Re: [asterisk-users] 2x* and realtime
Is there a way to check if a peer is registered with the other box and forward the call there if a call comes in? Yes, you can (if nothing else, I'm fuzzy this morning) try forwarding the call and it will fail if the device is not registered because Asterisk will report it not found with a SIP 404 signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling Functions from AEL2
Douglas Garstang wrote:
I am trying to call the DUNDILOOKUP dialplan function from ael2, like this:
context route {
Set(PATH=${DUNDILOOKUP(${EXTEN},DUNDIRegistr)});
}
The DUNDILOOKUP function returns no data. However, when I call it exactly the
same way in a regular context, it DOES return data.
[route]
exten = _X.,n,Set(PATH=${DUNDILOOKUP(${EXTEN},DUNDIRegistr)})
That works. Could this possibly be an AEL2 bug? This is Asterisk 1.4 beta2.
Doug.
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I'll look into what I can find as soon as time permits (my company is
entering a beta release today), but my first suggestion to debug what's
happening is to do 'show dialplan route' and see what the compiled
dialplan shows. That could help you figure out if it's an AEL2 bug,
because it would show what the AEL2 compiler did with that line.
Hope this helps,
SKM
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[asterisk-users] AEL2 Catching on?
Is it just me or am I seeing more AEL2 code in people's examples? Could it be that AEL2 is starting to finally catch on? SKM -AEL2 Fanatic, Potato Eater, and General Lurker ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT But So Ungodly Important
Gents, First, let me apologize for cross-posting and for posting off topic. Cross post was only to reach members of one list that may not be on the other. Those of you that know me, know that I don't post off topic very often, let alone put out a list wide request for help. however, a client of mine is part of a rather large webhost company. You all may have read that there is a new exploit that can be run against users of outlook internet explorer, using VML. Once the exploit is run, it not only allows code to be run on the client machine, but apparently it spreads itself across servers connected to the now infected _SERVER_. This problem is currently spreading across a huge number of hosting companies, and I've been asked to use any and all contacts I can to get help with trying to find a resolution. Currently, verisign, paypal, and quite a few other companies are assisting us and others, but this is about to reach critical mass. If anyone thinks they may be able to help, please contact me ASAP. In case you don't know me by this email, but maybe by a previous list email (that is no longer used because I don't work there anymore), my previous list email(s) is/are: [EMAIL PROTECTED] [EMAIL PROTECTED] Thank you all for your consideration, and I must apologize profusely for needing to resort to these lists, but I don't have many other contacts I can connect with other than via this list. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT But So Ungodly Important
Gentlemen, An update on my prior post. I have not confirmed a solution is in place, but I do know that a gent has identified the virus, and symantec has confirmed it's new. I don't know the prefix, but it'll be named after my coworker.dcollins is the name it'll be under. I'll update if we have a fix, once I have confirmation -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-dev] Re: [asterisk-users] OT But So Ungodly Important
Please _don't_ ! I'm sympathetic to your situation, and we have all shouted for help in a crisis, but. This VectorGraphics+Javascript+IE+windows exploit has _no_ relevance to this group. The only way that it is related to a discussion of asterisk source code development would be if either: a) You are emphasizing the importance of safe coding style to protect against buffer overruns and can indicate somewhere in the asterisk codebase where you feel we should be doing better. b) The vulnerability in some way impacts directly on asterisk (say via the new http/manager stuff). There are several new vulnerabilities discovered each day, and there are lists for them. This isn't one. Tim Panton www.mexuar.com Mr Panton, I apologize, I intended to send that particular post to _only_ the users list, as an offering to anyone who may have needed the information. Also, let me apologize again for the original request for help. I only posted because I was asked to contact anyone and everyone I possibly could to offer payment for assistance. If I could tell you the name of the company I'm working for, you'd understand why it was an extremely large concern. Again, I apologize, I know that posting so seriously off topic is not condoned, and I have done my best to stick with this. Thank you all for your understanding, and hopefully your forgiveness. I've definitely been one of the first on various lists to bitch about non-compliance with the rules and standard etiquette of lists. Cheers all, I hope to have more on topic items to post about soon. In particular I've been playing with some mysqlsql addon code lately :) Sherwood McGowan signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT But So Ungodly Important
Just wanted to apologize again for the OT post. Also, I'll not be posting further on this subject. If you want fix information, contact me offlist and I'll forward any information I'm given by DCollins -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT But So Ungodly Important
calvis wrote: Have you check out http://www.f-secure.com/weblog/ to see if it is related your problem? They offer a few solutions. Charles Alvis Internet Technology Group, Inc. Redmond, WA Personal Blog http://www.spamspotter.com Thank you for the link! I've forwarded it to the admins attempting to battle the issue (I'm staying out of the way since I concentrate on voip-side issues). I'll post to the list if it helps, so that everyone can benefit. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to Use SER ?
Ryan Burke wrote: Thanks for the info. So it was really just one server that handled 2.5k user registrations and up to 500 concurrent calls? Do you remember anything about the codecs? Was there any transcoding done, music on hold, queues, etc? Usually for a dual Xeon 3Ghz people say they get about 250 concurrent calls and maybe 1k users registered before things start acting flaky. I really appreciate the info. I'm looking forward to hearing about your current project when you get a chance to write it up. Thanks again, Ryan Ryan, Thanks for your question, it made me remember a little more :) There was no music on hold, no transcoding, no queues, all lines used G.711u codec. Remember, this was a Internet Telephone Service Provider for residential business services, so queues, MOH, and the like were not implemented. -- S McGowan VoIP Consultant [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
good stuff mate. a few clarifications: you had static extensions.conf, realtime sipusers, etc, right? Also, abt features like call fwding, etc, which one is better, performance wise, using a mysql db, or use Asterisk's internal DB(berkeley db, isnt it?using those DBput n DBget ops)??Anyone's got any figures for these? This somewot spoils the fun in Asterisk, when talking of performance, to query the DB for every call . Sort of pulls things down. Any comments or observations guys? - Ben. Ben, Yes, static extensions.conf, realtime everything else. A realtime dialplan never made much sense to me, as the dialplan shouldn't (in my humble opinion) be that fluid anyway, it should be fairly static. In terms of spoiling the fun and/or performance issues, let me note that in my current implementation we not only have options being queried but also realtime billing, permissions, limits, and carrier/trunk performance data, all being pulled and calculated via the database. I also have handy little timers returning the length of time it takes to do the processing from request receipt to dial, and I'm still currently under 1-2 seconds for entire call preparation including all the logic that goes along with checking all features, the current account's account status, balance and limits, AND all parent accounts in it's billing chain. I haven't done a head to head with the berkley DB, but I think part of the reason it's so fast is due to the highly normalized database structure, which allows for efficient query design. It's not all third form, but almost there :D. I'm in the last days of ALPHA now with my current project. Once we launch BETA, which will be a semi-public testing by invitation (Murph, you still going to participate?), I should be able to find a few minutes to outline the design. One other quick thing, the berkley DB doesn't allow for clustering either, MySQL does. Very nice to have your database distributed across multiple nodes, makes for an easier time designing the failovers :D Cheers, Sherwood ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
I would like to know how you got Asterisk to function with 2500 SIP registrations. Did you have qualify enabled? Yes, qualify was enabled, using the standard length of qualification period between checks. Very few accounts had custom qualify settings. What about the 500 simultaneous calls? How many SQL hits were you doing (all said and done). Any performance logs from the SQL server? I can't believe you got all this running on one box! You have to remember, 500 simultaneous calls is not the same as something like 20 calls per second. some of those calls may have been quite long, and once the call's been placed, there's no database work being done until the call ends. I wish I had statistics from that setup, but I don't, we spent so much time implementing new features and chasing down problems caused by using a pre-RTA version of Asterisk with a patched in RTA setup. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
S McGowan, I don't know if you missed my question (from the slew of questions you've received and answered), but I was wondering about transcoding and PSTN channels. What kind of codecs were used and was there any transcoding happening? Was this box only responsible for VoIP-to-VoIP calls or was there also PSTN trunks as well? Again, I'm amazed by this example since it seems to be way over what anyone else normally reports as usable. Thanks again, Ryan Ryan, I answered, but for some reason this pop account tends to be strange... Anyway, we were not doing any transcoding and our PSTN connectivity was handled via a Tier 1 ISP that does SIP only PSTN connectivity solutions with G.711u. So, basically as far as Asterisk was concerned, there was SIP and RDP, that's all. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
Kristian Kielhofner wrote: Rushowr wrote: S McGowan, I don't know if you missed my question (from the slew of questions you've received and answered), but I was wondering about transcoding and PSTN channels. What kind of codecs were used and was there any transcoding happening? Was this box only responsible for VoIP-to-VoIP calls or was there also PSTN trunks as well? Again, I'm amazed by this example since it seems to be way over what anyone else normally reports as usable. Thanks again, Ryan Ryan, I answered, but for some reason this pop account tends to be strange... Anyway, we were not doing any transcoding and our PSTN connectivity was handled via a Tier 1 ISP that does SIP only PSTN connectivity solutions with G.711u. So, basically as far as Asterisk was concerned, there was SIP and RDP, that's all. So there was 2500 SIP registrations with qualify, 500 active calls with SIP and RTP, realtime, and CDR logging via MySQL (all on the same box)? What source changes did you make? What OS tweaks? -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users None, literally. CentOS 4.3, Asterisk Trunk that was updated practically weekly, at least on the dev box. The production server wouldn't get recompiled unless a fix was in from trunk. Incidentally, I just doublechecked my numbers with my former co-worker. He confirms we had roughly the following numbers/setup: *2,500 registered SIP users, 95% being qualified by Asterisk *Max of 300 concurrent calls, with about half that on average (my mistake earlier with the 500 estimation) *Realtime SIP Users/Peers, Voicemail, and dialplan calls *Static extensions with MySQL queries for data retrieval/manipulation *NO Reinvites allowed due to the fact that most clients were residential behind NAT. Hardware: CPU:Dual 3Ghz XEON RAM:2GB RAM -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
adebayo omo-dare wrote: Hi Sheerwood, I unfortunately saw a bit of what I percieve to be an error in what you said. BerkeleyDB does in fact support replication across nodes - see: http://www.sleepycat.com/docs/ref/rep/intro.html - possibly you meant to say the version implemented in * does not support replication. If so, I do appoligise for being a little pedantic. I have only just started to look at *'s code - so what I say further is with a great deal of hesitation when directly referenced to *. However, I work with both Berkely (on a programming level) and MySQL in a telecom (soft-switch) environment. In terms of performance (judged as speed), a comparison between MySQL and Berkeley would be like comparing a top of the range Mercedes to an F1 racing car. Overheads from MySQL come in the form of SQL translation, use of Sockets, etc... This is in addition to its size. Yet, the choice between the two, is a lot more complex, IMHO, than mereley thinking in terms of performance. And possible High Availability solutions, in their own rights, taking in to consideration that * will be working in concert with numerous other environments, programmes and requirments, are diverse enough to make each deployment a little unique - thereby making each option a potential liability. One rule of thumb for us has always been - if you need raw speed, and intend to deal with the data in a very restricted/rigid/well defined manner - opt for Berkeley. But if you want a great deal of fluidity, and intend, or may at some time intend, for that data to interact with other applications and potential requirements - Opt MySQL. It is possibly also best to work with what you feel most comfortable with first and then experiment to see if you may require the services of the other. ps. In terms of querying a DB for every call, I would presume that a DB is an active and fragile thing and the provision of ACID ensures that everything that occurs with it does so with a certain measure of safety. In fact, due to the random manner of requests, you will find it, in complete terms, actually performs a lot better than any other form of retrieval. Hope this, in some manner, helps Bayo Bayo, Thanks for your input! I was actually not aware that Berkley DB allowed replication. The primary reason for using MySQL (and PostgreSQL in some of my other projects) is the ease with which you can have the data used in other systems. Thanks again for your input :) signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
Kristian Kielhofner wrote: [EMAIL PROTECTED] wrote: Again, I'm amazed by this example since it seems to be way over what anyone else normally reports as usable. Exactly! -- Kristian Kielhofner ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well, first of all, how many people are attempting to run an entire ITSP off of Asterisk? Also, how many of those reports you refer to are SIP only? There's amazingly little system utilization when all you're doing is processing the call requests and logging CDRs? Past that, I can't say anything more to convince you all that this is true without sounding like I'm trying too hard for belief. I'd release the name of the ITSP (some of you may remember it anyway) that I used to work for, and once the one I currently work for goes ahead into public release I'll be more than glad to share the name. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ANI and Meetme...
Natambu Obleton wrote: Ok. First question is how to make it say my number back. Like if you call extension 1000 from extension 1001, I want it to say “Number is 1,0,0,1” like an ANI number? Help. Also I want to setup a meetme conference so that it asks “Enter conference number” then execute meetme($entered_number) I feel dumb asking because these sound like they should be so easy, but I can’t find any help with this. Thanks. Natambu Obleton Network Engineer FastTrack Communications [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] (970) 247-3366 office (970) 247-2426 fax ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The first item (repeating ANI number): ; Use the saydigits app to repeat the ANI to the caller exten = _X.,1,Answer() exten = _X.,n,Wait(2) exten = _X.,n,SayDigits(CALLERID(ani)) Cheers -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to Use SER ?
Marco Mouta wrote: Hi all, I'm planing to develop a solution based on Asterisk for about 300 users. My question now is, do I really need to use openSER as the sip proxy and Asterisk for the PBX functions? Can i trust in a solution only with Asterisk to make all this install? Please help me with your experience on this kind of asterisk solutions. I've googled and read about asterisk at large scale solutions, but still in doubt. http://www.voip-info.org/wiki-Asterisk+at+large -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In my experience, yes you can use *just* asterisk for the implementation of a large scale setup, you just better be sure you've planned it out well. I've set up a few large scale Asterisk implementations, covering more than 1K users on a single box. And that was in 2005 using trunk. There were problems, but all in all it was (and is, for the former client) not a bad implementation. If you're just looking at a large PBX install, you're definitely fine with a well planned system. Just my $0.02, not to be taken as a guarantee ;-) -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
Benjamin Jacob wrote: Rushowr wrote: ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Date: Tue, 19 Sep 2006 05:52:52 -0500 Marco Mouta wrote: Hi all, I'm planing to develop a solution based on Asterisk for about 300 users. My question now is, do I really need to use openSER as the sip proxy and Asterisk for the PBX functions? Can i trust in a solution only with Asterisk to make all this install? Please help me with your experience on this kind of asterisk solutions. I've googled and read about asterisk at large scale solutions, but still in doubt. http://www.voip-info.org/wiki-Asterisk+at+large -- Com os melhores cumprimentos, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users In my experience, yes you can use *just* asterisk for the implementation of a large scale setup, you just better be sure you've planned it out well. I've set up a few large scale Asterisk implementations, covering more than 1K users on a single box. And that was in 2005 using trunk. There were problems, but all in all it was (and is, for the former client) not a bad implementation. If you're just looking at a large PBX install, you're definitely fine with a well planned system. Just my $0.02, not to be taken as a guarantee ;-) You mean, 1 K simultaneous calls? or 1 K registered users(yes yes. this is the one!!)??? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Sorry, should have been a little more specific. I've had Asterisk running realtime SIP users/peers and realtime sql calls from the dialplan (all with MySQL), and have had around 2.5k registered users and a peak (that I recall) of around 500 concurrent calls. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
Ryan wrote: Can you explain your design in a little more detail? What kind of hardware did you use to get over 1k users on a single box and 500 concurrent calls? Sounds like a very interesting medium-large scale implementation that others could learn from. thanks, Ryan I'll do the best I can from memory and without violating confidentiality :) The build was for a startup ITSP and was the first of that scale that either myself or my associate who worked for the client had done. The hardware was something along these lines, but I cannot be absolutely sure: 3Ghz Dual XEON CPU 1GB RAM 2 1Gb NICs I dont remember the hard drive specs at all, but that's more elementary anyway. We initially set up the systems with CentOS 4.2 or 4.3, can't remember. MySQL 4.x (latest 4.x version from summer 2005) Asterisk HEAD (constantly updating and recompiling, at the time the realtime arch wasn't fully in place) MySQL addons package Realtime SIP clients Statically configured SIP trunks, which provided our PSTN connections. I cannot disclose the company, but the trunk provider is/was extremely huge, a Tier 1 ISP. MySQL CDRs (the cdr addon) User options and feature controls accessed in realtime via a MySQL table designated for the purpose (basically an options table, with things like call_forward (y/n) columns). LOTS of custom monitoring done in regards to Asterisk status information Custom PHP/MySQL/Apache web interface for provisioning, configuration, and general administration written by yours truly, including polling Asterisk for the status of a client UA when that client's config is being viewed, provisioning (TFTP) handlers, etc... Hope this is a good start, anything else you want to know, I'll do my best. Also, once I finish my latest ITSP launch project, I'll be able to (hopefully) give a better example, one with failover, custom CDRs, custom LeastCost+BestPerformance routing, etc...etc... Even realtime billing, which the previous client didn't have, AND reseller support at the ITSP levelcan't say more yet, but it'll be rather huge I'm sure. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: FollowMe question
Hall, Eric M. wrote: I got the config working. Not sure if someone has pre-recorded sounds for this app or not. Looked all over for them and I'm unable to locate them.If anyone has sound file they would like to share that would help me greatly. Thanks *Sent:* Friday, September 15, 2006 5:23 PM *To:* 'asterisk-users@lists.digium.com' *Subject:* FollowMe question Group Does anyone have the FollowMe sound files? Do I need to record them? Also does anyone have a working followme.conf file that they would share? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I wouldn't mind a shot at creating the sound files in my little studio here. Just give me a set of prompts/messages to record and I'll contribute :) -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] When does Scalability requests Asterisk to U se SER ?
[EMAIL PROTECTED] wrote: Can you explain your design in a little more detail? What kind of hardware did you use to get over 1k users on a single box and 500 concurrent calls? Sounds like a very interesting medium-large scale implementation that others could learn from. thanks, Ryan (NOTE: I sent the original reply about 3 hours ago and have not seen it post, so I'm resending. I apologize for any double receipts of the message.) I'll do the best I can from memory and without violating confidentiality :) The build was for a startup ITSP and was the first of that scale that either myself or my associate who worked for the client had done. The hardware was something along these lines, but I cannot be absolutely sure: 3Ghz Dual XEON CPU 1GB RAM 2 1Gb NICs I dont remember the hard drive specs at all, but that's more elementary anyway. We initially set up the systems with CentOS 4.2 or 4.3, can't remember. MySQL 4.x (latest 4.x version from summer 2005) Asterisk HEAD (constantly updating and recompiling, at the time the realtime arch wasn't fully in place) MySQL addons package Realtime SIP clients Statically configured SIP trunks, which provided our PSTN connections. I cannot disclose the company, but the trunk provider is/was extremely huge, a Tier 1 ISP. MySQL CDRs (the cdr addon) User options and feature controls accessed in realtime via a MySQL table designated for the purpose (basically an options table, with things like call_forward (y/n) columns). LOTS of custom monitoring done in regards to Asterisk status information Custom PHP/MySQL/Apache web interface for provisioning, configuration, and general administration written by yours truly, including polling Asterisk for the status of a client UA when that client's config is being viewed, provisioning (TFTP) handlers, etc... Hope this is a good start, anything else you want to know, I'll do my best. Also, once I finish my latest ITSP launch project, I'll be able to (hopefully) give a better example, one with failover, custom CDRs, custom LeastCost+BestPerformance routing, etc...etc... Even realtime billing, which the previous client didn't have, AND reseller support at the ITSP levelcan't say more yet, but it'll be rather huge I'm sure. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fedora
bilal ghayyad wrote: Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hrmwell, first, you should think about using CentOS, as Fedora is the development branch. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using residential voip for business?
Rich Adamson wrote: Rushowr wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Christopher Corn wrote: thanks for the reply. why are residential lines cheaper than businesses? say for unlimited, it always costs more for residential. */Michael Graves [EMAIL PROTECTED]/* wrote: I'd just use a service that's being offered to business customers...like Nuvio's nPBX. While they don't support Asterisk directly some of their resellers will support using *. I've used it for about 6 months and its been very reliable. The only annoying thing is that they only support SIP connections. The rumour is that they may eventually offer an IAX2 based account for Asterisk users...but I've not yet heard if this is actually going to happen. FWIW, I ported my DIDs to Nuvio so that's where my incomming calls come from. I split my outgoing calls across Nuvio, Nufone Voxee. Michael --Original Message Text--- *From:* Christopher Corn *Date:* Sun, 10 Sep 2006 17:20:37 -0700 (PDT) i see. thanks for the info. */[EMAIL PROTECTED]/* wrote: Its a trickish business, when they say unlimited and you make more than 2500 minutes they cut you off. -- Original message -- From: Christopher Corn [EMAIL PROTECTED] I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it for a business and it is over-utilized, it will be canceled. is this true for all residential voip plans? i have a small office of about 4 or 5 phones. i tend to chose residential plans because they have the unlimited offer for outgoing/incoming. thx Typically a business offering costs more because the provider offers higher availability, reliability, call quality, etc... That's not true at all. I worked for a large telco for 20+ years (in all engineering disciplines), and the only reason business plans are more expensive then residential plans is that businesses generate more traffic. More traffic translates into more infrastructure costs (eg, central office equipment, trunks, etc). Businesses and homes generally use cable pairs (or fiber) out of the same cable, use the same central office line cards, etc. There is no difference in terms of availability, reliability or call quality. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You're mistaking VOIP for PSTN. Yes, businesses generate more traffic, but they also are bigger support customers and are far more willing to bitch over every single penny they think they should be discounted. Businesses cost moreperiod. More support staff needed, more reliability needed (unless you want your biz products to be blackballed as total shit), etc... Not only that, but let's face it gents, when you're going to be making money off of a product or service, they cost more. Everyone's more than happy to dig into the pockets of someone who's making money off of their services. Cheers -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK
Re: [asterisk-users] Asterisk Realtime Arch - static or realtime?
Benjamin Jacob wrote: Hello ppl, Wanted to know your experiences, if you've worked with Asterisk Realtime Architecture. Which one do you prefer, static or realtime? I personaly think, the static architecture is a better solution, cuz, in the realtime config, to check the dialplan(n hence the sql database) for each and every call, will be expensive. With the static architecture, you reload only when required(which practicaly will not be happening too often). Care to share your experiences? cheerz Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Personally, I use realtime for a LOT of the configuration and whatnot, but I don't touch the dialplan with RT. That being said, I'm personally fairly happy with the realtime support. Example system: Recent Trunk of both Asterisk and MySQL addons, large AEL2 dialplan, primarily SIP system. -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Realtime Arch - static or realtime?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Benjamin Jacob wrote: Rushowr wrote: Benjamin Jacob wrote: Hello ppl, Wanted to know your experiences, if you've worked with Asterisk Realtime Architecture. Which one do you prefer, static or realtime? I personaly think, the static architecture is a better solution, cuz, in the realtime config, to check the dialplan(n hence the sql database) for each and every call, will be expensive. With the static architecture, you reload only when required(which practicaly will not be happening too often). Care to share your experiences? cheerz Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Personally, I use realtime for a LOT of the configuration and whatnot, but I don't touch the dialplan with RT. That being said, I'm personally fairly happy with the realtime support. Example system: Recent Trunk of both Asterisk and MySQL addons, large AEL2 dialplan, primarily SIP system. So you are using the flat file extensions.conf for your dialplans?? If so, isn't it a pain to keep editing that file, when the need arises? I too liked Realtime, for sip friends etc. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No, mainly because I work mostly on huge projects, and not on PBX's. Even with doing PBX setups, however, I don't find the need to modify the dialplan that often. Plus, the RT dialplan setups still follow the numbered execution model don't they? I work in AEL. I can drop a line here, add a line there, and I don't have to doublecheck all my logic routing to make sure I didn't miss a call somewhere. :-) - -- S McGowan VoIP Consultant [EMAIL PROTECTED] - -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg - -END PGP PUBLIC KEY BLOCK- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) iD8DBQFFBVamlfQsv7FBhp8RAiyCAJ0Vzd1Jj3tMOob1MV0+pWPuzP++BQCglrZB YWEcRF901wp8qSmbrYH+mLI= =cwrC -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Static RealTime - SIP.CONF
Hugo wrote: Anyone could help to use Static RealTime with SIP.CONF. I use Dynamic Realtime successfully. In fact, I want to know how to compos the correct DB(postgres or mysql) fields (I think STATIC configuration is different from DYNAMIC). Regards, Hugo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users www.voip-info.org is an amazing tool, and it's referenced FREQUENTLY. 10 seconds in a web browser brought me this link: http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip Which, amazingly enough, contains information about setting up the tables for RealTime Sip -- S McGowan VoIP Consultant [EMAIL PROTECTED] -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg -END PGP PUBLIC KEY BLOCK- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip peer question
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dijkstra, Roelof wrote: Hello, We currenty have an asterisk cluster running, with a quad PRI and a quad BRI. This all works pretty well. What i was wondering: If i do a show sip peers I see all the ip addresses of the phones that registered, also, when restarting the server. Is there any way of copying this information to another server? Regards, Roelof Dijkstra Network Engineer EMEA Compuware Europe BV The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yes, basically you need to servers to share the same database source. If you're not using MySQL, ODBC, PostgreSQL, etc, you may want to look into it, I personally don't remember if you can share the astdb SKM -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: ENCRYPTED WITH GPG iD8DBQFFAWmmlfQsv7FBhp8RAnX6AKCyc8rOYiAwdoLSIGgSVMBAT0zEfACgr+lZ R1V9XzWjKDBQajegJ+qwsdI= =61P6 -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using residential voip for business?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Christopher Corn wrote: thanks for the reply. why are residential lines cheaper than businesses? say for unlimited, it always costs more for residential. */Michael Graves [EMAIL PROTECTED]/* wrote: I'd just use a service that's being offered to business customers...like Nuvio's nPBX. While they don't support Asterisk directly some of their resellers will support using *. I've used it for about 6 months and its been very reliable. The only annoying thing is that they only support SIP connections. The rumour is that they may eventually offer an IAX2 based account for Asterisk users...but I've not yet heard if this is actually going to happen. FWIW, I ported my DIDs to Nuvio so that's where my incomming calls come from. I split my outgoing calls across Nuvio, Nufone Voxee. Michael --Original Message Text--- *From:* Christopher Corn *Date:* Sun, 10 Sep 2006 17:20:37 -0700 (PDT) i see. thanks for the info. */[EMAIL PROTECTED]/* wrote: Its a trickish business, when they say unlimited and you make more than 2500 minutes they cut you off. -- Original message -- From: Christopher Corn [EMAIL PROTECTED] I spoke to a voip provider today who mentioned that though they offer an unlimited plan, if we use it for a business and it is over-utilized, it will be canceled. is this true for all residential voip plans? i have a small office of about 4 or 5 phones. i tend to chose residential plans because they have the unlimited offer for outgoing/incoming. thx From: Christopher Corn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] using residential voip for business? Date: Sun, 10 Sep 2006 23:42:58 + ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --0-531322068-1157934037=:7889-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Typically a business offering costs more because the provider offers higher availability, reliability, call quality, etc... - -- S McGowan VoIP Consultant [EMAIL PROTECTED] - -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg - -END PGP PUBLIC KEY BLOCK- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32)
Re: [asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 RR wrote: I am currently running this with UnixODBC - FreeTDS - MSSQL Server 2K ( please don't hate me for using an 'evil empire' product amongst the pure sanctity of open source :D). But the results are, well...So far so good. But I can't say much because the most i've tried is 4 concurrent connections to the DB for users trying to access their voicemails but it does well. All I do is voicemail and conference so my extensions.conf is literally 20-30 lines. Would love to put a few hundred users come in to see what breaks first. Would also love to hear other people's experiences. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Well RR, you may at least get to live vicariously... The implementation I'm testing this for is ITSP level setup. Last ITSP I did work for was running over 2K users on a box with averages of 200-250 concurrent calls at any given time and they only did end-user implementation. This company I'm with now is doing reseller accounts and possibly wholesaling at some point, and they have a fairly large group of customers who buy other reseller products from them, so I'm confident we'll get stress tested HARD. :-) I'll do my best to post my experiences either on the list or online. Personally I'd like to see more information on the IMAP setups, there's little ODBC and even less IMAP implementation docs out there. I'm mildly afraid to use the IMAP setup because I'm worried about the user setups, but can't find anything on it Cheers! SKM -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: ENCRYPTED WITH GPG iD8DBQFFARlolfQsv7FBhp8RAkBLAJ44OhHYZuv0gxThtUGLnw1wW4EMHgCfS0QP K6lMbrGg0lu04qeERMp9NqE= =bYYA -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What don't I get about SIP?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mike wrote: Thanks Tim. I've been trying to find out what's happening. Basically, somehow, it seems that my Polycom 501 knows what extensions are valid and which aren't in my dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems to return it this info (sort of :valid, invalid or could be valid, need more digits to know) when I press send. I know it sounds mad, and I would love nothing more than being told I am an idiot because or x and y. Why do I feel that this info is passed from Asterisk to the 501? Well, take the following (very simple) dialplan [context_a] Exten = 1234,1,Noop(foo) Exten = _9,1,Noop(bar) Exten = i,1,Noop(invalid) What happens when I dial out is the following: 1) 1234: Noop(foo) ; good 2) 4: A congestion tone is heard from the phone (but Asterisk's CLI doesn't show anything...no sent into invalid extension '4' in context 'context_a', but no invalid handler 3) 934 : It's invalid, but it could match the pattern is I added some digits. I expect an invalid extension message, but what actually happens is the phone tries the send something (I can see an icon moving on the phone) but the phone stays quiet (no stuttering tone or whatever). It waits, I can input more digits on the phone. Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. Pierre Sent: September 8, 2006 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid try and let Asterisk handle the error ?(instead of the phone trying to do what it think is best and handling the error on it's own). Is there an Asterisk setting for that? Failing that, is there a Polycom setting to disable this intelligent error handling? Mike -- Tim St. Pierre IP telephony specialist sip://[EMAIL PROTECTED] Toronto: 647 722 6930 Toll-Free 1 888 488 6940 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Silly idea, why don't you sniff the packets being sent over port 5060? You'll be able to verify the conversation taking place. - -- S McGowan VoIP Consultant [EMAIL PROTECTED] - -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW
Re: [asterisk-users] What don't I get about SIP?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk Mike Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 192.168.1.200 : 5060 (NAT) Found user '000f42056d58-1' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.1.200:2228 Found description format PCMU Found description format PCMA Found description format G729 Found description format telephone-event Capabilities: us - 0x106 (gsm|ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 965 in context_a (domain test.test.ca) Reliably Transmitting (NAT) to 45.67.312.45:5060: SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 192.168.1.200;branch=z9hG4bK93732511F5970F9E;received=45.67.312.45 From: CAP sip:[EMAIL PROTECTED];tag=DAD6C20C-68263D4F To: sip:[EMAIL PROTECTED];user=phone;tag=as4db2b55c Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rushowr Sent: September 8, 2006 4:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: Thanks Tim. I've been trying to find out what's happening. Basically, somehow, it seems that my Polycom 501 knows what extensions are valid and which aren't in my dialplan. Obviously, the 501 doesn't really know that, but Asterisk seems to return it this info (sort of :valid, invalid or could be valid, need more digits to know) when I press send. I know it sounds mad, and I would love nothing more than being told I am an idiot because or x and y. Why do I feel that this info is passed from Asterisk to the 501? Well, take the following (very simple) dialplan [context_a] Exten = 1234,1,Noop(foo) Exten = _9,1,Noop(bar) Exten = i,1,Noop(invalid) What happens when I dial out is the following: 1) 1234: Noop(foo) ; good 2) 4: A congestion tone is heard from the phone (but Asterisk's CLI doesn't show anything...no sent into invalid extension '4' in context 'context_a', but no invalid handler 3) 934 : It's invalid, but it could match the pattern is I added some digits. I expect an invalid extension message, but what actually happens is the phone tries the send something (I can see an icon moving on the phone) but the phone stays quiet (no stuttering tone or whatever). It waits, I can input more digits on the phone. Let's just take 1) and 2). Why is Asterisk not going into the i extension like it should? Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim St. Pierre Sent: September 8, 2006 2:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? With SIP, asterisk processes the digits it receives in the invite from the Polycom. There is no communication of dialplan information in SIP. The polycom should send the digits as soon as you press dial. You can program the polycom with a dialplan that will tell it when to send the digits, but that only works if you dial off-hook. I like on hook dialling, since it sends what i tell it, when I tell it. This should never happen when you press dial - it should try right away. My 301 does this, maybe they changed something in the newer firmware? -Tim On September 8, 2006 14:33, Mike wrote: I've been running into an issue with my Polycom 501 and Asterisk. I realized, after much mucking around, that when I dial a number (and press the send key) that is invalid , but could still match an Asterisk pattern (example: I dial 567, which is not a valid extension, but my diaplan accepts _567 as a pattern) instead of sending the call as is and ultimately failing, the phone is intelligent enough to sit and wait for extra digits in case I meant to dial 567111. Now thats a problem for me. How can I make Asterisk (or the 501) treat the attempted extension 567 as a valid
Re: [asterisk-users] What don't I get about SIP?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mike wrote: But that's the whole freaking problem!!! If I could do that, I would. But Asterisk keeps on sending the 484 Address incomplete message, and the Polycom keeps on waiting silently and patiently for me to put in the needed extra digit(s). When I pick up my home phone, and I forget a number, the phone company does wait a few seconds for the last digit. But there is a timeout, and eventually I get a fast busy. That`s what I want. And apparently, I can`t get that. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 6:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Not much you can do about that other than: exten = _X.,1,Playback(dial-real-number-you-moron) exten = _X.,2,Hangup Mike wrote: That's a good idea, and I tried, but as far as I know the digitmap setting of the Polycom allows me to enable the phone to dial automatically after a pattern is used (ex : [9]xx), but it doesn’t allow me to consider a too short string as being invalid (ex if I miss a digit and just dial 9-555-55- and then press send. Am I wrong? Cause did try the above example, and I got a 484 response back... Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: September 8, 2006 5:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] What don't I get about SIP? Mike wrote: It's not a silly idea, I've been doing some sniffing and debugging with my limited knowledge of the whole process. I found this in the debug stream after having dialed 965). Notice this line: SIP/2.0 484 Address Incomplete. Is this what I was suspecting, that it knows it could match a pattern (_9X) with a few more digits and so waiting for those digits from the user? How can I disable this or turn it off? The Polycom 501 supports 484 responses, but how can I either: 1) Disable it in the phone 2) Disable it in Asterisk I didn't even know that Polycom supported 484. Update the dialplan on your Polycom to make sure it will never send a partial number. You will no longer have to press Dial then either. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It's actually your phone's responsibility to respond to the 484 and/or a dial timeout. ;-) - -- S McGowan VoIP Consultant [EMAIL PROTECTED] - -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb
Re: [asterisk-users] Auto Dialer question
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hall, Eric M. wrote: Hello group I have a customer that has asked me to build an auto dialer that will call customer a few day before an appt and remind them of the time and date of the appt. Does anyone have any good links for apps that could do this type of auto calling? They also request that information be pulled from a database and be able to pull reports on who was called and if they call was picked up. Thanks for any help the group could give me! Eric ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Creative use of cron and some simple bash dialplan scripting will do this fairly easily. - -- S McGowan VoIP Consultant [EMAIL PROTECTED] - -BEGIN PGP PUBLIC KEY BLOCK- Version: GnuPG v1.4.5 (MingW32) - WinPT 0.12.3 mQGiBETxLJERBACrFvzk3Hd8AO9aGCSSgoabp8GGS7jYhR1UP9zqYeJIHeH+/r/D sCL0mPUGX1+FnVlh5UAO0Q3hueCdtgbAhdqMJMDhjQ2Tm10kBWu2DjWrLVnGx0QD Id1XAiQ1WIJkE2VqphKD0WVMsyxj08w+o+DwjD+mu3GCgitRTVOB9OnzpwCg3Ynx BHlbNUzLTp+3oUuudndpaiEEAIlBCJoIg+zCTg4/kFjsWfSYo3kTwNoQPqqMINMe GM15CkRvXgUdMgJMPeEqXNmfnUUHNf/6KD2WpP5kJcBZdNWHicvS+A+P1Sjuybio 5XlJgMDW5tzCX0V45n+RgZQjHMg1wpcv0eVOMhmaSL4eC7MyUnZBHzuBYmgNMpiM EF2wA/4y+hhoZ2SYUzTWk4QUPL8yaHTNS/4/aH8AB5cyRNljqT5//AXzYF3AxMZX bslWy4MtzX9CI9Zg8hxIzcaYp/oeFSVrv6Or/8ZRQk2T+eB7ymPY6T+SOcKfTgR2 f9kzlxtPjRK/nXDovjaaOGl0U0NaPemB0w8fEuNkF4LxKdAea7QgUyBNY0dvd2Fu IDxydXNob3dyQHBocmVha2VyLm5ldD6IYAQTEQIAIAUCRPEskQIbAwYLCQgHAwIE FQIIAwQWAgMBAh4BAheAAAoJEJX0LL+xQYafrbQAoKFzcLsRIkXWL1wzldi2iG4l FHD/AKCguGXH7GtZKpQfFct6vQUOnJuUB7kCDQRE8SygEAgAlOYMwiFKPALEpi/X Cb3kTzpDqi9yvlijssnyxY2IxTYJHheE2dkITtdmgFlfud0lCLiSVhf8i9Y2YCar I+Djz7/LTlX4lhcDBeAaSHfDUtr5jTn3caK5A3inCAxoI7Um9Sy3fSyW9DMww2Mj t+ysQ2XuXpRZ984/3X79kNttae7L3FqASHjfflUFhBukxpSAn5evmkAnmZDhjy5a Z9Ut+DGDQOG2qvDTZM/RFDyodLIRoW9AK2O3A7CtVjZVOTSjDdhdOsHzsuBioh51 ngfUo4B3hDy+tv5qtzD5UjVj8g+oFqDpjo7mj7EwhD/AqHxg6yKqOtVLTmeEdZzW RMMGkwADBQgAjutKcj73K0GqhlKP3D3plXXBLOeAnoUBMoxbd7u7HigTXkTeq7gX c+zC6pu3atL1piRBOTYPiflf36hkph+EC9Zu7fBmaIdKRqltV9m+XB5l6Kw/C4go hTeLFI5A61GmiyQ5NPRpaeERGba+EoWswYIUxkCmr7I02DL8R72oLu6bb+bevCz5 d1AKrY2Vg3M8IXhGHPrYoFup6EYC6Thp2wRG4vBtpQStFbdYjXNBYmwWNERPzOzb k3pU8y96X7mqLHbv6gi5wapJyPidasc3VtU7RrwSEsYDoc2nf+6KzZMTT3rnB9RL gns2mcXM/4utmBWzSL7tnil5mlI9dynHQYhJBBgRAgAJBQJE8SygAhsMAAoJEJX0 LL+xQYafclwAnAmrmJpITi7ngFNR/obx/l6tNPRqAJ477VYqaBg58lc+TlGK1DoA HeMrow== =GJrg - -END PGP PUBLIC KEY BLOCK- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) iD8DBQFFAjPwlfQsv7FBhp8RAll4AJ47wfZpO4TxEXPxUip35ga9TVcOdQCgrYKQ n4ulcCLaoFy0PtjuiBt2GdI= =y2hr -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Response to KP Flemming...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Joe Shmoe wrote: You say its not your code. But yet, why would you actually admit to one of your own leaking it. Well some research has been done one the code.. here's what we found.. the g723.1 library code that was posted matches the library code distributed by Digium and committed to CVS by Mark in March 2003, with the one exception that the tab_lpc.c file that was distributed by the poster had CRLF line endings in it, where the one from Digium CVS had only LF endings. The module code was identical to: http://svn.digium.com/view/asterisk/trunk/codecs/codec_g723_1.c?rev=5869view=markup Also if you want to know if Digium fully complies the the GPL no. They dont. Digium has added a paragraph of text under the symbol ASTERISK_GPL_KEY in include/asterisk/module.h which every Asterisk module must return when a function *key() is called by the module loader. This paragraph makes a claim that modules must only be released under the GPL license, not any other license, which excludes GPL compatible licensing and thereby constitutes an additional restriction which is explicitly prohibited by section 7 of the GPL. see http://www.eff.org/legal/cases/Lexmark_v_Static_Control/20041026_Ruling.pdf for additional information on this type of activity and generally why that paragraph cant even be legally copyrighted (at least in America, where digium is based). Missed the link for the Codec's? Here ya go! http://s6.quicksharing.com/v/6876458/_codec.tgz.html __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users *sniffsniff* I smell a Troll(yes I know I fed it, but c'mon, that was funny) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: ENCRYPTED WITH GPG iD8DBQFFAA4GlfQsv7FBhp8RAnFfAKC1gyqKQna37OOye4a51u8X4ii+yQCggPO1 ygyyN4k+I7orGvq7++0ChMs= =aWgp -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Experiences, Tips on Voicemail storage using ODBC or IMAP?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey all, I'm looking into setting up a system or two with either IMAP or ODBC storage of Voicemail messages and wanted to hear about your experiences, gather tips or warnings, etc, before I go diving too deep into it. Are either of those storage methods working reliably for any of you? What are some of the issues you had to deal with when setting it up? What's the performance like? You get the general idea... Quick stats on base test systems: Latest SVN trunk as of this morning Gentoo MySQL 5.0 Realtime sip and iax peers/users Realtime sip/iax/voicemail config LARGE dialplan Thanks in advance for any input, SKM -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: ENCRYPTED WITH GPG iD4DBQFFABL5lfQsv7FBhp8RAvYiAJjevWSJt1CaFGtnFe7qP8gnMQ3GAJ9WQs8O Pf6NQSvMmRzS6Y9Rc+tNdQ== =sMwA -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conditional IF based on IP address?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Hsieh wrote:
Greetings,
Is it possible to create a conditional IF inside extensions.conf based
on the source IP address of a SIP phone (as opposed to extension)? What
I would like to do is the following:
1. If SIP phone IP belongs to 192.168.0.0/24 http://192.168.0.0/24
subnet, set CALLERID=
2. Else, set CALLERID=
Thanks in advance for any examples or help.
Steve
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Steve, yes you can do it.
You'll need to use the SIPPEER function (available in trunk only I
believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If() to do
what you want.
SKM
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Re: [asterisk-users] Conditional IF based on IP address?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rushowr wrote:
Steve Hsieh wrote:
Greetings,
Is it possible to create a conditional IF inside extensions.conf based
on the source IP address of a SIP phone (as opposed to extension)? What
I would like to do is the following:
1. If SIP phone IP belongs to 192.168.0.0/24 http://192.168.0.0/24
subnet, set CALLERID=
2. Else, set CALLERID=
Thanks in advance for any examples or help.
Steve
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Steve, yes you can do it.
You'll need to use the SIPPEER function (available in trunk only I
believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If() to do
what you want.
SKM
I'm sorry, you need ${SIPPEER(${EXTEN}|ipaddr)} to retrieve the data you
want
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=YU5E
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Re: [asterisk-users] Conditional IF based on IP address?
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Steve Hsieh wrote:
Thanks, Russ!
Any suggestions on how to apply a subnet mask so that I can match an IP
that belongs to 192.168.0.0/23 http://192.168.0.0/23, for example? Or
would the only way be to match the string using REGEX?
On 9/6/06, *Rushowr* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Rushowr wrote:
Steve Hsieh wrote:
Greetings,
Is it possible to create a conditional IF inside extensions.conf
based
on the source IP address of a SIP phone (as opposed to
extension)? What
I would like to do is the following:
1. If SIP phone IP belongs to 192.168.0.0/24
http://192.168.0.0/24 http://192.168.0.0/24
subnet, set CALLERID=
2. Else, set CALLERID=
Thanks in advance for any examples or help.
Steve
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Steve, yes you can do it.
You'll need to use the SIPPEER function (available in trunk only I
believe) to get ${SIPPEER(ipaddr)}. You can then use Set and If()
to do
what you want.
SKM
I'm sorry, you need ${SIPPEER(${EXTEN}|ipaddr)} to retrieve the data you
want
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Your best bet would be to use the REGEX function to match the first
three octets :)
Rushowr
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RE: [asterisk-users] includes in realtime ??
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Jacob Sent: Monday, September 04, 2006 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] includes in realtime ?? Hello ppl, Is it possible to include contexts in the RealTime scenario?? If not, wots the work around?? Thanks in advance. Ben. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Amazing how the wiki has this vast amount of AT LEAST info to start your research on http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Zaptel-1.2.8 compile problem
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vidura Senadeera Sent: Monday, September 04, 2006 5:15 AM To: asterisk-users@lists.digium.com Cc: asterisk-dev@lists.digium.com Subject: [asterisk-users] Zaptel-1.2.8 compile problem Hi, I have problem in compiling zaptel-1.2.8. My Linux version is 2.6. asterisk version and libpri versions are 1.2.11 and 1.2.3. Please refer the attached txt files for Linux version information and output of zaptel compile. I will be highly appreciated that any one can help me on this regard. -- Thanks Regards, Vidura B. Senadeera. -- Thanks Regards, Vidura B. Senadeera. For the love of all things you hold holy, why is it that people cannot learn to NOT CROSS POST!?! I, for one, don't appreciate getting 4 copies of the above message. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help compiling asterisk-addons on Debian?
You need to install libmysqlclient15dev, it's saying it can't find the header files it requires. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Aloi Sent: Friday, August 25, 2006 8:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Help compiling asterisk-addons on Debian? Hello All - Running the following: Debian Stable Asterisk SVN-branch-1.2-r41069 Checked out the following from SVN: asterisk-addons/branches/1.2 When I attempt to compile asterisk-addons I get the following: /usr/src/asterisk-addons $ make /clip asterdev1:/usr/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c` app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directory cdr_addon_mysql.c:39:20: errmsg.h: No such file or directory res_config_mysql.c:53:19: mysql.h: No such file or directory res_config_mysql.c:54:27: mysql_version.h: No such file or directory res_config_mysql.c:55:20: errmsg.h: No such file or directory make -C format_mp3 all make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3' /clip Is this looking for MySQL? I do have MySQL installed and running, a bit confused here anyone have any thouhts? -- -- Christopher T Aloi -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Keys pressed not registering ...
Try changing the DTMF mode for that line, I've found that if rfc2833 doesn't work, inband will -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lenny Sent: Saturday, September 02, 2006 4:28 AM To: Asterisk-Users@lists.digium.com Subject: [asterisk-users] Keys pressed not registering ... Hello all, For some reason when dialing in I get the IVR or if I forward to my conference line... any keys pressed seem like they aren't received .. Like I'm pressing them, but they aren't being registered with the server .. Any ideas? I'm using the vmware nerdvittles build, the latest trixbox v1.1 .. FreePBX 2.1.1. Everything else works just fine. I'm using VoIPDiscount for outgoing and Stana-in/Stanaphone to receive calls. Any help is appreciated.. Regards, LB ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Connecting two asterisk servers
In short, yes... The wiki (http://www.voip-info.org) has documentation on how to configure your servers, how to configure the dialplan, etcI don't mean to single you out mate, but has anyone else noticed an increase in the number of questions being asked that could have been answered simply by visiting the wiki, reading the sample docs in the package, or even doing a Google search? I seem to recall the general rule of this list is that you should have already at least tried to find the answer. Here's a few links to get you started: The Asterisk Wiki, Asterisk Guru, Getting Started, GNU Inter, AGI Guide, O'reilly Onlamp Article - by John Todd, One Unified. It took me more time to cut and past those links than it did to find them, they were on the Asterisk.org support page. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy BoySent: Tuesday, August 29, 2006 11:16 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Connecting two asterisk servers Hi friends,Thank you to all for your response and cooperation to me. I have a doubt.I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India).1) Is it possbile to connect these two * servers?2) The person who is registered with Florida * server is able to make call to another person, who is registered with Delhi * server (like Intercom)?Looking forward to your response. Thank you.With ward regards,Chandra. Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension Ring on Multiple Phones
That's very very odd...that should work fine :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120) ; works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new line above, but when I dial the DID 879-6777 it rings on extension 120 only. Have I missed a step? Larry Jonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension Ring on Multiple Phones
Then entire OLD extension must be removed so the new one will match -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120); works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the CLI, see the new line with 'show dialplan' and actually see the new line above, but when I dial the DID 879-6777 it rings on extension 120 only. Have I missed a step? Larry Jonathan k. Creasy wrote: EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk with PABX
*dunks email in bucket* Heheh...Gee, ya think, Dean? Pardon my possession of an opinion. *Cautiously waits for next flame* SKM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Monday, August 28, 2006 9:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk with PABX Hey if you don't need the work just say no, I'm sure someone else will be happy to take the money from them. Maybe the monkey? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rushowr Sent: Monday, 28 August 2006 2:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk with PABX Too true too true Personally, I think trying to use Trixbox to learn Asterisk is akin to a monkey humpin' a footballIt's just not right. Anywhohad to do my smartass deed for the day Rushowr (Hates getting contracts to fix someone's AAH/TrixBox/FreePBX phone system) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Monday, August 28, 2006 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with PABX Dean Collins wrote: Yes it is possible. May I suggest you spend more time with www.voip-info.org Or even better download www.trixbox.org on an old server to get an idea of how configs work. Getting Trixbox would help him understand how Trixbox configs work, not how Asterisk configs work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Max number of SIP devices registered to anextension
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 To a single extension? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon Galbraith Sent: Sunday, August 27, 2006 8:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Max number of SIP devices registered to anextension Is there a maximum number of SIP devices that can be registered to an extension? -brandon -- Brandon Galbraith Email: [EMAIL PROTECTED] AIM: brandong00 Voice: 630.400.6992 A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: ENCRYPTED WITH GPG iD8DBQFE8jX2lfQsv7FBhp8RAsLaAKCn1dikeDpAZrRQ/1ESAcsFso09egCgrJBE AvXmh2VtjGAJPvixfnpwEWM= =cyAY -END PGP SIGNATURE- To a single extension? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon GalbraithSent: Sunday, August 27, 2006 8:16 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Max number of SIP devices registered to anextension Is there a maximum number of SIP devices that can be registered to an extension?-brandon-- Brandon GalbraithEmail: [EMAIL PROTECTED] AIM: brandong00Voice: 630.400.6992"A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost" PGPexch.htm.asc Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Max number of SIP devices registered to anextension
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Actually, isn't there SLA work being done in the trunk right now? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Sent: Monday, August 28, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Max number of SIP devices registered to anextension 1 On 8/27/06, Brandon Galbraith [EMAIL PROTECTED] wrote: Is there a maximum number of SIP devices that can be registered to an extension? -brandon -- Brandon Galbraith Email: [EMAIL PROTECTED] AIM: brandong00 Voice: 630.400.6992 A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: ENCRYPTED WITH GPG iD8DBQFE8uzVlfQsv7FBhp8RAhPAAJ9U0DSatH/FMxbCdosRnCuDB1zcagCZAYfO KWWtuXx54Rq8+Yky7HZCr2g= =Klim -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] RE: [asterisk-dev] Phone status
IIRC, you'll want to look at 'hint' extensions, and possibly subscriptions to get status updates From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MirSent: Monday, August 28, 2006 9:34 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] RE: [asterisk-dev] Phone status Your are right, I dont have to invent the wheel again, and I'm getting cleverer by looking at other peoples code. But this does not solve my problems, I have worked in the PABX business as a software developer for about 8 years, and coming to * is not all that easy. For instance, * does not give you very good information of the state of extensions (like we are used to in the "old-fashioned" PABX business), or maybe I'm not good at finding the information. I'm trying to port an existing Windows application to *, its a dialer, used to dial and se information about received calls. I know how to dial new calls, by using ORIGINATE on the AMI. I can receive some status information via the AMI, but consider this example: I receive a call, which I accept. I get an event from theAMI,that the call is now in the UP state. I receive another call, I get en event from the AMI, that the new call is in the RINGING state. So far, so good. I now answer the other call (for instance by the line button on my phone). Both calls are now in the UP state, who am I talking to? This, and many other questions, are currently making me even more thin haired than normal :-) Michael 2006/8/25, C F [EMAIL PROTECTED]: So how about inventing a car? The auto industry is much more profitable.The point; there is no point in reinventing the wheel, why are you writing this from scratch?On 8/24/06, Mir [EMAIL PROTECTED] wrote: What do you mean? I'm not looking for someone elses work, I'm developing an application from scratch. Michael 2006/8/24, Andrew Kirch [EMAIL PROTECTED]: Umm Flash operator panel? Andrew From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Mir Sent: Thursday, August 24, 2006 2:18 PM To: asterisk-users@lists.digium.com; asterisk-dev@lists.digium.comSubject: [asterisk-dev] Phone status Hi I'm working on a project, where I need the status of every telephone on the system. (Idle,ringing,busy) If a phone is busy, I also need to know the callerid of the other end. I have made a deamon, which query Asterisk every second for active calls, this works by issuing a "Status" to the manager-interface, and processing the return data and then put the result into a MySQL table. The clients will query the MySQL table every second for the state of their phone, if there are no records with their numbers in it, they are considered idle. This works fine for calls from one SIP-phone to the other, this is for instance what it look like when extension 310 is connected to extension 311: Event: Status Privilege: Call Channel: SIP/310-08697fb8 CallerID: 310 CallerIDName: unknown Account: State: Up Link: SIP/311-0868fd98 Uniqueid: 1156442804.74 Event: Status Privilege: Call Channel: SIP/311-0868fd98 CallerID: 311 CallerIDName: Snom Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 13 Link: SIP/310-08697fb8 Uniqueid: 1156442804.73 That is pretty easy to decode. However when an external call is made to a SIP-phone, the result is different, this is a call from another Asterisk via an IAX trunk: Event: Status Privilege: Call Channel: SIP/311-08695698 CallerID: 35254390 CallerIDName: unknown Account: State: Up Link: IAX2/MR-1 Uniqueid: 1156442974.76 Event: Status Privilege: Call Channel: IAX2/MR-1 CallerID: 35436121 CallerIDName: unknown Account: State: Up Context: macro-vm Extension: s Priority: 5 Seconds: 9 Link: SIP/311-08695698 Uniqueid: 1156442974.75 The actual callerid of the caller is 3536121, 35254390 is the called number. How do I get the information, that 35436121 is connected to 311? Am I doing it in a stupid way, I'm aware that the Manager can give me realtime events, but I'm under the impression, that it is not very stable in a high traffic environment? Any help or good ideas would be appriceated. Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
RE: [asterisk-users] Max number of SIP devices registered toanextension
Well, since you can technically only have one phone registered to an extension, you'll need to do a simultaneous ring setup in your dial: Dial(SIP/1SIP/2SIP/3.) I may be having a momentary brain freeze about the '' but I believe that's right... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brandon GalbraithSent: Monday, August 28, 2006 11:35 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Max number of SIP devices registered toanextension I'm attempting to have multiple phones (geographically seperated) register to a single extension, so when the extension is dialed, any phone can pick up the call. Is this better handled by having each phone have a seperate extension, and handle the call routing in a dial plan? -brandon On 8/28/06, Rushowr [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE-Hash: SHA1Actually, isn't there SLA work being done in the trunk right now? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Matt Sent: Monday, August 28, 2006 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Max number of SIP devices registered to anextension 1 On 8/27/06, Brandon Galbraith [EMAIL PROTECTED] wrote: Is there a maximum number of SIP devices that can be registered to an extension? -brandon-- Brandon Galbraith Email: [EMAIL PROTECTED] AIM: brandong00 Voice: 630.400.6992 "A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -BEGIN PGP SIGNATURE-Version: GnuPG v1.4.5 (MingW32)Comment: ENCRYPTED WITH GPGiD8DBQFE8uzVlfQsv7FBhp8RAhPAAJ9U0DSatH/FMxbCdosRnCuDB1zcagCZAYfOKWWtuXx54Rq8+Yky7HZCr2g==Klim-END PGP SIGNATURE-___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Brandon GalbraithEmail: [EMAIL PROTECTED]AIM: brandong00Voice: 630.400.6992"A true pirate starts drinking before the sun hits the yard-arm. Ya. --thelost" ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] manual mods with GUI in place
You'll want to put them in the _additional.conf files, because AAH/TB/FPBX doesn't always play nice with changes to the configuration files that it modifies directly. Rushowr / SKM From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Curt ShafferSent: Monday, August 28, 2006 2:54 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: [asterisk-users] manual mods with GUI in place This post spurred off of the comment of Michael Collins on the Asterisk with PABX thread. I am going to post the relevant information here: I started w/ AAH, then went back and learned the dialplan apps, scripting, etc. For some guys like me, it's easier to start with a working (if limited) system, and then tinker with it, break it, etc. After breaking a few systems I then went back and did a vanilla install to learn some more. I ended up settling on a compromise: I load Trixbox and then make a bunch of manual mods. I get the best of both worlds - a system that has all of the prereqs loaded for me, plus a GUI for stuff that I don't want to do a cmd line and also the power and flexibility of hand-editing my .conf files to get exactly what I want out of the dialplan. For those wondering how to get started, I can highly recommend STARTING with Trixbox, but definitely don't STOP with Trixbox. After you play with a pre-installed, working system, go out and get your hands dirty on a plain install. You'll be better off for it in the long run. Having both GUI and cmd line experience will make you a well-rounded Asterisk user. -MC I started w/ AAH, then went back and learned the dialplan apps, scripting, etc. For some guys like me, it's easier to start with a working (if limited) system, and then tinker with it, break it, etc. After breaking a few systems I then went back and did a vanilla install to learn some more. I ended up settling on a compromise: I load Trixbox and then make a bunch of manual mods. I get the best of both worlds - a system that has all of the prereqs loaded for me, plus a GUI for stuff that I don't want to do a cmd line and also the power and flexibility of hand-editing my .conf files to get exactly what I want out of the dialplan. For those wondering how to get started, I can highly recommend STARTING with Trixbox, but definitely don't STOP with Trixbox. After you play with a pre-installed, working system, go out and get your hands dirty on a plain install. You'll be better off for it in the long run. Having both GUI and cmd line experience will make you a well-rounded Asterisk user. -MC My question to everyone is this..This is where I am at now. I have been using FreePBX for about a year, after moving from [EMAIL PROTECTED] I am starting to need some manual changes and modules. My question is can anyone point me in a direction on how to learn how to create these. I read the ORiley book and thumbed though some of the others, although I plan on reading them all the way through as time permits. I guess my question is where do I add these things. I would still like to use FreePBX because it just saves a ton of coding but I want to add my own things too. Do I put them in the *_additional configs (which appear to be written over by freePBX), the .conf files or the features.conf? Any web links with beginner how tos or more info on this would be appreciated as well! I didnt want to cross post ;) Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk with PABX
Too true too true Personally, I think trying to use Trixbox to learn Asterisk is akin to a monkey humpin' a footballIt's just not right. Anywhohad to do my smartass deed for the day Rushowr (Hates getting contracts to fix someone's AAH/TrixBox/FreePBX phone system) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Monday, August 28, 2006 1:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk with PABX Dean Collins wrote: Yes it is possible. May I suggest you spend more time with www.voip-info.org Or even better download www.trixbox.org on an old server to get an idea of how configs work. Getting Trixbox would help him understand how Trixbox configs work, not how Asterisk configs work. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Max Time
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
First big question is are you checking beforehand how long the limit should
be by calculating ((BALANCE / RATE) / 1000)
If you're not, that would be why it doesn't disconnect the customer within a
time period that wouldn't result in a negative balance.
Other than that, you might want to possibly check if your script is getting
the dialstring properly. Do you need to escape the / characters in it? What
I'd personally do is set up some Verbose() statements in my scripts to
output debugging data.
Hope this helps!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Abdul
Sent: Sunday, August 27, 2006 2:54 AM
To: Asterisk-Users@lists.digium.com
Subject: RE: [asterisk-users] Call Max Time
Hello,
Could you tell how i can use it in PERL AGI script?
currently i am using in my AGI with this format, but some time call
is not disconnecting customers talking without money.
$dialstr = SIP/terminator/15745405022|350|tTL(653044:7000:5000);
$AGI-exec('Dial', $dialstr);
regards,
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First big question is are you checking beforehand how long
the limit should be by calculating ((BALANCE / RATE)/
1000)
If you're not, that would be why it doesn't disconnect the
customer within a time period that wouldn't result in a negative balance.
Other than that, you might want to possibly check if your
script is getting the dialstring properly. Do you need to escape the /
characters in it? What I'd personally do is set up some Verbose() statements in
my scripts to output debugging data.
Hope this helps!
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
AbdulSent: Sunday, August 27, 2006 2:54 AMTo:
Asterisk-Users@lists.digium.comSubject: RE: [asterisk-users] Call
Max Time
Hello,Could you tell how i can use it in PERL AGI
script?currently i am using in my AGI with this format, but some time
call is not disconnecting customers talking without money.$dialstr =
"SIP/terminator/15745405022|350|tTL(653044:7000:5000)";$AGI-exec('Dial',
$dialstr);regards,
Get your own web
address for just $1.99/1st yr. We'll help. Yahoo!
Small Business.
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RE: [asterisk-users] Call Max Time
from within asterisk, just run the following command: show application Verbose That'll fill you in. Your other solid option is to search the wiki From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AbdulSent: Sunday, August 27, 2006 4:05 AMTo: Asterisk-Users@lists.digium.comSubject: RE: [asterisk-users] Call Max Time Hi,i am using the same calculating ((BALANCE / RATE) / 1000) method to return tTL.and i am sure my GAI is working well. but could u tell me how i can set Verbose() sepecial for my dialstring?Regards, Get your own web address for just $1.99/1st yr. We'll help. Yahoo! Small Business. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Shared NFS or Shared MySQL for redundant secondaryserver?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Personally I've used the shared database method previously, I've even setup a mysql cluster and had each asterisk host be a query node. SKM From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Aloi Sent: Sunday, August 27, 2006 5:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Shared NFS or Shared MySQL for redundant secondaryserver? Hey List! What are your thoughts on redundancy?? Is it best to have the two Asterisk boxes share a /etc/asterisk directroy; so if one falls out of service the other takes over or is it best to have each node pull from a shared DB?? Cheers! -- -- Christopher T Aloi -- -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: ENCRYPTED WITH GPG iD8DBQFE8hBAlfQsv7FBhp8RArRfAKCVVuCcF+aSpLijO2rWZPa+Len05ACg1JaL z5bCCH/cWkJIAqKxsQMtC1U= =nhUW -END PGP SIGNATURE- Personally I've used the shared database method previously, I've even setup a mysql cluster and had each asterisk host be a query node. SKM From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher AloiSent: Sunday, August 27, 2006 5:31 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Shared NFS or Shared MySQL for redundant secondaryserver? Hey List!What are your thoughts on redundancy?? Is it best to have the two Asterisk boxes share a /etc/asterisk directroy; so if one falls out of service the other takes over or is it best to have each node pull from a shared DB?? Cheers!-- --Christopher T Aloi-- PGPexch.htm.asc Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call Max Time
Set(TIMEOUT(absolute)=seconds) Change seconds to the number of seconds you want to allow a call to last From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of AbdulSent: Sunday, August 27, 2006 1:21 AMTo: Asterisk-Users@lists.digium.comSubject: [asterisk-users] Call Max Time Hi All,Could anyone give me idea, How i can set Call Max Time, so in pariticular time the call should disconnect automatically.I will be appriciate for your helps.Abdul Get your email and more, right on the new Yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: column width in CLI
I think he actually needs show channels verbose *CLI help show channels Usage: show channels [concise|verbose] Lists currently defined channels and some information about them. If 'concise' is specified, the format is abridged and in a more easily machine parsable format. If 'verbose' is specified, the output includes more and longer fields. Cheers SKM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Sent: Wednesday, August 23, 2006 6:59 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: column width in CLI Try show channels concise -- -- Steven http://www.glimasoutheast.org Shaun Hofer [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi, Can the column width for commands run in the Asterisk CLI be increased? Currently when I run 'show channels' I can't see the whole channels id/name as its to long for the columns width and is cut off. I need to grab a list of active channels, which is currently not do able. Thanks Shaun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Adding/Removing Prefixes
I now need to remove the 9 but then prefix another number onto
the phone number before dialing now but am unsure how to do
this is the dialplan.
Simple...for instance, if you wish to prefix 123 before the number just do:
Dial(SIP/123${EXTEN}
Would someone be able to point me in the right direction or
provide an example diaplan that does this?
Many Thanks in Advance
SP
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RE: [asterisk-users] How to set externip in sip.conf automatically?
I believe you want to use ${ENV(variable)}.. From asterisk's CLI:
*CLIshow function ENV
-= Info about function 'ENV' =-
[Syntax]
ENV(envname)
[Synopsis]
Gets or sets the environment variable specified
Note that ENV is a function...you need to encase the argument inside
parentheses
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Larry Alkoff
Sent: Wednesday, August 23, 2006 9:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to set externip in sip.conf
automatically?
As stated in the original post, when I entter the IP with an
editor directly into sip.conf calls work just fine but I am
looking for a way to have that done _automatically_.
The Asterisk - Future of Telephony book says it is possible
for Asterisk to access a Linux environment variable containing
the IP information in the form of ${ENV{variable}}.
It doesn't seem to work. I am asking how to make it work.
Larry
Watkins, Bradley wrote:
If you already have the IP in a file, why don't you set it up so the
file itself says: externip=xx.xx.xx.xx and then do a #include in
sip.conf for the /etc/myip file? I believe you'll have to do a sip
reload either way (which can obviously be part of your cron job) if
you're not already, but that should do what you're looking to do.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Larry
Alkoff
Sent: Tuesday, August 22, 2006 9:34 PM
To: Asterisk-users; Austin-asterisk-users
Subject: [asterisk-users] How to set externip in sip.conf
automatically?
I need to give Asterisk access to my external IP address
to prevent
the NAT problem where caller cannot hear the callee's voice.
According to Asterisk - The Future of Telephony page 92 Environment
Variables:
Environment variables are a way of accessing Unix environment
variables from within Asterisk. They are referenced in the form of
${ENV{var}}
where var is the Unix environment variable you wish to reference.
My external IP is placed each night in a file call /etc/myip and
placed in the $MYIP variable by /etc/bashrc when an shell is loaded.
So I have /etc/myip refreshed each night in a cron job and when a
shell is opened /etc/bashrc does:
export MYIP=`cat /etc/myip`
To access the variable in sip.conf I have tried:
externip=${ENV(EXTERNIP)}
and
${ENV($EXTERNIP)}
but neither seems to work.
Is this the correct syntax? Did I misinterpret the book?
I say neither seems to work because When I hard code
externip=69.91.84.176
there are no NAT problems but when I try to access the $MYIP
variable
either of the ways above NAT prevents me hearing the callee's voice.
I have tried but not found a way to directly access the contents of
MYIP to the console using the CLI. Is there a way to see or
set _any_
Linux enviromnent variable using the CLI? More generally, how do I
access the Linux shell from the CLI?
The problem with simply using
externip=69.91.94.176
is that number is subject to change and I don't know an easy way to
automatically write the value into sip.conf programatically.
I could have just said how do I do this but wanted to show
that I've
done my homework.
Thanks for any help.
Larry
--
Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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Larry Alkoff N2LA - Austin TX
Using Thunderbird on Linux
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RE: [asterisk-users] MySQL CDR
Download the asterisk-addons package. It contains several addons, including all the mysql additions. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Quintana Cruz Sent: Thursday, August 24, 2006 4:06 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] MySQL CDR Hi everyone, I finished installing the Xorcom Rapid's Asterisk Packages with amportal (1.10.10), but i wasn't able to find the asterisk-mysql package. Any idea what happened there?, Is there another reposiitory for that package for asterisk 1.0.11. Or could somebody send me the cdr_addon_mysql.so file? Thanks for your responses, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Trunk with multiple IPs?
I wish I could offer some direct help on whether or not your method with a comma separated list would work, but I can't. However, you could always create a few entries using different formats and then run some tests against them -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin Lawetz Sent: Wednesday, August 23, 2006 9:42 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Trunk with multiple IPs? Still no answers huh? I've asked a couple of time how to do this, and by the lack of answers, I'm guessing there is no way. The workaround unfortunately is to create an entry for each IP address in the range (I hope you don't have to open up a whole C class) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Warren (mailing lists) Sent: August 22, 2006 3:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Trunk with multiple IPs? How do I enter a trunk with multiple IPs. xyz voip provider has 4 IPs and I want to allow incoming from any of them: 1.1.1.1, 1.1.1.2, 1.1.1.3 and 1.1.1.4 Do I put 4 separate host= lines, do I put a single host=line that is comma separated or do I have to set up 4 separate incoming trunks? TIA, Warren ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Setting the contact header on outbound INVITE
Not last I heard...I just fought with this yesterday From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael LunsfordSent: Tuesday, August 22, 2006 8:10 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Setting the contact header on outbound INVITE Is there anyway to set the Contact header on outbound INVITEs such as there is for the REGISTER? I would also like to be able to set the Contact header on responses. Thanks, Michael This email may contain confidential information. If you are not the intended recipient, please advise by return email and delete immediately without reading or forwarding to others. -- Cbeyond ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange SIP response
Diego, I've encountered this before, let me review a couple of old logs and notes and I'll get back to regarding this. Cheers, SKM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Andrés Asenjo González Sent: Tuesday, August 22, 2006 7:26 PM Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Strange SIP response Rushowr wrote: Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one of my personal favorites Yes, I have used it. The lines are extracted from a sip debug on the CLI. I'm going to paste more lines: Sip read: SIP/2.0 480 Temporarily Unavailable To: sip:[EMAIL PROTECTED]:6198;tag=e4331437 From: 24307022sip:[EMAIL PROTECTED];tag=as288765a2 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1;received=172.16.1.3 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 7 headers, 0 lines -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.50 Transmitting: ACK sip:[EMAIL PROTECTED]:6198 SIP/2.0 Via: SIP/2.0/UDP 172.16.1.3:5060;branch=z9hG4bK73129cd1 From: 24307022 sip:[EMAIL PROTECTED];tag=as288765a2 To: sip:[EMAIL PROTECTED]:6198;tag=e4331437 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 192.168.1.50:6198 -- SIP/EXT25-a454 is circuit-busy == Everyone is busy/congested at this time I have not detected packet losses even. Thanks for your response. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Andres Asenjo G. Sent: Tuesday, August 22, 2006 6:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange SIP response Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help compiling asterisk-addons on Debian?
Do you have the development libraries installed too? I believe on Debian it's something like libmysqlclient From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher AloiSent: Friday, August 25, 2006 8:36 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [asterisk-users] Help compiling asterisk-addons on Debian? Hello All -Running the following:Debian StableAsterisk SVN-branch-1.2-r41069Checked out the following from SVN:asterisk-addons/branches/1.2 When I attempt to compile asterisk-addons I get the following: /usr/src/asterisk-addons$ make/clipasterdev1:/usr/src/asterisk-addons# make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE `ls *.c`app_addon_sql_mysql.c:23:19: mysql.h: No such file or directory cdr_addon_mysql.c:38:19: mysql.h: No such file or directorycdr_addon_mysql.c:39:20: errmsg.h: No such file or directoryres_config_mysql.c:53:19: mysql.h: No such file or directoryres_config_mysql.c:54:27: mysql_version.h: No such file or directory res_config_mysql.c:55:20: errmsg.h: No such file or directorymake -C format_mp3 allmake[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'/clipIs this looking for MySQL? I do have MySQL installed and running, a bit confused here anyone have any thouhts? -- --Christopher T Aloi-- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SSH connection hangs on logout?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Thursday, August 24, 2006 2:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SSH connection hangs on logout? On Wed, Aug 23, 2006 at 11:03:23PM -0700, Steve Edwards wrote: On Thu, 24 Aug 2006, Jeremy McNamara wrote: Rushowr wrote: Hey all, I have an interesting issue that just recently started when I grabbed a copy of the trunk about a week ago and compiled it. Ever since that compile, if I start Asterisk (disconnected terminal, using safe_asterisk to launch) and then continue on about my work with it, when I disconnect my SSH terminal (using latest version of PuTTY) the session no longer closes it just hangs. I've even changed the Putty setting to close the window even on unclean exit but it still hangs the connection... I had something similar once with Zabbix a while back, but never Asterisk. Anyone else experience this? Start asterisk using safe_asterisk or via asterisk -f I prefer the safe_asterisk shell script, since if asterisk seg faults, there is a good chance asterisk will get automatically restarted. Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You may need to redirect stdin, stdout, stderr like: run_asterisk\ 0/dev/null\ 1/dev/null\ 2/dev/null\ In other words: A plain 'asterisk' (without '-c' and such) that daemonizes and does exactly that for you, among others. Asterisk is a daemon, rather than an interactive program. Thus its handling for SIGHUP is to re-read configuration rather than detach from the terminal. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Shoots out own flames before emailing Gents, asstated in my email, I am using safe_asterisk. Additionally, even when I started asterisk by hand, it was always forked off of my tty. However, even if I DID have it connected to my tty, I'd have to issue stop now before getting to the command prompt and being able to issue logout to bash. Try this sometime gents, you'll see what I mean...issue a ! From the *CLI...then type logout...You'll be told that you're not in a login shell and to use exit. Wow.. -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (MingW32) Comment: ENCRYPTED WITH GPG iD8DBQFE7cEuwWoA8HY7JXYRAqtoAJwNX8/L7OFuXvTPobOvJ8cH0Iei9QCaApf0 S4BH1uc4ZxWxei0gRy+qKy0= =6PsA -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Quiet on the list today?
Just gotta check, I've never seen a complete day with no posts ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SSH connection hangs on logout?
Hey all, I have an interesting issue that just recently started when I grabbed a copy of the trunk about a week ago and compiled it. Ever since that compile, if I start Asterisk (disconnected terminal, using safe_asterisk to launch) and then continue on about my work with it, when I disconnect my SSH terminal (using latest version of PuTTY) the session no longer closes it just hangs. I've even changed the Putty setting to close the window even on unclean exit but it still hangs the connection... I had something similar once with Zabbix a while back, but never Asterisk. Anyone else experience this? SKM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Strange SIP response
Have you run SIP DEBUG PEER 192.168.1.60? It may help...tcpdump is also one of my personal favorites -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Diego Andres Asenjo G. Sent: Tuesday, August 22, 2006 6:50 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Strange SIP response Hi, I am getting the following message on the CLI: -- Got SIP response 480 Temporarily Unavailable back from 192.168.1.60 -- SIP/EXT23-d910 is circuit-busy and the call hangs up. The peer is correctly registered and I'm not getting unavailable messages. I really need help with this error. -- MENSAJE ENVIADO CON WMAIL 1.01 UNIVERSIDAD DEL CAUCA ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Variable to show caller id for a current call?
Wait a minutewhy are you putting 227 into the CALLERID function? You
should read this:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+callerid
The (number) portion is the argument to CALLERID telling it what to give
you, not what to insert/write
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable to show caller id for a
current call?
But how do you get that with GetVar? I am trying to do this
through the API. I tried:
Action: GetVar
Variable CALLERID(227)
and I tries:
Action: GetVar
Variable ${CALLERID(227)}
Neither returned anything.
How can I do this? Alternately... Is there a way to have a
program fired off when an extension rings that will have the
caller id passed to it as part of the call?
W
Rushowr wrote:
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable to show caller id for a
current call?
Is there a variable that can be gotten with GetVar to show the
callerid of the current incoming call in progress at a sip
extension?
For instance, a caller from 516-922-9463 calls extension 234.
I would like to be able to be able to get back the
516-922-9463 if I pass 234.
Also, can this be done while the extension is ringing?
TIA,
Warren
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RE: [asterisk-users] re-writing the dial plan - some hints please
You can't use that realtime field in an include statement... However, you
could use context names like caller-conference and caller-longdistance and
then call the context dynamically with Goto(caller-${key}).
Otherwise, you're going to have to do it with logic routing. May I suggest
at LEAST using Subroutines (end the subroutines with the Return command, and
call them with Gosub), or maybe even doing a little logic magic with GotoIf
statements? Of course, all of that logic routing can be a pain, but to
further alleviate my personal dialplan code pains, I use AEL2 which makes
the code a little more like Perl/C/PHP code, and allows for If/Else
statements, etc...
Just my 0.02.
SKM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald Wiplinger
Sent: Tuesday, August 22, 2006 8:03 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] re-writing the dial plan - some hints please
My dialplan grew over the last months and I want to
restructure it. What hints do you have for me?
There are some points I want to do, but none of my tests worked.
I use realtime, and have there a field called key, which can
have several flags.
E.g. a flag if the user is allowed to use a conference room,
can call long distance, can call overseas, can call local
pstn, different tariffs,
I tried something like:
[test-key]
exten = _.,1,NoOp(variable key is ${key}) exten =
_.,2,Set(flag_int =${CUT(key,,1)}) exten =
_.,3,Set(tarif=${CUT(key,,2)}) exten = _.,4,NoOP(flag_int is
${flag_int} and tarif is ${tarif})
and wanted to use this variables in the next context, by using
include statments, but it did not work.
[caller]
include = test-key
include = A
include = B
...
The idea was to set at each entrance point first all flags and
variables. Than I can use a common dialplan.
If a flag is set, than I could include another context.
Unfortunately there is no IF()include. I might be able to set
a jump in each context to the end if the flag is not set.
Any idea how I can do that?
Any ideas of structuring the dialplan more efficiently?
bye
Ronald
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RE: [asterisk-users] if command for or missing callerid?
Gotoif($[${ISNULL(${CALLERID(number)})} = 1]?ask4cardnum:doagi_astcc)
:-)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Ronald Wiplinger
Sent: Tuesday, August 22, 2006 7:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] if command for or missing callerid?
I am looking for a way to make a decission in the dialplan if
I have a caller id or not.
What I want to do with it:
Call on the PSTN line should either use astcc.agi with the
caller-id in place as card number, or asking for the calling
card number.
How can I make this gotoif ???
bye
Ronald
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[asterisk-users] Setting RPID privacy?
Hello all, Just had a question that I've not been able to find a suitable answer for. When we receive calls on SIP, we can get SIP_HEADER(Remote-Party-ID) and check the privacy flag for what privacy is requested. Now, since SIP_HEADER is not writable, how can I set the privacy flag in the RPID header? Should I just use CallingPres? Just set the CID to RESTRICTED ? Any hints, suggestions? SKM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Setting RPID privacy?
Nevermind Gents and Ladies, I looked AGAIN at the dialplan command list and found the one word I missed before when scanning SetCallerPres...Independent as in Channel Independent. Please excuse my error. SKM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rushowr Sent: Tuesday, August 22, 2006 8:55 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Setting RPID privacy? Hello all, Just had a question that I've not been able to find a suitable answer for. When we receive calls on SIP, we can get SIP_HEADER(Remote-Party-ID) and check the privacy flag for what privacy is requested. Now, since SIP_HEADER is not writable, how can I set the privacy flag in the RPID header? Should I just use CallingPres? Just set the CID to RESTRICTED ? Any hints, suggestions? SKM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Variable to show caller id for a current call?
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable to show caller id for a
current call?
Is there a variable that can be gotten with GetVar to show the
callerid of the current incoming call in progress at a sip extension?
For instance, a caller from 516-922-9463 calls extension 234.
I would like to be able to be able to get back the
516-922-9463 if I pass 234.
Also, can this be done while the extension is ringing?
TIA,
Warren
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RE: [asterisk-users] Variable to show caller id for a current call?
Well, for one, you could set something like CID = ${CALLERID(number)} in the
dialplan, and then GetVar CID
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Variable to show caller id for a
current call?
But how do you get that with GetVar? I am trying to do this
through the API. I tried:
Action: GetVar
Variable CALLERID(227)
and I tries:
Action: GetVar
Variable ${CALLERID(227)}
Neither returned anything.
How can I do this? Alternately... Is there a way to have a
program fired off when an extension rings that will have the
caller id passed to it as part of the call?
W
Rushowr wrote:
${CALLERID(number)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Warren (mailing lists)
Sent: Monday, August 21, 2006 1:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Variable to show caller id for a
current call?
Is there a variable that can be gotten with GetVar to show the
callerid of the current incoming call in progress at a sip
extension?
For instance, a caller from 516-922-9463 calls extension 234.
I would like to be able to be able to get back the
516-922-9463 if I pass 234.
Also, can this be done while the extension is ringing?
TIA,
Warren
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[asterisk-users] Quick, hopefully easy, question
Hey all, I've done some peeking around and can't find a GOOD listing of what the currently supported SIP headers are that Asterisk supports. My main reason is to get the CallerID/RPID settings for whether or not to display, but there's others as well. Anyone have a link? SKM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
*steps slowly to the soapbox* Can we please get this pissing match over with? The horse is dead, stop beating it and bury the corpse for chrissake *steps down from soapbox* That's all I got *checks the fire extinguisher and awaits the flames to be redirected* SKM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara Sent: Wednesday, August 16, 2006 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. Ok - And the problem is? Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
Oh my gawdwhy are my emails taking so long to publish? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rushowr Sent: Thursday, August 17, 2006 9:43 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 'Hosting' *steps slowly to the soapbox* Can we please get this pissing match over with? The horse is dead, stop beating it and bury the corpse for chrissake *steps down from soapbox* That's all I got *checks the fire extinguisher and awaits the flames to be redirected* SKM -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara Sent: Wednesday, August 16, 2006 3:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 'Hosting' Douglas Garstang wrote: Well, we're talking about several dozen, maybe 100, companies, per Asterisk box here. Ok - And the problem is? Jeremy McNamara ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: what is the real use of AEL?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barzilai Sent: Friday, August 18, 2006 2:55 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Re: what is the real use of AEL? Steve Murphy wrote: ... [a lot of well-written arguments] ... And, pardon the shameless plug here, but for all you fence sitters, I invite you to try AEL in/for your dialplans, and give me feedback! If the majority of those who use it feel it's useless, I'll drop it and do other useful things for Asterisk-- there's plenty to do! murf Please, don't! Even if it last only a few versions, it will be worth it! BarZ Murf, I think you know where I stand on this ;-) Rushowr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Dialplan or matching
IIRC, You can use REGEXes in your extension matchingDon't have a handy link, but if I find it, I'll forward -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William Moore Sent: Friday, August 18, 2006 1:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dialplan or matching On 8/18/06, David Cook [EMAIL PROTECTED] wrote: Maybe I'm daft, but can asterisk to 'or' logic in dialplan matches sort of like the SPA's can? Tollfree numbers for example. I can have a line for each combination: exten = _1800NXX, Dial, exten = _1866NXX, Dial, exten = _1877NXX, Dial, exten = _1888NXX, Dial, But I want to do is something like this: exten = _18[0678][0678]NXX, Dial, . This syntax is valid and would work for what you're doing, but as you said, there is a chance of logic error in it. Or to prevent the logic error which albeit small, the above would create: exten = _18[00,66,77,88:2]NXX, Dial, .. (representing that the next 2 chars must equal any of '00'.'66','77' or '88' As for this syntax, Asterisk does not respect the [], so it parses the 66 as the priority. I have no idea how to properly do this in one line if it is possible. Kinsey ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] astbill white screen!!
Sounds like a sessions error -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Milioto Sent: Thursday, August 17, 2006 2:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] astbill white screen!! Hi all, I've installed asterisk and astbill according with all recommendation (mysql5, drupal included with astbill, php, apache2...). When I write http://server_adress/astbill, I get a white screen page. Browser doesn´t give me an error page, it just a white screen page. However asterisk doesnt have any problem, and works well with mysql. I also have installed Drupal 4.7.3 linked to other database with other user and password working well. And I have phpMyAdmin too. All working very good at the same server. I tried changing index.php to phpinfo.php in the same directory and it works well too. Can anybody help me with that please? Any suggestion will be very appreciated. Thanks, very much in advance Sebastian On 7/14/06, varun [EMAIL PROTECTED] wrote: Hello, Our asterisk server is on Centos 4.2 We want to use Astbill. Astbill requires Drupal and mysql 5. I could not find rpms mysql5 for centos. We are getting mysql extensions issues because of php-mysql. How do we solve this ? Any other billing software that similar to Astbill ? Thanks Varun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 'Hosting'
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, August 17, 2006 4:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk 'Hosting' -Original Message- From: Douglas Garstang Sent: Thursday, August 17, 2006 2:17 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk 'Hosting' **snip** I spent 8+ hours a day, 5+ days per week for over 6 months thinking how these functions fit within the realm of Asterisk. At every single turn, after going down every single path, there where limitations that forced us to backtrack and evaluate a different approach. A script that could handle call routing, in conjection with MySQL and stored procedures was the only way to implement our requirements. The MySQL command had limitations, realtime was way too resource intensive, unreliable and undocumented and so on. Yep... i definitely haven't thought about this at all. Oops. I almost forgot intra-organisational 4 digit extension dialling. Not just company, but organisational, where a company may have multiple organisational units. It might be possible to hack together a flat intra-business 4 digit extension dial lookup in the native dialplan, but trying to make it a multi-level organisation lookup would be pure hell... unless you farm the task out to a more advanced scripting langauge like python, perl whatever. I see the MySQL dial plan command still doesn't support stored procedures either, unless you hack around with the source. I've just recently come up against this limitation. Care to share info/code concerning making stored procs work with the addon? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
MySQL Addon and MySQL5 Stored Procs (WAS: RE: [asterisk-users] Asterisk 'Hosting')
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Thursday, August 17, 2006 4:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Asterisk 'Hosting' Hi. I only just stumled across it myself. I was trying to prove a point to Jeremy. On the voip wiki: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+MYSQL under a comment titled 'Calling MySQL 5 stored procedures from app_mysql', it looks like someone has managed to modify the source to get it to work. I haven't tried it yet... Doug. Thanks for the point, I hadn't noticed that comment. I'll be implementing it on one of my dev systems tonight and testing a multitude of stored procedures that we had planned. If it works reliably, I'll be reporting it as a bug in the digium bug tracker and submitting the modification as a patch (with proper credit to the original poster of the mod of course). S McGowan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sending Email From A Dial Plan
Instead of SYSTEM(), you could use an AGI possibly.
Cheers,
SKM
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damien
Gabrielson
Sent: Thursday, August 17, 2006 6:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Sending Email From A Dial Plan
Hi,
I'm looking for a simple way to send email from a dial plan. I have searched
around quite a bit looking for a solution for this and I'm surprised that I
haven't found anything useful yet other than using the
System() application. I would like to be able to change the subject
dynamically based on ${EXTEN} and the body is not important. I was hoping to
have a one line command from the System() application without having to
write a script or any other dependency. Has anyone implemented anything like
this?
Thanks,
Damien
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RE: [asterisk-users] Asterisk Real Time and sip.conf file used at
Realtime configuration is when you tell Asterisk to use the database for reading the sip global configuration items. Static configuration is when you use the sip.conf file to store the sip global configuration items. You cannot mix the two. That's all. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yusuf Sent: Wednesday, August 16, 2006 3:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Real Time and sip.conf file used at On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote: Is it possible to use Asterisk RealTime and also config files (like sip.conf) at the same time? As much as I know, only one thing can be used and I need them both working!... Yes, you can use both at the same time. The only restriction is that you cannot use the realtime static configuration and realtime configuration. -- Hi realtime static configuration and realtime configuration??? What is the difference, can you please explain? thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OPENSER / SER and Asterisk
I use Asterisk Realtime a LOT, it's pretty much the core of all my consulting jobs in the last year. If you still need help, I'll try to assist you as much as possible. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, August 16, 2006 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OPENSER / SER and Asterisk *lol* The cryptic replies have been exactly my problem as well! -Original Message- From: kjcsb [mailto:[EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 12:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OPENSER / SER and Asterisk Absolutely. The SER/OpenSER documentation is terrible, and if you post to the OpenSER mailing list, you get very cryptic replies. ___ Whilst I would agree with you regarding SER, the documentation of OpenSER is far better. Documentation of Asterisk Realtime on the other hand. Now *that's* terrible. Cameron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CallerID is not displaying for my incoming calls
What's the Dial command being used to pass the call to the Softphones? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy BoySent: Wednesday, August 16, 2006 3:23 AMTo: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] CallerID is not displaying for my incoming calls Hi,As you said, I have changed my configurations. But, callerid is not displaying. What I have to do? Please tell me.ThanksRegards,Chandra.Rich Adamson [EMAIL PROTECTED] wrote: Crazy Boy wrote: Hi Friends, We have installed Asterisk with Digium 04B card (4 FXO ports). Now, I have connected my PSTN line directly to first port. I am making outgoing calls and receiving incoming calls successfully through my Asterisk. The problem is: When I am receiving a call from outside (PSTN), I am not getting the callerid number and getting callerid as "Asterisk" in my softphones (XLite). Here I am giving my configuration files. zaptel.conf file contents: loadzone = us defaultzone=us fxsks=1-4 zapata.conf file contents: [channels] context=incoming signalling=fxs_ks busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes cancallforward=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callerid=asreceived language=en usecallerid=yes hidecallerid=no echocancel=yes transfer=yes immediate=no group=1 callgroup=9 pickupgroup=9 channel = 1The above entries appear to be reasonable and correct. If you have not properly set rxgain and txgain, it "could" impact callerid. If those gains are too high/low, asterisk will not properly read the callerid data when sent to you. extensions.conf file contents: [incoming] exten = s,1,Answer exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,Background(/tmp/virg2) exten = s,6,Goto(s,1) include = leader Got event 18 (Ring Begin)... Aug 14 14:11:58 WARNING[26744]: pbx.c:5869 pbx_builtin_dtimeout: DigitTimeout is deprecated, please use Set(TIMEOUT(digit)=timeout) instead. Aug 14 14:11:58 WARNING[26744]: pbx.c:5845 pbx_builtin_rtimeout: ResponseTimeout is deprecated, please use Set(TIMEOUT(response)=timeout) instead.The above two WARNING statements are telling you that either you are copying those exten= statements from someone's old config files, or, you haven't read the asterisk documentation. The message is telling you that your statement "exten = s,3,DigitTimeout,5" should be replaced with the Set(TIMEOUT(digit)=timeout) syntax. Your statements are still executing properly today, but the next time you upgrade asterisk code, they are likely to fail due to the old syntax not being supported.Try 'show function TIMEOUT' from your CLI and read it. What I have to do to display the PSTN caller number on my softphones? Please tell me. Looking forward to your response. Thank you.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CallerID is not displaying for my incoming calls
Chandra,
Unfortunately, I can't help you too much, because I've not
worked a lot with Zap. However, this message:
Aug 17 19:45:41 ERROR[10449]: callerid.c:276 callerid_feed:
fsk_serie made mylen 0 (-8)
Seems interesting...My guess is that the callerid
information is corrupted or something, because it's a negative value, not a 0 or
positive. Possibly you have your CID Signalling set to the wrong value... One
thing you could try just to get a better idea of what (if anything) is actually
read from the callerid and what the presentation is set to, is to modify the
your dialplan to output the data to your console (I use verbose 2 so I don't
have to read the extra info:
[incoming]exten = s,1,Wait(4)exten =
s,n,Answer
exten = s,n,Verbose(2|CallerID info received:
${CALLERID(all)}) ; shows CID info
exten =
s,n,Verbose(2|Presentation Setting: ${CALLINGPRES}) ; shows CID
presentationexten = s,n,SetMusicOnHold(default)exten =
s,n,Set(TIMEOUT(digit)=5)exten = s,n,Set(TIMEOUT(response)=10)exten
= s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten =
s,n,Hangup()include = leader
Hope this is helpful in
some way...
Rushowr
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy
BoySent: Friday, August 18, 2006 1:14 AMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[asterisk-users] CallerID is not displaying for my incoming
calls
Hi Rushowr,Thank you for response.Here I am giving
my config files and error message. Please see it.zaptel.conf contents:loadzone =
usdefaultzone=usfxsks=1-4zapata.conf
contents:[channels]context=incomingsignalling=fxs_ksbusydetect=1busycount=7relaxdtmf=yescallwaiting=yescallwaitingcallerid=yesthreewaycalling=yescancallforward=yesechocancelwhenbridged=yesrxgain=0.0txgain=0.0callerid=asreceivedlanguage=enusecallerid=yeshidecallerid=noechocancel=yestransfer=yesimmediate=nomusiconhold=defaultringtimeout=8000cidsignalling=dtmfcidstart=ringgroup=1callgroup=1pickupgroup=1channel
= 1sip.conf
contents:[105]type=friendusername=105secret=ravicallerid="RaviKanth"host=dynamiccontext=leadercanreinvite=nonat=yesdtmfmode=rfc2833allow=allextensions.conf
contents:[incoming]exten = s,1,Wait(4)exten =
s,n,Answerexten = s,n,SetMusicOnHold(default)exten =
s,n,Set(TIMEOUT(digit)=5)exten =
s,n,Set(TIMEOUT(response)=10)exten =
s,n,Background(/tmp/virg2)exten = s,n,Goto(s,1)exten =
s,n,Hangup()include = leader[leader]exten =
105,1,Dial(SIP/105,15)exten = 105,2,Voicemail(u105)exten =
105,3,Voicemail(b105)exten = 105,4,Hangupexten =
_9XX,1,Dial(Zap/1/${EXTEN:1}) ; Mobile phoneexten
= _5,1,Dial(Zap/1/${EXTEN:1}) ;
Local Landlineinclude = internal[internal]exten = 105,
1, Dial(SIP/105,15)When somebody calls from outside (Eg: mobile), I am
getting this below error message on Asterisk console:Error Message:Aug 17 19:45:41
ERROR[10449]: callerid.c:276 callerid_feed: fsk_serie made mylen 0
(-8)Aug 17 19:45:41 WARNING[10449]: chan_zap.c:6087 ss_thread:
CallerID feed failed: SuccessAug 17 19:45:41 WARNING[10449]:
chan_zap.c:6131 ss_thread: CallerID returned with error on channel
'Zap/1-1'Please tell me the solution. Looking forward to your kind
response. Thank you.Regards,Chandra.Rushowr
[EMAIL PROTECTED] wrote:
What's the Dial command being used to pass the call to
the Softphones?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Crazy
BoySent: Wednesday, August 16, 2006 3:23 AMTo:
[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] CallerID is not
displaying for my incoming calls
Hi,As you said, I have changed my configurations. But,
callerid is not displaying. What I have to do? Please tell
me.ThanksRegards,Chandra.Rich Adamson
[EMAIL PROTECTED] wrote:
Crazy
Boy wrote: Hi Friends, We have installed
Asterisk with Digium 04B card (4 FXO ports). Now, I have
connected my PSTN line directly to first port. I am making outgoing
calls and receiving incoming calls successfully through my
Asterisk. The problem is: When I am receiving a call from
outside (PSTN), I am not getting the callerid number and
getting callerid as "Asterisk" in my softphones (XLite). Here I
am giving my configuration files. zaptel.conf file
contents: loadzone = us defaultzone=us
fxsks=1-4 zapata.conf file contents:
[channels] context=incoming signalling=fxs_ks
busydetect=1 busycount=7 relaxdtmf=yes
callwaiting=yes callwaitingcallerid=yes
threewaycalling=yes cancallforward=yes
echocancelwhenbridged=yes rxgain=0.0 txgain=0.0
callerid=asreceived language=en usecallerid=yes
hidecallerid=no echocancel=yes transfer=yes
RE: [asterisk-users] Asterisk Real Time and sip.conf file used at thesame time
You CAN use both. You cannot use both if you tell asterisk to get the WHOLE sip configuration file from the database. But, in your case, realtime peers and users -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, August 16, 2006 7:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Carlos Chavez Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Real Time and sip.conf file used at thesame time I guess my problem might be that, because I pretend Asterisk to use my sip.conf static configuration file and also MySQL tables referenced in extconfig.conf like this: [settings] sipusers = mysql,asterisk,sip sippeers = mysql,asterisk,sip voicemail = mysql,asterisk,voicemail While I'm using one thing I can't use the other right??? Thanks once more, Ricardo. Quoting Carlos Chavez [EMAIL PROTECTED]: On Wed, 2006-08-16 at 19:03 +0100, Ricardo Carvalho wrote: Is it possible to use Asterisk RealTime and also config files (like sip.conf) at the same time? As much as I know, only one thing can be used and I need them both working!... Yes, you can use both at the same time. The only restriction is that you cannot use the realtime static configuration and realtime configuration. -- Carlos Chavez Prats Director de TecnologÃa Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Recent additions to the DigiumAsterisk development team
Just in case Murf doesn't get around to answering this one, I'll stab it... For one thing, I can code in a style that is similar to many programming languages, which can reduce the learning curve for many people, and personally I think it makes the code MORE readable because If statements follow a structure familiar to me and many others. Additionally, I personally think it's a little more manageable and have had clients remark on this as well after I show them comparable code from AEL2 and the traditional Asterisk DP code. As to your question about what can AEL do that an AGI/FastAGI app can't? I'll name the number one and number two items on my list: AEL doesn't require calling an external application AEL doesn't use an interpreted language such as PHP or Perl, both of which seem to be the languages of choice for most of you cats out there writing AGIs. AGI+Perl and AGI+PHP take a double performance hit compared to anything written purely in the dialplan, because an interface has to be opened between the two apps, control is passed back and forth, and the interpreter has to run against the script. AEL2 is pre-parsed by the AEL2 compiler and converted into Asterisk dialplan code when loaded into the system, not just in time. Anywho, that's just my two cents... Sherwood McGowan Consultant, AEL2 Zealot -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Wednesday, August 16, 2006 8:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Recent additions to the DigiumAsterisk development team Steve Murphy joined our development team at the beginning of June. Steve (murf on IRC/Mantis) had rewritten Asterisk's expression parser and the AEL language parser as a volunteer community member, along with various other bug fixes and improvements. Which makes me think, what is the real use of AEL. I have taken a look at it, and it makes asterisk's config file almost as unreadable as SER. What exactly does AEL do that a well written AGI / FastAGI app doesn't? I would think (but I'm surely wrong) that it would be better to do work on having well defined APIs that allow us to script Asterisk (such as AGI and the Manager interface) rather than invent Yet Another Pseudo Programming Language - which is going to be an endless task... Don't you think? That being said, just like the rest of the community, I'm very happy with Kevin's exciting announcement! Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with Hangup
I have to say that I'm experiencing the same issues, using the latest SVN -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chan Kwang Mien Sent: Monday, August 14, 2006 8:26 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problems with Hangup Hi, my test-bed is : sipphone -- Asterisk PBX -- PSTN -- Cell Phone sipphone was able to setup a connection to Cell Phone. When sipphone hangs up, Cell Phone also hangs up. However, when Cell Phone hangs up, sipphone was not able to hang up. Could it be that Asterisk was not able to recognise the hangup tone when the Cell Phone hangs up ? Does anyone know what the reason is ? Thank you. regards, Kwang Mien ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Problems with Hangup
My PSTN termination is through a provider, with a SIP connection between myself and their systems. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua Colp Sent: Monday, August 14, 2006 6:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Problems with Hangup - Original Message - From: Rushowr [mailto:[EMAIL PROTECTED] To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial Discussion' [mailto:[EMAIL PROTECTED] Sent: Mon, 14 Aug 2006 09:28:29 -0300 Subject: RE: [asterisk-users] Problems with Hangup I have to say that I'm experiencing the same issues, using the latest SVN You both need to enlighten us on what technology is in use to get to the PSTN. If it is analog and disconnect supervision is not available, then you'll be having fun. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chan Kwang Mien Sent: Monday, August 14, 2006 8:26 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Problems with Hangup Hi, my test-bed is : sipphone -- Asterisk PBX -- PSTN -- Cell Phone sipphone was able to setup a connection to Cell Phone. When sipphone hangs up, Cell Phone also hangs up. However, when Cell Phone hangs up, sipphone was not able to hang up. Could it be that Asterisk was not able to recognise the hangup tone when the Cell Phone hangs up ? Does anyone know what the reason is ? Joshua Colp Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Macro inside macro
Hey Attilla, thanks for the update. I'm also working on a solution, but
unfortunately the system I'm working with needs the separate macros. I'll
update the list if anything gets worked out.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Attilla De
Groot
Sent: Monday, August 14, 2006 12:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Macro inside macro
Well I solved the problem, by just making it one macro, not a macro inside
another one.
[macro-record]
exten = s,1,Setvar(CALLFILENAME=CALL-${ARG1}-${MACRO_EXTEN:4}-$
{TIMESTAMP})
exten = s,2,Monitor(wav,${CALLFILENAME},m})
exten = s,3,setcallerid(${ARG2})
exten = s,4,dial(${ARG3})
exten = h,1,System(/etc/asterisk/mail.sh [EMAIL PROTECTED] $
{CALLFILENAME} )
Regards,
Attilla
On Aug 13, 2006, at 11:21 PM, Attilla De Groot wrote:
On Aug 13, 2006, at 7:53 PM, Gonzalo Servat wrote:
I think what you probably want is:
exten = _*22*X.,1,Macro(record,conference,${EXTEN:4})
If you have _*23*., it means it will match *23 as well as
*23*, but not *23*123456 which is probably what you
want. Try:
exten = _*23*X.,1,Macro(record|dialout|31455200025|SIP/${EXTEN:4}
@voipbuster)
I made both changes, but that wasn't a problem. If I understand it
correctly, this only changes that the extension got more specific,
right ?
Also, from memory, the h extension gets executed from the main
context. After making the above changes, try adding this:
exten = _*23*X.,h,System(/etc/asterisk/mail.sh [EMAIL PROTECTED]
${CALLFILENAME} )
.. and remove the h extension from macro-record.
Let me know if the above helps.
I did make the changes, but it didn't solve the problem. It should
be h,1,System by the way ;)
I have some log here:
-- Executing Macro(SIP/attilla-0dce, record|dialout|31455200025|
SIP/[EMAIL PROTECTED]) in new stack
-- Executing SetVar(SIP/attilla-0dce, CALLFILENAME=CALL-
dialout-08001234-20060813-231921) in new stack
-- Executing Monitor(SIP/attilla-0dce, wav|CALL-
dialout-08001234-20060813-231921|m}) in new stack
-- Executing GotoIf(SIP/attilla-0dce, 0?macro-record|s|
4:macro-record|s|5) in new stack
-- Goto (macro-record,s,5)
-- Executing Macro(SIP/attilla-0dce, dialout|31455200025|SIP/
[EMAIL PROTECTED]) in new stack
-- Executing SetCallerID(SIP/attilla-0dce, 31455200025) in
new stack
-- Executing Dial(SIP/attilla-0dce, SIP/
[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/sip1.voipbuster.com-f715 is making progress passing it
to SIP/attilla-0dce
-- SIP/sip1.voipbuster.com-f715 answered SIP/attilla-0dce
== Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/
attilla-0dce' in macro 'dialout'
== Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/
attilla-0dce' in macro 'record'
== Spawn extension (macro-dialout, s, 2) exited non-zero on 'SIP/
attilla-0dce'
As you can see it executes everything perfectly and I was expecting
that after this the script would be executed.
Regards,
Attilla
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RE: [asterisk-users] Macro inside macro
I know I could do that, and I do for some instances, but the biggest point of writing the macros (at least in my case) is to reduce typing etc... Murf and I have uncovered a bug relating to this. Macros can call macros just fine pre-dial, but once there's been a hangup, we've discovered that Asterisk no longer processes the macros properly. A bug has been submitted to the digium bug system. Cheers, Sherwood McGowan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Monday, August 14, 2006 1:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Macro inside macro Any reason that you can't set variables before you use Gosub, then access them in the subroutine? Attilla De Groot wrote: On Aug 14, 2006, at 6:39 PM, Eric ManxPower Wieling wrote: Rushowr wrote: Hey Attilla, thanks for the update. I'm also working on a solution, but unfortunately the system I'm working with needs the separate macros. I'll update the list if anything gets worked out. pbx-1*CLI show application gosub pbx-1*CLI -= Info about application 'Gosub' =- [Synopsis] Jump to label, saving return address [Description] Gosub([[context|]exten|]priority) Jumps to the label specified, saving the return address. pbx-1*CLI Already considered this option, but I want to give it some arguments. And that isn't possible with gosub. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk and PHP?
AGI+PHP would be a good place to do all of this. However, be aware that interpreted code such as PHP incurs a performance hit and may not be suitable for very large installations, in addition to the issue of passing call control away from Asterisk in general. (ref: "Asterisk Performance", Joachim Vanheuverzwijn/Zoa). If I'm mistaken, somebody please correct me :) I'm mistaken, please, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lennie De VilliersSent: Monday, August 14, 2006 5:09 PMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Asterisk and PHP? Hi, I found: http://www.voip-info.org/wiki/view/Asterisk+AGI+php Is there more such examples or tutorials available that show me how to control Asterisk using PHP? I want to be able to have full control over Asterisk using PHP. For example: * Execute PHP code when there's an incoming call. * Handle or control the incoming call via PHP code. * Execute PHP code when done with incoming call. * Etc... Kind Regards, Lennie De Villiers ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] In CDR record not what I want
It's because the standard CDR engine uses the last ${EXTEN} value as the
destination number
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthias
Fechner
Sent: Friday, August 11, 2006 6:08 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] In CDR record not what I want
Hi,
I have the following rules:
exten = 4441,1,NoOp(--- ${CALLERID} calling on capi-extern (${EXTEN}) ---)
exten = 4441,2,Goto(dialin-privat,s,1) exten = 4441,3,Hangup
[dialin-privat]
; Log incoming calls
exten = s,1,LDAPget(CALLERIDNAME=daheim) exten =
s,2,NoOP(--CALLERID=-${CALLERID}-, CALLERIDNUM=-${CALLERIDNUM}-,
EXTEN=-${EXTEN}--) ...
my CDR records says now that a call from unkown to s happened.
Is it possible that in the CDR record the number which has been called is
saved and not s?
e.g.
number unkown called 4441
Best regards,
Matthias
--
Programming today is a race between software engineers striving to build
bigger and better idiot-proof programs, and the universe trying to produce
bigger and better idiots. So far, the universe is winning. -- Rich Cook
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RE: [asterisk-users] Auto retry on Busy
The reason he might want it is because it's a feature offered by many POTS
and Mobile Telcos. I know that's why I've played with it, the ITSPs and SIP
Termination providers I consult for want to have as many if not more
features to offer than the POTS and Mobile guys.
Cheers,
Rushowr - Sherwood McGowan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin Smith
Sent: Friday, August 11, 2006 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto retry on Busy
Why don't you just test for the dial status after the dial command
completes? I don't really see why you want something to keep dialing until
it gets through, but this would work.
[something]
1,1,Dial(zap/,sip/, etc/whatever, 10)
1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
1,n(LINEBUSY), Wait(30)
1,n,goto(something,1,1)
1,n(OTHER), do something else
Sure it is pretty rough, but the basics are there. Also you might want to
read this: http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Kevin
Noah Silverman wrote:
Hi,
Does anybody have an easy solution for this.
I want something that will keep trying a busy number every 30 seconds
until it gets through.
I've tried retrydial, but can't get it to work.
Any suggestions?
Thanks,
-N
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RE: [asterisk-users] Auto retry on Busy
Yep, my point exactly.
Since I'm in the middle of another ITSP project, I'll be hitting this again,
and will share anything I come up with. I've had thoughts, but never tested
it.
SHerwood
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Novack
Sent: Friday, August 11, 2006 3:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto retry on Busy
Also many so-called legacy hybrid PBX switches have had this for many a
year Hard to compete when well used features that have been around for 20
years are lacking
John Novack
Rushowr wrote:
The reason he might want it is because it's a feature offered by many POTS
and Mobile Telcos. I know that's why I've played with it, the ITSPs and SIP
Termination providers I consult for want to have as many if not more
features to offer than the POTS and Mobile guys.
Cheers,
Rushowr - Sherwood McGowan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin
Smith
Sent: Friday, August 11, 2006 2:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Auto retry on Busy
Why don't you just test for the dial status after the dial command
completes? I don't really see why you want something to keep dialing
until it gets through, but this would work.
[something]
1,1,Dial(zap/,sip/, etc/whatever, 10)
1,n,gotoif[${DIALSTATUS}=BUSY]?(LINEBUSY):(OTHER)
1,n(LINEBUSY), Wait(30)
1,n,goto(something,1,1)
1,n(OTHER), do something else
Sure it is pretty rough, but the basics are there. Also you might want
to read this:
http://www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Kevin
Noah Silverman wrote:
Hi,
Does anybody have an easy solution for this.
I want something that will keep trying a busy number every 30 seconds
until it gets through.
I've tried retrydial, but can't get it to work.
Any suggestions?
Thanks,
-N
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RE: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found
Uh, what's your Register statement for those SIP DIDs look like? If you don't specify the number after a /, you'll be handed calls for that line, but specifying 's' as the extension. register = user[:secret[:[EMAIL PROTECTED]:port][/extension] I consider that last argument required anymore Sherwood -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Friday, August 11, 2006 3:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inbound Calls SIP/2.0 404 Not Found You might want to take a look at 'sip debug' to see what your provider is actually sending you. Its likely they aren't sending you the 9495551212 sting as you are expecting. Thanks - Just to be clear - I just replaced the real digits with - I want to direct these to specific extensions. So maybe I should have used or something else? I tried this: exten=_9495551212,1, Goto(mainmenu,s,1) But still to no avail. On 8/11/06, Vadim Berezniker [EMAIL PROTECTED] wrote: Perhaps the context in sip.conf doesn't match the context in the dial plan. I'm trying to get inbound DIDs working via SIP. I have 20 DIDs coming in via a single SIP profile in sip.conf. I was hoping to have these matched in extensions.conf, so I have setup lines like this: exten=949271,1, Goto(mainmenu,s,1) Unfortunately these aren't getting matched and I'm getting this error: Looking for s in druid-default (domain 949271) SIP/2.0 404 Not Found Any hints or tips? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Realtime SIP Authentication
username + secret From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Thursday, August 10, 2006 7:53 AMTo: asterisk-users@lists.digium.comSubject: [asterisk-users] Realtime SIP Authentication Hi All,I'm using Realtime for SIP users and I looking to find a way to be able to authenticate users based on both the username and IP of the incoming call the reason being I have different users connecting from same IP but using different usernames.I have read that setting type=peer is only matched on IP address/port.Is it possible to configure Realtime to match on username and IP? Ron. Check Out the new free AIM(R) Mail -- 2 GB of storage and industry-leading spam and email virus protection. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
