. Stories in Dev mailing list say Realtime is mis-used or should be improved. So, what's the bottom line ? Can I consider anything I can do with .conf files can be done with a combination of .conf files and Realtime. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Realtime: Should I say or should I go (now) ?
I am using Realtime in virtually all my projects. So far, I haven't had any major issues. It saves a lot of headache for profile/dialplan updates, at least for me! So I say, GO! - Ben Olivier wrote: Hi, I'm working on a 500 seats Asterisk project. I'm wondering whether or not I should consider using Asterisk Realtime and a database to manage phones registrations. Stories in Dev mailing list say Realtime is mis-used or should be improved. So, what's the bottom line ? Can I consider anything I can do with .conf files can be done with a combination of .conf files and Realtime. Regards ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
