[asterisk-users] Asterisk PBX causes mysql to take more CPU time
Hi All, I am using asterisk-1.4.22 with mysql as my storage medium for voice messages.Right now i am running 700+ extensions with this setup . *System Configuration:* asterisk-1.4.22.1 (using odbc storage for voice messages) unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 mysql-server-5.0.45-7.el5 CentOS-5.2 (64 bit) *The following are the issues i am facing:* 1) CPU time always shows 30%-90% for mysql when asterisk is running . The CPU time goes normal for mysql when asterisk is stopped. 2) calls are getting dropped suddenly. 3) Some answered calls does not hear anything at the receiver/initiater end. 4) asterisk itself gets crashed at times 5) Core dumps are getting created. Is there any optimizations or configurations needs to be done on my server in order to drill down this issue? Please help me out in troubleshooting this issue, This problems are getting worsen day by day. Thanks in advance. -- always withsmile vimurli ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX causes mysql to take more CPU time
MURALI V escribió: Hi All, I am using asterisk-1.4.22 with mysql as my storage medium for voice messages.Right now i am running 700+ extensions with this setup . *System Configuration:* asterisk-1.4.22.1 (using odbc storage for voice messages) unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 mysql-server-5.0.45-7.el5 CentOS-5.2 (64 bit) *The following are the issues i am facing:* 1) CPU time always shows 30%-90% for mysql when asterisk is running . The CPU time goes normal for mysql when asterisk is stopped. Obviously the MySQL CPU time droppes to near zero if the only process who uses it stops! Check what kind of queries asterisk is making and if that can be optimized. 2) calls are getting dropped suddenly. 3) Some answered calls does not hear anything at the receiver/initiater end. 4) asterisk itself gets crashed at times 5) Core dumps are getting created. Follow the guidelines for obtaining a backtrace with GDB. These can be found in the file asterisk/doc/backtrace.txt Look in http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging If you learn to read the backtrace, you can know with great detail what is causing asterisk to crash. You may run into known bugs, so you can check the bugtracker and/or the recent versions Changelog to see if it's solved. Maybe you'll have to decide to upgrade it or not, or upgrade in a separate machine and test throughoutly before putting the new version into production. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
On Sat, Jul 12, 2008 at 01:14:36AM +0100, Grey Man wrote: On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis [EMAIL PROTECTED] wrote: I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. There's already 4 public images on ec2 mentioning Asterisk in their names so wouldn't it be easier to try out one of those rather than install all the bits and pieces on a base Linux image? An interesting paper on ec2 and Asterisk would be one that discusses what the call quality is like from both inside and outside the US. When I briefy ran up an instance at this time last year it actually seemed ok. From a provider's point of view running Asterisk on the ec2 cloud does pose some interesting questions. As a quick and dirty estimate if you assume one of the standard small ec2 instances could cope with 100 simultaneous g711 calls (I don't know if that is the case just guessing) then you'll chew up approx. 2MB/s (you pay for bandwidth both ways). Assuming that you'd then have 1MB/s average to account for quite and busy call times then it would be 3.6GB/hour or 86.4GB/day. At the Amazon price of $0.10/GB that's $8.64/day or $260/month. The server instance will cost you $72/month so total cost for 100/calls per month is $332. A typical dedicated server for $300/month is roughly equivalent to an ec2 small instance and comes with 500GB of bandwidth/month which is only a fifth of what's required but you could probably get the extra 2TB/month thrown in for $32/month making the dedicated server and ec2 prices the same. Another option, especially for lighter loads, is a virtual server instance. There are many providers who will provide you a Xen guest for e.g. 20$ or 30$ a month. For a lighter load this should be more than enough. Those normally come with a dedicated IP address. Support plans vary greatly from one provider to another. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. Thanks! Ronald Lewis Denver, Colorado http://ronaldlewis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
Very cool, you've piqued my interest. Since I haven't launched an instance before, where's the best place to learn to do that? What's the approximate monthly cost of hosting an Asterisk PBX on EC2? Ronald Lewis wrote: I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. Thanks! Ronald Lewis Denver, Colorado http://ronaldlewis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis [EMAIL PROTECTED] wrote: I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. There's already 4 public images on ec2 mentioning Asterisk in their names so wouldn't it be easier to try out one of those rather than install all the bits and pieces on a base Linux image? An interesting paper on ec2 and Asterisk would be one that discusses what the call quality is like from both inside and outside the US. When I briefy ran up an instance at this time last year it actually seemed ok. From a provider's point of view running Asterisk on the ec2 cloud does pose some interesting questions. As a quick and dirty estimate if you assume one of the standard small ec2 instances could cope with 100 simultaneous g711 calls (I don't know if that is the case just guessing) then you'll chew up approx. 2MB/s (you pay for bandwidth both ways). Assuming that you'd then have 1MB/s average to account for quite and busy call times then it would be 3.6GB/hour or 86.4GB/day. At the Amazon price of $0.10/GB that's $8.64/day or $260/month. The server instance will cost you $72/month so total cost for 100/calls per month is $332. A typical dedicated server for $300/month is roughly equivalent to an ec2 small instance and comes with 500GB of bandwidth/month which is only a fifth of what's required but you could probably get the extra 2TB/month thrown in for $32/month making the dedicated server and ec2 prices the same. There are serious pros and cons between these approaches. With the ec2 you don't get a permanent static IP, with a dedicated server you do. With ec2 you could scale up and down between 1 server and 4 servers at the drop of a hat to save costs and cope with peak and quite times, with dedicated servers you're stuff with 12 or 24 month contracts for the number of servers you'd need under maximum load. And then of course the major factor for both is what the call quality will be like. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
Strange prices, look at ovh, 233€/m~~350$ And ovh provide a REAL unlimited. CPU Intel Xeon X5355 1x 4x 2.66 GHz L2: 8Mo, FSB: 1333MHzQuadruple Coeur Architecture64 bits RAM 8 Go FBDIMM DDR2 HDD 2x 750 Go Type HDD SATA2 RAID HARD 1Interfaces 2 x 1 Gbps SPEED 2 Gbps Traffic UNLIMITED IP fixe2 adresses IP Fail-over http://www.ovh.com/fr/items/ip_failover.xml+8 adressesVPS Ready http://www.ovh.com/fr/produits/offres_vps.xml[image: Oui]IP Fail-over VPS64 IP (/26)IP enregistrées RIPE[image: Oui] Sauvegarde FTP http://www.ovh.com/fr/items/sauvegarde_ftp.xml750 Go 2008/7/12 Grey Man [EMAIL PROTECTED]: On Fri, Jul 11, 2008 at 7:50 PM, Ronald Lewis [EMAIL PROTECTED] wrote: I've just added a PREVIEW release of my upcoming how-to guide for Asterisk PBX on EC2. It is based on months of testing and evaluating Asterisk on EC2. It addresses all kinks and showstoppers that many people have experienced over the past year or so. Because this is a preview, it is not the final version of this guide. It is subject to change (format, copy, layout, etc.) To view and download this guide, please visit http://ronaldlewis.com/2008/07/08/asterisk-pbx-on-amazon-ec2-how-to-guide-almost-complete/ Please take this opportunity to test the guide and provide any feedback. The official release is set for Wednesday, July 16 and will be available on CloudCrunch. There's already 4 public images on ec2 mentioning Asterisk in their names so wouldn't it be easier to try out one of those rather than install all the bits and pieces on a base Linux image? An interesting paper on ec2 and Asterisk would be one that discusses what the call quality is like from both inside and outside the US. When I briefy ran up an instance at this time last year it actually seemed ok. From a provider's point of view running Asterisk on the ec2 cloud does pose some interesting questions. As a quick and dirty estimate if you assume one of the standard small ec2 instances could cope with 100 simultaneous g711 calls (I don't know if that is the case just guessing) then you'll chew up approx. 2MB/s (you pay for bandwidth both ways). Assuming that you'd then have 1MB/s average to account for quite and busy call times then it would be 3.6GB/hour or 86.4GB/day. At the Amazon price of $0.10/GB that's $8.64/day or $260/month. The server instance will cost you $72/month so total cost for 100/calls per month is $332. A typical dedicated server for $300/month is roughly equivalent to an ec2 small instance and comes with 500GB of bandwidth/month which is only a fifth of what's required but you could probably get the extra 2TB/month thrown in for $32/month making the dedicated server and ec2 prices the same. There are serious pros and cons between these approaches. With the ec2 you don't get a permanent static IP, with a dedicated server you do. With ec2 you could scale up and down between 1 server and 4 servers at the drop of a hat to save costs and cope with peak and quite times, with dedicated servers you're stuff with 12 or 24 month contracts for the number of servers you'd need under maximum load. And then of course the major factor for both is what the call quality will be like. Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX How-to Guide for Amazon EC2
On Sat, Jul 12, 2008 at 2:16 AM, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote: Strange prices, look at ovh, 233€/m~~350$ And ovh provide a REAL unlimited. And I'm sure there will be someone somewhere who has a better deal as well. Whenever hosting gets mentioned in tech forums you always end up with lots of opinions (or pitches depending on how well they are disguised). My post was more in the vein of ballpark figures to compare the technincal merits of a grid approach, in this case manifested by Amazon's ec2, to a dedicated server approach for running a VoIP provider service using Asterisk. It's probably already gone off the OP's original topic enough now that they'll be people jumping up and down that this should be on the Biz list... Regards, Greyman. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk PBX using Outbound proxy
On 18/04/2008, Rosa De Santis [EMAIL PROTECTED] wrote: Hi all. Please, how can I configure an Asterisk PBX using an outbound proxy (that resolve NAT Traversal) I'm trying using the outboundproxy and outboundproxyport values in sip.conf but the PBX don't get registered on the outbound proxy side. I'm using SER + Asterisk with Jasomi outbound proxy solution, and I want the PBX to have a SIP trunk, but in SER i see the pbx sip user registered as [EMAIL PROTECTED] , not the SER ip. Any ideas please? Thanks in advance, Rosa. What register = line are you using? Take a look at: http://bugs.digium.com/view.php?id=12474 Perhaps it will help? Regards, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PBX using Outbound proxy
Hi all. Please, how can I configure an Asterisk PBX using an outbound proxy (that resolve NAT Traversal) I'm trying using the outboundproxy and outboundproxyport values in sip.conf but the PBX don't get registered on the outbound proxy side. I'm using SER + Asterisk with Jasomi outbound proxy solution, and I want the PBX to have a SIP trunk, but in SER i see the pbx sip user registered as [EMAIL PROTECTED] , not the SER ip. Any ideas please? Thanks in advance, Rosa. _ Blog your life in 3D with Windows Live Writer. http://www.windowslive.com/overview.html?ocid=TXT_TAGLM_Wave2_wl_writer_022008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk PBX '' '||' Grandstream GXP-2000 problem
Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHz Box... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to hold and proceed to answer line 2 and forward line 2 to to the requested extension. The problem is when they attempt to pick line 1 off the hold in order to handle that call, line 1 is either dropped or the Grandstream Phone freezes and the user is forced to rest the phone. The situation persist whenever there are multiple lines active with incoming calls and upon answering one, placing the line on hold and attempting to answer the other lines active calls will be dropped the the phone just hangs/freezes. We know that the call is dropped because the people call back complaining about being hung up on We have had our dedicated T1 (for voice only) tested several times and it is good. We have had the Asterisk PBX completely redone and gone over thoroughly and are at the point where we are suspecting the configuration file for the Grandstream GXP-2000 Telephone as the culprit. We would like to know what suggestions anyone out there might have if any... Thanks, Jose P. Espinal DomiNET ___ --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 PBX '' '||' Grandstream GXP-2000 problem
On 1/15/07, J. Espinal [EMAIL PROTECTED] wrote: Hi People, We use the Grandstream GXP-2000 phones, firmware 1.1.1.14, Asterisk PBX, Slackware Linux 10.2, loaded on a Intel(R) Pentium(R) 4 CPU 2.66GHzBox... The issues that we are experiencing involves our Telephone Operator's/Receptionist whom answer multiple incoming calls... As an example.., when they answer line 1 and Line 2 starts to ring they would ask the person on line 1 to hold and proceed to answer line 2 and forward line 2 to to the requested extension. The problem is when they attempt to pick line 1 off the hold in order to handle that call, line 1 is either dropped or the Grandstream Phone freezes and the user is forced to rest the phone. The situation persist whenever there are multiple lines active with incoming calls and upon answering one, placing the line on hold and attempting to answer the other lines active calls will be dropped the the phone just hangs/freezes. We know that the call is dropped because the people call back complaining about being hung up on We have had our dedicated T1 (for voice only) tested several times and it is good. We have had the Asterisk PBX completely redone and gone over thoroughly and are at the point where we are suspecting the configuration file for the Grandstream GXP-2000 Telephone as the culprit. We would like to know what suggestions anyone out there might have if any... Thanks, Are you using G.729? Last I heard, grandstreams could only have one call via G.729 at a time. It had something to do with the licensing that they used, I think. Just a thought... ___ --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 PBX to a Nortel MICS PBX
Can you be more specific? What sort of linkages are available between the two offices? CP On 22-Oct-06, at 10:38 PM, dthurn wrote: What's the best way to connect an Asterisk PBX to a Nortel MICS PBX. I have two offices that I want to link together. TTFN ___ --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
[asterisk-users] Asterisk PBX to a Nortel MICS PBX
What's the best way to connect an Asterisk PBX to a Nortel MICS PBX. I have two offices that I want to link together. TTFN ___ --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 PBX in Debian
Hi Please use proper quoting... See below On Sat, Oct 08, 2005 at 12:23:21AM -0400, [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, October 07, 2005 8:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk PBX in Debian On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote: Besides I found that using packages with asterisk on debian can do odd things to your custom sound files if you do a remove. Regarding the sounds files: I don't think that the way Asterisk installer handles them is very optimal either. Your message got me thinking, though. I believe that Debian is right installing all sounds to /usr/share/asterisk/sounds . But /var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be kept for custom sounds that are never touched by the package. I figure that file.c:build_filename could be changed to do the following: if exists /var/lib/asterisk/sounds/filename return /var/lib/asterisk/sounds/filename else if exists /usr/share/sounds/asterisk/filename return /usr/share/sounds/asterisk/filename What do you think? I figure I'll try to push this into Debian first. (If this is indeed a good idea) Using /var works, but setting it in asterisk could be a pain when it comes to voicemail prompts. Plus, extensions.conf would need to grow and become a little cluttered. Unless of course, one could do something to specify a new root voicemail path, and if the file is not found it plays from the default. You missed the point: you still keep the same configuration. filename should first be looked for in the custom directory and only afterwards in /usr/share/sounds/asterisk/sounds . voicemail/ is currently (in Debian) a symlink from the sounds directory to /var/lib/asterisk/voicemail -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX in Debian
On Fri, Oct 07, 2005 at 12:51:25AM -0500, Carlos Prieto wrote: Hi everyone. I've installed Asterisk PBX using apt packages, but i don't have actually any Digium card, so i want to use ztdummy. I've tried to modify the Makefiles in the debian source package, i don't get any error, but still the ztdummy module doesn't get compiled. What file exactly did you edit? What command did you run? I suspect the file you have edited got overrun. Does anybody has idea how to get the ztdummy module using the debian package system? I'm not sure you need. Packages from deb http://rapid.dotsrc.org/rapid sarge main already have ztdummy and ztdummy is on by default in the zaptel-source package. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX in Debian
Why bother with packages anyhow? I just installed debian base and did a cvs get for head, and all good to go. Besides I found that using packages with asterisk on debian can do odd things to your custom sound files if you do a remove. Regards, Greg -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, October 07, 2005 7:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk PBX in Debian On Fri, Oct 07, 2005 at 12:51:25AM -0500, Carlos Prieto wrote: Hi everyone. I've installed Asterisk PBX using apt packages, but i don't have actually any Digium card, so i want to use ztdummy. I've tried to modify the Makefiles in the debian source package, i don't get any error, but still the ztdummy module doesn't get compiled. What file exactly did you edit? What command did you run? I suspect the file you have edited got overrun. Does anybody has idea how to get the ztdummy module using the debian package system? I'm not sure you need. Packages from deb http://rapid.dotsrc.org/rapid sarge main already have ztdummy and ztdummy is on by default in the zaptel-source package. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX in Debian
On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote: Why bother with packages anyhow? I just installed debian base and did a cvs get for head, and all good to go. And if you have several systems? Besides I found that using packages with asterisk on debian can do odd things to your custom sound files if you do a remove. Regarding the sounds files: I don't think that the way Asterisk installer handles them is very optimal either. Your message got me thinking, though. I believe that Debian is right installing all sounds to /usr/share/asterisk/sounds . But /var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be kept for custom sounds that are never touched by the package. I figure that file.c:build_filename could be changed to do the following: if exists /var/lib/asterisk/sounds/filename return /var/lib/asterisk/sounds/filename else if exists /usr/share/sounds/asterisk/filename return /usr/share/sounds/asterisk/filename What do you think? I figure I'll try to push this into Debian first. (If this is indeed a good idea) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX in Debian
Or why not create all sound files under /usr/share/asterisk/sounds and then subdirs from there for your own touched files i.e. /usr/share/asterisk/sounds/custom ? -- Michael Coburn -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, October 07, 2005 8:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk PBX in Debian On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote: Why bother with packages anyhow? I just installed debian base and did a cvs get for head, and all good to go. And if you have several systems? Besides I found that using packages with asterisk on debian can do odd things to your custom sound files if you do a remove. Regarding the sounds files: I don't think that the way Asterisk installer handles them is very optimal either. Your message got me thinking, though. I believe that Debian is right installing all sounds to /usr/share/asterisk/sounds . But /var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be kept for custom sounds that are never touched by the package. I figure that file.c:build_filename could be changed to do the following: if exists /var/lib/asterisk/sounds/filename return /var/lib/asterisk/sounds/filename else if exists /usr/share/sounds/asterisk/filename return /usr/share/sounds/asterisk/filename What do you think? I figure I'll try to push this into Debian first. (If this is indeed a good idea) -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX in Debian
Michael Coburn wrote: Or why not create all sound files under /usr/share/asterisk/sounds and then subdirs from there for your own touched files i.e. /usr/share/asterisk/sounds/custom ? -- Michael Coburn Short answer is because we live by rules. Any good linux distro follows standards regarding the proper location of files. /usr/local /opt and /var/lib would be more appropriate for locally added sounds. This is good reading for people new to linux. Maybe you always wondered why there is a /bin, /usr/bin and /usr/local/bin? This will enlighten you: http://www.debian.org/doc/packaging-manuals/fhs/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, October 07, 2005 8:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk PBX in Debian On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote: Why bother with packages anyhow? I just installed debian base and did a cvs get for head, and all good to go. And if you have several systems? Besides I found that using packages with asterisk on debian can do odd things to your custom sound files if you do a remove. Regarding the sounds files: I don't think that the way Asterisk installer handles them is very optimal either. Your message got me thinking, though. I believe that Debian is right installing all sounds to /usr/share/asterisk/sounds . But /var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be kept for custom sounds that are never touched by the package. I figure that file.c:build_filename could be changed to do the following: if exists /var/lib/asterisk/sounds/filename return /var/lib/asterisk/sounds/filename else if exists /usr/share/sounds/asterisk/filename return /usr/share/sounds/asterisk/filename What do you think? I figure I'll try to push this into Debian first. (If this is indeed a good idea) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX in Debian
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Friday, October 07, 2005 8:17 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk PBX in Debian On Fri, Oct 07, 2005 at 07:37:26AM -0400, [EMAIL PROTECTED] wrote: Why bother with packages anyhow? I just installed debian base and did a cvs get for head, and all good to go. And if you have several systems? I would make a custom package in that case, for easy updating. Depends of course if you are using head or not. Besides I found that using packages with asterisk on debian can do odd things to your custom sound files if you do a remove. Regarding the sounds files: I don't think that the way Asterisk installer handles them is very optimal either. Your message got me thinking, though. I believe that Debian is right installing all sounds to /usr/share/asterisk/sounds . But /var/lib/asterisk/sounds (/usr/local/asterisk/sounds? ) should still be kept for custom sounds that are never touched by the package. I figure that file.c:build_filename could be changed to do the following: if exists /var/lib/asterisk/sounds/filename return /var/lib/asterisk/sounds/filename else if exists /usr/share/sounds/asterisk/filename return /usr/share/sounds/asterisk/filename What do you think? I figure I'll try to push this into Debian first. (If this is indeed a good idea) Using /var works, but setting it in asterisk could be a pain when it comes to voicemail prompts. Plus, extensions.conf would need to grow and become a little cluttered. Unless of course, one could do something to specify a new root voicemail path, and if the file is not found it plays from the default. Greg ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX in Debian
Hi everyone. I've installed Asterisk PBX using apt packages, but i don't have actually any Digium card, so i want to use ztdummy. I've tried to modify the Makefiles in the debian source package, i don't get any error, but still the ztdummy module doesn't get compiled. Does anybody has idea how to get the ztdummy module using the debian package system? Thanks for your help. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX
Hi List I am very new to Asterisk but have been alloted a job to replace my traditional PBX with it. Kindly provide me some useful info (PDF's etc) to setup Asterisk with FXO and FXS both. I have to cater some 60 users with 10 simultaneous calls. Regards _ Biography of Shah Rukh. His profile, awards, films. http://server1.msn.co.in/Profile/shahrukh.asp Find more here! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX
Hi Kapil, AFAIK, there are no such PDF's that exist unless someone has really spent time compiling such information, which will be great to see. However, if you check out www.voip-info.org, its a complete mine of useful information regarding doing what you wish to. Regards, Sahil Gupta VoiceValley On Wed, 21 Sep 2005, kapil dhawan wrote: Hi List I am very new to Asterisk but have been alloted a job to replace my traditional PBX with it. Kindly provide me some useful info (PDF's etc) to setup Asterisk with FXO and FXS both. I have to cater some 60 users with 10 simultaneous calls. Regards _ Biography of Shah Rukh. His profile, awards, films. http://server1.msn.co.in/Profile/shahrukh.asp Find more here! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX
On Wed, 21 Sep 2005, kapil dhawan wrote: Hi List I am very new to Asterisk but have been alloted a job to replace my traditional PBX with it. Kindly provide me some useful info (PDF's etc) to setup Asterisk with FXO and FXS both. I have to cater some 60 users with 10 simultaneous calls. Regards Go to asteriskathome.sourceforge.net and download [EMAIL PROTECTED] Be *VERY*** *VERY* *VERY* careful with the CD you burn, though, as when booted it will erase your hard drive and install CentOS ***WITHOUT WARNING Other than that, it is a very good way to replace a PBX for an office of that size. Of course, you will need to select phones, server hardware, PSTN interconnect hardware, etc. as well. Considering how important phones are to the average business, you might want to consider hiring a consultant (might I recommend cough cough me?) to help get you up to speed. It's just a thought, but getting help from someone who has already done this might keep you from making a few expensive mistakes (ie: buying equipment that is over/ underpowered, unreliable, low quality, etc.) If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this list will be your best tools. Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX
On Wed, September 21, 2005 15:14, Tom Rymes said: On Wed, 21 Sep 2005, kapil dhawan wrote: SNIP If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this list will be your best tools. Tom I'd like to add Google to that shortlist: Searchphrase + site:voip-info.org or Searchphrase + site:lists.difium.com will help you quickly search the wiki and list archives... Good luck! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX
On Wed, September 21, 2005 15:19, Francesco Peeters said: On Wed, September 21, 2005 15:14, Tom Rymes said: On Wed, 21 Sep 2005, kapil dhawan wrote: SNIP If you choose to do it yourself, [EMAIL PROTECTED], voip-info.org, and this list will be your best tools. Tom I'd like to add Google to that shortlist: Searchphrase + site:voip-info.org or Searchphrase + site:lists.difium.com will help you quickly search the wiki and list archives... Good luck! Oops! Typo! Searchphrase + site:lists.digium.com is the correct syntax... Sorry! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX
Searchphrase + site:lists.difium.com The above is good when searching for information on Joe Diffie -- Otherwise, you'll want: Searchphrase + site:lists.digium.com :) Nathan -- - Nathan E. Pralle Give the Director a Serpent Deflector www.nathanpralle.com - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX
On Wed, September 21, 2005 15:30, Nathan Pralle said: Searchphrase + site:lists.difium.com The above is good when searching for information on Joe Diffie -- Otherwise, you'll want: Searchphrase + site:lists.digium.com :) Nathan G I already corrected myself... I canna help them list servers take so long! ;-) ---FP ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk PBX and Siemens Hipath 3750
Hello I am planning to build a small PBX using TDM22B. We have a Siemens Hipath 3750 in operation already. When I manage to complete my PBX using TDM22B I would ofcourse like to be able to connect my Asterisks PBX with the Siemens Hipath 3750 PBX. Will there be any issues regarding my plan ? Or is there any other issues that I need to take into account vis-a-vis Siemens PBX. I have never done all this before so I would appreciate any inputs. Thanks in advance Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX with X100P in India
I'm currently trying to set up an Asterisk PBX system in India. However I'm having trouble configuring the X100P to dial out on the POTS line. Does anyone have any knowledge about this? I know the telephone system is a bit different in India, so would the X100P not be suitable? Is there a change I need to make in the Zaptel.conf or zapata.conf? Because I'm using [EMAIL PROTECTED] 0.6, the extensions.conf is also pretty frustrating... Any help here would be appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX with X100P in India
what kind of problems do u have? can u explain more in detail so we can try helping you? best regards El vie, 15-04-2005 a las 01:29 -0700, Min Hwan Chang escribió: I'm currently trying to set up an Asterisk PBX system in India. However I'm having trouble configuring the X100P to dial out on the POTS line. Does anyone have any knowledge about this? I know the telephone system is a bit different in India, so would the X100P not be suitable? Is there a change I need to make in the Zaptel.conf or zapata.conf? Because I'm using [EMAIL PROTECTED] 0.6, the extensions.conf is also pretty frustrating... Any help here would be appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX with X100P in India
My Setup is below:
X100P connected to POTS
Siemens IP Phone
[EMAIL PROTECTED] v0.6
When I try dialing out the POTS line in India using a Siemens IP
phone, I'll hear the phone ring for a second, then hear loud beeps.
The loud beeps I believe are from the POTS line?
I am currently trying to dial a Cell phone number within India, so the
general format is 09X.
I have my extensions.conf set up like so for Outbound Context:
--
[outbound-local]
exten = _X,1,Macro(dialout-default,09${EXTEN})
[outbound-ld]
exten = _001NXXNXX,1,Macro(dialout-default,${EXTEN})
-
For the life of me I can't' seem to figure out what the problem is.
Thanks for the help!
Regards,
Min
On 4/15/05, Julio Saura [EMAIL PROTECTED] wrote:
what kind of problems do u have?
can u explain more in detail so we can try helping you?
best regards
El vie, 15-04-2005 a las 01:29 -0700, Min Hwan Chang escribió:
I'm currently trying to set up an Asterisk PBX system in India.
However I'm having trouble configuring the X100P to dial out on the
POTS line. Does anyone have any knowledge about this?
I know the telephone system is a bit different in India, so would the
X100P not be suitable? Is there a change I need to make in the
Zaptel.conf or zapata.conf?
Because I'm using [EMAIL PROTECTED] 0.6, the extensions.conf is also
pretty frustrating...
Any help here would be appreciated.
___
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[Asterisk-Users] Asterisk PBX Manager
Title: Asterisk PBX Manager Does anyone on this list have any experience Thirdlane.com's Asterisk PBX Manager? And if so what do you think of it? Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Manager
I didn't test it on a live system. Just on their demo but it looks very good. On Tue, 1 Mar 2005 11:46:40 -0500, Michael Di Martino [EMAIL PROTECTED] wrote: Does anyone on this list have any experience Thirdlane.com's Asterisk PBX Manager? And if so what do you think of it? Regards, Michael DiMartino Director of MIS The telx Group, Inc. 17 State St, 33rd Floor New York, NY 10004 T: 212.480.3300 X2022 C: 646.207.6603 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Pbx Manager Equivalent
http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from Thirdlane) looks like a great program for eye candy configuration of Asterisk. However it costs lost of $, and Im currently only an experimenter so to speak. Anyone advice of a decent alternative that is similar?? Currently, we only have VOIP connections, but will have a couple of Digium fxs/fxos soon to have a play with, so would be advantageous if it worked with these too Finally, Anyone know of a Digium hardware Reseller in the Uk at all?? Thx Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Pbx Manager Equivalent (in plain text - apologies to those that dont like HTML mail!!)
http://www.thirdlane.com/screenshots.htm (Asterisk PBX Manager from Thirdlane) looks like a great program for eye candy configuration of Asterisk. However it costs lost of $, and I'm currently only an experimenter so to speak. Anyone advice of a decent alternative that is similar?? Currently, we only have VOIP connections, but will have a couple of Digium fxs/fxo's soon to have a play with, so would be advantageous if it worked with these too. Finally, Anyone know of a Digium hardware Reseller in the Uk at all?? Thx Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX Manager
Is there anything open source out there that has the same or better feature set than Asterisk PBX Manager ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter Sent: Tuesday, November 30, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Manager I have this, it comes as a webmin module. I also got it with the intention of bundling it for clients. It costs $300 , and the license is tied to the NIC. While it wont do EVERYTHING, it will probably be sufficient for the user to set up extensions/phones/menus/voicemail/conferences. One thing that I am not happy with, is that it allows raw editing of the conf files. Gawd help us if a user gets into that lot. I emailed Third lane, and they replied staright away with an address where I could download an evaluation. I'd publish the url here, but there must be a reason why they don't show it on their web site. Oh, and by the way (this from a beginner), I found it by searching on the WIKI Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX Manager
AMP but you already knew that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Brecher Sent: Wednesday, December 01, 2004 6:28 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk PBX Manager Is there anything open source out there that has the same or better feature set than Asterisk PBX Manager ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter Sent: Tuesday, November 30, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Manager I have this, it comes as a webmin module. I also got it with the intention of bundling it for clients. It costs $300 , and the license is tied to the NIC. While it wont do EVERYTHING, it will probably be sufficient for the user to set up extensions/phones/menus/voicemail/conferences. One thing that I am not happy with, is that it allows raw editing of the conf files. Gawd help us if a user gets into that lot. I emailed Third lane, and they replied staright away with an address where I could download an evaluation. I'd publish the url here, but there must be a reason why they don't show it on their web site. Oh, and by the way (this from a beginner), I found it by searching on the WIKI Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Manager
Not yet. It's under development. Greg Alex Brecher wrote: Is there anything open source out there that has the same or better feature set than Asterisk PBX Manager ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter Sent: Tuesday, November 30, 2004 4:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Manager I have this, it comes as a webmin module. I also got it with the intention of bundling it for clients. It costs $300 , and the license is tied to the NIC. While it wont do EVERYTHING, it will probably be sufficient for the user to set up extensions/phones/menus/voicemail/conferences. One thing that I am not happy with, is that it allows raw editing of the conf files. Gawd help us if a user gets into that lot. I emailed Third lane, and they replied staright away with an address where I could download an evaluation. I'd publish the url here, but there must be a reason why they don't show it on their web site. Oh, and by the way (this from a beginner), I found it by searching on the WIKI Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX Manager
Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan management and assignment of scripts to extensions Voice Menu Manager - configuration of Auto Attendant and multi level voice menus Script Manager - creation of scripts for call handling (used by Extensions Manager) Conference Manager - configuration of conference rooms Configuration Editor - direct access to Asterisk configuration files Command Shell- web interface to Asterisk command line interface File Manager - intelligent upload and download for various configuration and support files ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX Manager
I've heard it mentioned but I've never seen where to download it, and if it isn't gpl'd then how much they want for it. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Bentley Sent: Tuesday, November 30, 2004 2:33 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Manager Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan management and assignment of scripts to extensions Voice Menu Manager - configuration of Auto Attendant and multi level voice menus Script Manager - creation of scripts for call handling (used by Extensions Manager) Conference Manager - configuration of conference rooms Configuration Editor - direct access to Asterisk configuration files Command Shell- web interface to Asterisk command line interface File Manager - intelligent upload and download for various configuration and support files ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Manager
I have tried to email them multiple times to get more information, but they have not responded to my requests. I do know that it is NOT GPLed and it is commercial software.I have emailed ThirdLane to ask about a demo and costs, as I am looking for something to bundle for my customers to use. Robert Darren Bentley wrote: Hi, I haven't seen any mention of this on the list. I'm curious if anyone has tried it and can share some opinions on it? http://www.thirdlane.com/screenshots.htm http://www.thirdlane.com/opensource.htm#manager Defaults Manager - initial PBX configuration Device Manager - management of devices (phones) Mailbox Manager - configuration of user mailboxes Extensions Manager - dialplan management and assignment of scripts to extensions Voice Menu Manager - configuration of Auto Attendant and multi level voice menus Script Manager - creation of scripts for call handling (used by Extensions Manager) Conference Manager - configuration of conference rooms Configuration Editor - direct access to Asterisk configuration files Command Shell- web interface to Asterisk command line interface File Manager - intelligent upload and download for various configuration and support files ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX Manager
I have this, it comes as a webmin module. I also got it with the intention of bundling it for clients. It costs $300 , and the license is tied to the NIC. While it wont do EVERYTHING, it will probably be sufficient for the user to set up extensions/phones/menus/voicemail/conferences. One thing that I am not happy with, is that it allows raw editing of the conf files. Gawd help us if a user gets into that lot. I emailed Third lane, and they replied staright away with an address where I could download an evaluation. I'd publish the url here, but there must be a reason why they don't show it on their web site. Oh, and by the way (this from a beginner), I found it by searching on the WIKI Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX and backup Circuits
I am interested to know how one would calculate the amount of PSTN connection needed for backup on an Asterisk PBX that is being setup to receive its DIDs via a VoIP provide. To sum up what I am implementing: I am porting my DIDs to a VoIP provide so I will need a back up plan in place if the Data network fails. In addition, 911 will always be going out the PSTN so I know I need at least one POTs circuit. Calls inbound and outbound will always routed through the data network. In telco terms, you probably need to do a small Traffic Study; analyze the existing traffic for maximum number of simultanous calls, etc. If this is an existing business with an existing pbx, there are likely some usage statistics available within the pbx. If that's not available, some telephone companies will do the traffic study for you (don't need to tell them why your doing it, but rather to determine the number of telco lines needed for the business.) If that's not possible, ask the telco to provide you with a list of all calls with detail and run through the list to calculate the maximum number of simultanous calls. If this is a new installation with absolutely no history, your only option is to guess at the maximum. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX and backup Circuits
I am interested to know how one would calculate the amount of PSTN connection needed for backup on an Asterisk PBX that is being setup to receive its DIDs via a VoIP provide. To sum up what I am implementing: I am porting my DIDs to a VoIP provide so I will need a back up plan in place if the Data network fails. In addition, 911 will always be going out the PSTN so I know I need at least one POTs circuit. Calls inbound and outbound will always routed through the data network. Thanks, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX and backup Circuits
I am interested to know how one would calculate the amount of PSTN connection needed for backup on an Asterisk PBX that is being setup to receive its DIDs via a VoIP provide. To sum up what I am implementing: I am porting my DIDs to a VoIP provide so I will need a back up plan in place if the Data network fails. In addition, 911 will always be going out the PSTN so I know I need at least one POTs circuit. Calls inbound and outbound will always routed through the data network. Thanks, Kurt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi James, This is a feature that needs to be enabled on both the phones and on Asterisk. So after enabling on your BT100 you need to add 'cancallforward=yes' to each extension in sip.conf you would like to add this feature to as in :- [9500] context=internal type=friend username=9500 host=dynamic callerid=9500 disallow=all allow=ulaw allow=alaw dtmfmode=info mailbox=9500 callgroup=1 pickupgroup=1 cancallforward=yes Craig - Original Message - From: James Freire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 21, 2004 12:09 AM Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
- Original Message -
From: James Freire [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, August 20, 2004 9:09 AM
Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi All,
I am using a Grandstream BT100 and I have been trying to get the PBX
features to work for DND, call foward, etc. These functions do work when I
use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
functions (ie *78, *79) to work using my SIP phones. Is there a feature that
has to be enabled to do this? I know these functions are available within
the GS phone but all of them seem to just show the phone as being busy, even
though, say, call foward is supposed to foward. It just makes the phone
busy. I figure it would be easier just to have asterisk handling all those
PBX functions.
Thanks,
James
Someone correct me if I'm wrong but I believe you'll need the dialplan for
this one...
What I envision is doing something like this...
[verticalservice]
exten = *78,1,DbGet(${dnd}=features/dnd)
exten = *78,2,DbPut(features/dnd=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
exten = *78,102,GotoIf($[${dnd} = '0')]?103:104)
exteh = *78,103,DbPut(features/dnd=1)
exten = *78,104,Playback(pbx-dndenabled)
exten = *78,105,Hangup()
exten = *79 ... etc...
Then in your extension calling macro, you're going to want to check against
the DB like this...
[macro-insidedial]
exten = s,1,DbGet(${dnd}=features/dnd)
exten = s,2,DbGet(${fw}=features/fw)
exten = s,3,Dial(${ARG1},25,tT)
exten = s,4,VoiceMail(u${ARG1})
exten = s,5,Hangup()
exten = s,102,GotoIf($[${dnd} = '1']?200:2)
exten = s,103,GotoIf($[${fw} = '1']?300:3)
exten = s,104,VoiceMail(b${ARG1})
exten = s,200,VoiceMail(b${ARG1})
exten = s,201,Hangup()
exten = s,300,Dial(SIP/[EMAIL PROTECTED],60)
exten = s,301,Congestion()
be sure to include [verticalservice] in your inside-office context...
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RE: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi Craig, Thank you very much for the helpful information. I did enable that setting and it seems to have worked but not all the way. I do a *72 for an unconditional call forward + the number to forward to. Then when I dial the grandstream that has it enabled, asterisk just reponds that the extension is busy, the BT does not foward the call. I also get the following on the CLI -- Executing Dial(Zap/8-1, SIP/2000|20) in new stack -- Called 2000 -- Got SIP response 302 Moved Temporarily back from 64.201.13.50 -- SIP/2000-42e8 is busy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Craig Guy Sent: Friday, August 20, 2004 12:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi James, This is a feature that needs to be enabled on both the phones and on Asterisk. So after enabling on your BT100 you need to add 'cancallforward=yes' to each extension in sip.conf you would like to add this feature to as in :- [9500] context=internal type=friend username=9500 host=dynamic callerid=9500 disallow=all allow=ulaw allow=alaw dtmfmode=info mailbox=9500 callgroup=1 pickupgroup=1 cancallforward=yes Craig - Original Message - From: James Freire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 21, 2004 12:09 AM Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
On Fri, Aug 20, 2004 at 10:13:16AM -0700, Chris Shaw said:
- Original Message -
From: James Freire [EMAIL PROTECTED]
I am using a Grandstream BT100 and I have been trying to get the PBX
features to work for DND, call foward, etc. These functions do work when I
use my POTS phones hooked up to my Zap cards. But I cannot get the PBX
functions (ie *78, *79) to work using my SIP phones. Is there a feature that
has to be enabled to do this? I know these functions are available within
the GS phone but all of them seem to just show the phone as being busy, even
though, say, call foward is supposed to foward. It just makes the phone
busy. I figure it would be easier just to have asterisk handling all those
PBX functions.
James
Someone correct me if I'm wrong but I believe you'll need the dialplan for
this one...
What I envision is doing something like this...
[verticalservice]
exten = *78,1,DbGet(${dnd}=features/dnd)
exten = *78,2,DbPut(features/dnd=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
exten = *78,102,GotoIf($[${dnd} = '0')]?103:104)
exteh = *78,103,DbPut(features/dnd=1)
exten = *78,104,Playback(pbx-dndenabled)
exten = *78,105,Hangup()
exten = *79 ... etc...
Wouldn't you need to track each extension? something like:
exten = *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
exten = *78,2,DbPut(dnd/${CALLERIDNUM}=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
etc.?
The wiki has an exmple for call forwarding:
http://www.voip-info.org/wiki-Asterisk+call+forwarding
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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Wouldn't you need to track each extension? something like:
exten = *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
exten = *78,2,DbPut(dnd/${CALLERIDNUM}=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
etc.?
Yep! good catch! that's why I asked someone to correct me, I was in a hurry
and this was an on-the-fly kind of example...
You would need to do something like this, or make a key like
features/dnd-${CALLERIDNUM} would be best... Would also work for
forwarding...
-Chris
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RE: [Asterisk-Users] Asterisk PBX Functions via SIP phone
I am suprised that one would have to create a dialplan since its an already built in
function that works with regular POTS phones. Or is it because of the way DTMF is sent
via SIP?
Someone correct me if I'm wrong but I believe you'll need the dialplan for
this one...
What I envision is doing something like this...
[verticalservice]
exten = *78,1,DbGet(${dnd}=features/dnd)
exten = *78,2,DbPut(features/dnd=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
exten = *78,102,GotoIf($[${dnd} = '0')]?103:104)
exteh = *78,103,DbPut(features/dnd=1)
exten = *78,104,Playback(pbx-dndenabled)
exten = *78,105,Hangup()
exten = *79 ... etc...
Wouldn't you need to track each extension? something like:
exten = *78,1,DbGet(${dnd}=dnd/${CALLERIDNUM})
exten = *78,2,DbPut(dnd/${CALLERIDNUM}=1)
exten = *78,3,Playback(pbx-dndenabled)
exten = *78,4,Hangup()
etc.?
The wiki has an exmple for call forwarding:
http://www.voip-info.org/wiki-Asterisk+call+forwarding
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Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
I am suprised that one would have to create a dialplan since its an already built in function that works with regular POTS phones. Or is it because of the way DTMF is sent via SIP? I don't know digium's long range plans, but looking through chan_sip.c NONE of the vertical service codes are mentioned anywhere... A quick look through chan_zap reveals all of them... So for right now it's not implemented in SIP... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Chris Shaw wrote: I am suprised that one would have to create a dialplan since its an already built in function that works with regular POTS phones. Or is it because of the way DTMF is sent via SIP? I don't know digium's long range plans, but looking through chan_sip.c NONE of the vertical service codes are mentioned anywhere... A quick look through chan_zap reveals all of them... So for right now it's not implemented in SIP... Well, here we stumble over the SIP religion again. First, a phone connected to an RJ11 jack in a Digium card is a stupid phone. All the intelligence lies in the zaptel driver and asterisk. Most SIP phones are more clever (at least expected to be much more clever than the GS :-). Look at the SIPURA, where you are able to implement vertical service codes in the SIPura. Asterisk should not bother with DND and forwards, the SIP phone does. Just send the call to the phone. Some of these phones are complete Linux systems with IPsec, multiple lines and a lot of routing intelligence. There's also a discussion between Asterisk developers on whether these codes should be fixed in the channel or in the dial plan. At least, they should be configurable since there's no global standard (again). Or there may be, but there are still differences between countries and providers... * Executive summary: SIP is designed for very intelligent end-points. * A PBX with analogue lines is designed for central intelligence. * Asterisk will always be in the middle of these kind of discussions, and it'll be fun each time we try to sort it out. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
- Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 1:37 PM Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone Chris Shaw wrote: I am suprised that one would have to create a dialplan since its an already built in function that works with regular POTS phones. Or is it because of the way DTMF is sent via SIP? I don't know digium's long range plans, but looking through chan_sip.c NONE of the vertical service codes are mentioned anywhere... A quick look through chan_zap reveals all of them... So for right now it's not implemented in SIP... Well, here we stumble over the SIP religion again. First, a phone connected to an RJ11 jack in a Digium card is a stupid phone. All the intelligence lies in the zaptel driver and asterisk. Most SIP phones are more clever (at least expected to be much more clever than the GS :-). Look at the SIPURA, where you are able to implement vertical service codes in the SIPura. Asterisk should not bother with DND and forwards, the SIP phone does. Just send the call to the phone. Some of these phones are complete Linux systems with IPsec, multiple lines and a lot of routing intelligence. There's also a discussion between Asterisk developers on whether these codes should be fixed in the channel or in the dial plan. At least, they should be configurable since there's no global standard (again). Or there may be, but there are still differences between countries and providers... * Executive summary: SIP is designed for very intelligent end-points. * A PBX with analogue lines is designed for central intelligence. * Asterisk will always be in the middle of these kind of discussions, and it'll be fun each time we try to sort it out. /Olle No, I agree completely with the way it works now, in fact I think it SHOULDN'T be implemented in SIP myself... Doing it in the dialplan (if your phone doesn't support it) works fine and doesn't break anything (that's the key right there). We need some more docs on how to do different things and I'm sure many people could contribute those, myself included... Some already have... The only thing is, if any of the apps you've written in your dialplan become obsoleted or change syntax, your whole implementation will get screwed over... I guess that's true with anything though... -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hi, sorry for interruption, but are there any guides for all possible Asterisk PBX functions that are available with no particular dialplan handling ? Thanks, Robert. - Original Message - From: James Freire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 6:09 PM Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
http://www.voip-info.org/wiki-asterisk+pbx+functions http://www.voip-info.org/wiki-asterisk+vertical+service+activation+codes - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 3:02 PM Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi, sorry for interruption, but are there any guides for all possible Asterisk PBX functions that are available with no particular dialplan handling ? Thanks, Robert. - Original Message - From: James Freire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, August 20, 2004 6:09 PM Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Hmm, Asterisk is certainly recognising the call forward. I had this error when I first tried it and resolved by adding the 'cancallforward=yes' in the sip.conf for that extension. I think you may actually have to restart asterisk to enable these functions rather than reload? I am also running the August 8 CVS head rather than the 'stable' version of Asterisk and my grandstream firmware is 1.0.5.10. Craig - Original Message - From: James Freire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 21, 2004 1:40 AM Subject: RE: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi Craig, Thank you very much for the helpful information. I did enable that setting and it seems to have worked but not all the way. I do a *72 for an unconditional call forward + the number to forward to. Then when I dial the grandstream that has it enabled, asterisk just reponds that the extension is busy, the BT does not foward the call. I also get the following on the CLI -- Executing Dial(Zap/8-1, SIP/2000|20) in new stack -- Called 2000 -- Got SIP response 302 Moved Temporarily back from 64.201.13.50 -- SIP/2000-42e8 is busy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Craig Guy Sent: Friday, August 20, 2004 12:36 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi James, This is a feature that needs to be enabled on both the phones and on Asterisk. So after enabling on your BT100 you need to add 'cancallforward=yes' to each extension in sip.conf you would like to add this feature to as in :- [9500] context=internal type=friend username=9500 host=dynamic callerid=9500 disallow=all allow=ulaw allow=alaw dtmfmode=info mailbox=9500 callgroup=1 pickupgroup=1 cancallforward=yes Craig - Original Message - From: James Freire [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, August 21, 2004 12:09 AM Subject: [Asterisk-Users] Asterisk PBX Functions via SIP phone Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of them seem to just show the phone as being busy, even though, say, call foward is supposed to foward. It just makes the phone busy. I figure it would be easier just to have asterisk handling all those PBX functions. Thanks, James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Functions via SIP phone
Craig Guy wrote: cancallforward=yes There is no such function in distributed chan_sip.c, ergo there can't be such a configuration parameter. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, PBX, VoIP and PRI
On Wed, 28 Jul 2004, Chris Johnson wrote: On Wed, 28 Jul 2004, Chris Johnson wrote: Why not plug the PRI into a TE410P in the asterisk box and handle both the ppp and the voice via asterisk? If it'll work, sounds great!! Anyone doing this? Sorry, there are no analog softmodem drivers yet. I have been living in isdn land for too long and forgot about the analog modems. There are indeed drivers for ppp over isdn direct but no softmodem for analog calls. We have solved this by routing those calls to our old pbx (over an E1 PRI) from which a couple of BRI:s go to isdn modems with analog capabilities. So, you should be able to hook up your old PRI equipment to Asterisk, i.e. PSTN -PRI- Asterisk -PRI- Old_equipment \ ---lan--- sip stuff This is sort of what we are doing. As far as I can tell the digital channels are passed transparently from one pri to the other once a call is set up. We can do both analog termination and direct isdn connections. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk, PBX, VoIP and PRI
Hi everyone, Im new to the list, so I apologize if this is a trite question. Were an ISP with a barely used PRI, and also currently use asterisk with POTS service on an audiocodes gateway. We have a notion that we want to take on a new revenue stream, with our model being as such: 1) Convert one of our PRIs into a two way voice/data bundle. 2) Plug this PRI into a piece of equipment (AS5300?) which can terminate both PPP, *AND* feed our Asterisk system. Wed transfer our POTS service over to the PRI and pick up the SIP signaling there. 3) In the extended model, use this setup to resell VoIP to our business ISP clients. Has anyone out there done this? If so, is the AS5300 the right choice? What caveats will I face?
Re: [Asterisk-Users] Asterisk, PBX, VoIP and PRI
On Wed, 28 Jul 2004, Chris Johnson wrote: 1) Convert one of our PRIs into a two way voice/data bundle. 2) Plug this PRI into a piece of equipment (AS5300?) which can terminate both PPP, *AND* feed our Asterisk system. We'd transfer our POTS service over to the PRI and pick up the SIP signaling there. 3) In the extended model, use this setup to resell VoIP to our business ISP clients. Has anyone out there done this? If so, is the AS5300 the right choice? What caveats will I face? Why not plug the PRI into a TE410P in the asterisk box and handle both the ppp and the voice via asterisk? I am not sure what you mean by Convert one of our PRIs into a two way voice/data bundle. Isn't that sort of what a PRI is anyway? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, PBX, VoIP and PRI
This is similar what we are looking to do with our customers, ISP providing VoIP withour * system to broadband customers. Having trouble getting the Mysql to work on RedHat 9.0 atm for the billing. but would be interested in talkingmore and sharingexperiences. Hi everyone, Im new to the list, so I apologize if this is a trite question. Were an ISP with a barely used PRI, and also currently use asterisk with POTS service on an audiocodes gateway. We have a notion that we want to take on a new revenue stream, with our model being as such: 1) Convert one of our PRIs into a two way voice/data bundle. 2) Plug this PRI into a piece of equipment (AS5300?) which can terminate both PPP, *AND* feed our Asterisk system. Wed transfer our POTS service over to the PRI and pick up the SIP signaling there. 3) In the extended model, use this setup to resell VoIP to our business ISP clients. Has anyone out there done this? If so, is the AS5300 the right choice? What caveats will I face?
RE: [Asterisk-Users] Asterisk, PBX, VoIP and PRI
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Wednesday, July 28, 2004 5:07 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk, PBX, VoIP and PRI On Wed, 28 Jul 2004, Chris Johnson wrote: 1) Convert one of our PRIs into a two way voice/data bundle. 2) Plug this PRI into a piece of equipment (AS5300?) which can terminate both PPP, *AND* feed our Asterisk system. We'd transfer our POTS service over to the PRI and pick up the SIP signaling there. 3) In the extended model, use this setup to resell VoIP to our business ISP clients. Has anyone out there done this? If so, is the AS5300 the right choice? What caveats will I face? Why not plug the PRI into a TE410P in the asterisk box and handle both the ppp and the voice via asterisk? If it'll work, sounds great!! Anyone doing this? I am not sure what you mean by Convert one of our PRIs into a two way voice/data bundle. Isn't that sort of what a PRI is anyway? Yes, but most ISPs use inward only PRI. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, PBX, VoIP and PRI
On Jul 28, 2004, at 2:07 PM, Peter Svensson wrote: On Wed, 28 Jul 2004, Chris Johnson wrote: 1) Convert one of our PRIs into a two way voice/data bundle. 2) Plug this PRI into a piece of equipment (AS5300?) which can terminate both PPP, *AND* feed our Asterisk system. We'd transfer our POTS service over to the PRI and pick up the SIP signaling there. 3) In the extended model, use this setup to resell VoIP to our business ISP clients. Has anyone out there done this? If so, is the AS5300 the right choice? What caveats will I face? Why not plug the PRI into a TE410P in the asterisk box and handle both the ppp and the voice via asterisk? He's (presumably) talking about PPP over a 56k modem, not PPP to a ISDN BRI. Asterisk can't do an internal 56k modem, at least not yet. I thought that someone said that you need the AS5350 to do combined voice and data, but I might be mistaken. It's been years since I've dealt with dialup. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, PBX, VoIP and PRI
Can the TE410P handle the ppp ( analog 56K inboud modem calls ( be a modem pool server )) without any extra hardware? At 04:07 PM 07/28/2004, you wrote: On Wed, 28 Jul 2004, Chris Johnson wrote: 1) Convert one of our PRIs into a two way voice/data bundle. 2) Plug this PRI into a piece of equipment (AS5300?) which can terminate both PPP, *AND* feed our Asterisk system. We'd transfer our POTS service over to the PRI and pick up the SIP signaling there. 3) In the extended model, use this setup to resell VoIP to our business ISP clients. Has anyone out there done this? If so, is the AS5300 the right choice? What caveats will I face? Why not plug the PRI into a TE410P in the asterisk box and handle both the ppp and the voice via asterisk? I am not sure what you mean by Convert one of our PRIs into a two way voice/data bundle. Isn't that sort of what a PRI is anyway? Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Jim Lund Fox Business Systems KansasNet [EMAIL PROTECTED] 531 Ft. Riley Blvd (785)776-1452Manhattan, KS 66502 http://www.kansas.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb
That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron J. Angel Sent: Wednesday, 23 June 2004 10:33 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb john lawler [EMAIL PROTECTED] wrote: You don't have to put this in the dialplan. It's one of those low-level functions in Asterisk (possibly controlled at the driver level-- I'm not sure about that). If you have an extension defined, pick up the handset and dial '*78', you should see on the Asterisk CLI: Enabled DND on channel whatever That assumes your using zaptel, no? This doesn't exist for other channels as it's not built in to the channel driver for anything but zaptel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk -- PBX Do Not Disturb
Yes this is where I wanted to use it the most, I use Asterisk as a software-only based PBX using SIP and IAX, I have an ATA device on my network connected to the PBX but it nor the actual analog phone has a DND function. I am hoping to implement an Asterisk-side based DND somehow but was wondering where to go from there. Thanks Steve Dean Collins wrote: That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron J. Angel Sent: Wednesday, 23 June 2004 10:33 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb john lawler [EMAIL PROTECTED] wrote: You don't have to put this in the dialplan. It's one of those low-level functions in Asterisk (possibly controlled at the driver level-- I'm not sure about that). If you have an extension defined, pick up the handset and dial '*78', you should see on the Asterisk CLI: Enabled DND on channel whatever That assumes your using zaptel, no? This doesn't exist for other channels as it's not built in to the channel driver for anything but zaptel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Stephen Rosebush, [EMAIL PROTECTED] http://www.desynched.org/ // PSTN USA:1-248-724-4452 x201 Netherlands:+31-(0)20-6598858 x63420 x201 // IP Phone FWD:63420 x201 IAXTEL: 1-700-356-6191 x201 SIP:sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk -- PBX Do Not Disturb
Interestingly enough, I just tried this on a SIP extension and it worked fine...although I did not see a report on the command line as expected. I used a Polycom IP600 to call an extension which is an analog phone plugged into one side of a Spiura SPA-2000. Keying *78 into the phone one the SPA generated a pulsing dialtone for 3 seconds then the dialtone returned to normal. I hungup then dialed that extension again and was routed directly to voicemail. I do have an X100p in my * server and zaptel is loaded. However, I don't use it as my fxo anymore, I just left it there a timing source for the conferences. Michael On Wed, 23 Jun 2004 10:08:46 +0200, Stephen Rosebush wrote: Yes this is where I wanted to use it the most, I use Asterisk as a software-only based PBX using SIP and IAX, I have an ATA device on my network connected to the PBX but it nor the actual analog phone has a DND function. I am hoping to implement an Asterisk-side based DND somehow but was wondering where to go from there. Thanks Steve Dean Collins wrote: That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron J. Angel Sent: Wednesday, 23 June 2004 10:33 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb john lawler [EMAIL PROTECTED] wrote: You don't have to put this in the dialplan. It's one of those low-level functions in Asterisk (possibly controlled at the driver level-- I'm not sure about that). If you have an extension defined, pick up the handset and dial '*78', you should see on the Asterisk CLI: Enabled DND on channel whatever That assumes your using zaptel, no? This doesn't exist for other channels as it's not built in to the channel driver for anything but zaptel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Stephen Rosebush, [EMAIL PROTECTED] http://www.desynched.org/ // PSTN USA: 1-248-724-4452 x201 Netherlands: +31-(0)20-6598858 x63420 x201 // IP Phone FWD: 63420 x201 IAXTEL:1-700-356-6191 x201 SIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 One day in your life shouldn't be a problem - 54-40 from One Day ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk -- PBX Do Not Disturb
I quote from the wiki on voip-info: In Asterisk, DND is controlled by dialing: *78 to turn *on* Do Not Disturb mode and *79 to turn *off* Do Not Disturb mode How would one implement this?? I've seen dialplans that created these extensions and you would need to use the Asterisk DB to set the DND flag on or off.. I am still pretty new and I would like to know how to implement this and maybe some other 'star' services? Thanks a bunch guys! -- Stephen Rosebush, [EMAIL PROTECTED] http://www.desynched.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk -- PBX Do Not Disturb
You don't have to put this in the dialplan. It's one of those low-level functions in Asterisk (possibly controlled at the driver level-- I'm not sure about that). If you have an extension defined, pick up the handset and dial '*78', you should see on the Asterisk CLI: Enabled DND on channel whatever And then if someone tries to call that channel, they'd get handled as if the person had the phone off the hook, even if it weren't. It's that simple. To turn it back off, you use '*79'. jl p.s. I think I actually added that info that you quoted from voip-info. If you think it's unclear, grab an account and reword it! Stephen Rosebush wrote: I quote from the wiki on voip-info: In Asterisk, DND is controlled by dialing: *78 to turn *on* Do Not Disturb mode and *79 to turn *off* Do Not Disturb mode How would one implement this?? I've seen dialplans that created these extensions and you would need to use the Asterisk DB to set the DND flag on or off.. I am still pretty new and I would like to know how to implement this and maybe some other 'star' services? Thanks a bunch guys! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb
john lawler [EMAIL PROTECTED] wrote: You don't have to put this in the dialplan. It's one of those low-level functions in Asterisk (possibly controlled at the driver level-- I'm not sure about that). If you have an extension defined, pick up the handset and dial '*78', you should see on the Asterisk CLI: Enabled DND on channel whatever That assumes your using zaptel, no? This doesn't exist for other channels as it's not built in to the channel driver for anything but zaptel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX - RT Integration
Greetings, I had been working on Asterisk (http://asteriskpbx.org) about 2 years ago . http://www.marko.net/asterisk/archives/0210/0107.html Last night with the help of Jesse's rt-soap-server.pl (and some prodding) I implemented a much cleaner, more repeatable * - RT phone gateway with some notes: http://megaglobal.net/docs/asterisk/html/rtasterisk.html Questions or comments appreciated. --Michael. -- . Michael Jastremski .. .. Network Engineer Megaglobal Networks Megaglobal.net .. Photographer Open Photo Project Openphoto.net . .. Resident West PhiladelphiaWestphila.net . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX Billing
Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk PBX Billing
ok, I'll bite :-) What the heck is a phone shop system ?? On Wed, 02 Jul 2003 09:48:44 +, shepherd fungayi wrote: Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX Billing
I'm very interested in the same thing for a hotel system I would like to implement. Anyone know if the country codes be tied to a pricing lookup table? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 5:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk PBX Billing
Shepherd- Having designed one of these in the past (in a higher level voice environment), I can tell you that this is not a small undertaking. It's at least as much an SQL job as a voice task. Usually the way to accomplish this is to establish more-or-less a pre-paid phone card system, where the shop prepays an overall amount for international calling access. Then you have to time each call as it is occurring, debiting each account, and the master account, in real-time. This can be a bit complex when you have 20 or 30 calls going at one time. You have to cut them off promptly when the money runs out (big problem). And you have to provide call detail and charges to them at the end of each call, using their own retail tariff. To add to the complexity, each country has a different tariff from the long distance carrier, and within the country, major cities often have special rates per minute. Mobiles have a different tariff too. Phone card platforms usually include a least-cost routing system which chooses a carrier real time based on the call. Tariffs change weekly and must be updated in the system. Anyway, I'm just scratching the surface! I'll write more when I can! Cheers Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] Asterisk PBX Billing
i think that the problem could be something more easy: it is possible inside asterisk to log all che calls of all the users and know the timing and the number called for each call? if it is possible to do that, could be possible to make a program that takes this files and generate the costs reading the log informations... so for me the real question is: there is a log of all the phone call that are made by asterisk? Angelo Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. It's at SS least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a pre-paid SS phone card system, where the shop prepays an overall amount for SS international calling access. Then you have to time each call as it is SS occurring, debiting each account, and the master account, in real-time. This SS can be a bit complex when you have 20 or 30 calls going at one time. You SS have to cut them off promptly when the money runs out (big problem). And SS you have to provide call detail and charges to them at the end of each call, SS using their own retail tariff. SS To add to the complexity, each country has a different tariff from the long SS distance carrier, and within the country, major cities often have special SS rates per minute. Mobiles have a different tariff too. Phone card SS platforms usually include a least-cost routing system which chooses a SS carrier real time based on the call. Tariffs change weekly and must be SS updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics, plasticity, and form. The greatest scientists are always artists as well. Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re[2]: [Asterisk-Users] Asterisk PBX Billing
That's all I would need, it would be easy enough to work out the cost after that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angelo Sampietro Sent: Wednesday, July 02, 2003 10:06 AM To: Scott Stingel Cc: [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing i think that the problem could be something more easy: it is possible inside asterisk to log all che calls of all the users and know the timing and the number called for each call? if it is possible to do that, could be possible to make a program that takes this files and generate the costs reading the log informations... so for me the real question is: there is a log of all the phone call that are made by asterisk? Angelo Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. SS It's at least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a SS pre-paid phone card system, where the shop prepays an overall amount SS for international calling access. Then you have to time each call SS as it is occurring, debiting each account, and the master account, SS in real-time. This can be a bit complex when you have 20 or 30 calls SS going at one time. You have to cut them off promptly when the money SS runs out (big problem). And you have to provide call detail and SS charges to them at the end of each call, using their own retail SS tariff. SS To add to the complexity, each country has a different tariff from SS the long distance carrier, and within the country, major cities SS often have special rates per minute. Mobiles have a different SS tariff too. Phone card platforms usually include a least-cost SS routing system which chooses a carrier real time based on the call. SS Tariffs change weekly and must be updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I SS can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics, plasticity, and form. The greatest scientists are always artists as well. Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re[2]: [Asterisk-Users] Asterisk PBX Billing
There is a CDR (Call Detail Record) which is accessible in two different ways. The first is via a simple comma delimited file which can be parsed and fed into whatever database that you want. The second way is to dump the CDR directly into MySQL, and extract accordingly. So the only trick there is to create a database for billing and create a relationship that will extract from the CDR database. Kim C. Callis -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Angelo Sampietro Sent: Wednesday, July 02, 2003 7:06 AM To: Scott Stingel Cc: [EMAIL PROTECTED] Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing i think that the problem could be something more easy: it is possible inside asterisk to log all che calls of all the users and know the timing and the number called for each call? if it is possible to do that, could be possible to make a program that takes this files and generate the costs reading the log informations... so for me the real question is: there is a log of all the phone call that are made by asterisk? Angelo Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. It's at SS least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a pre-paid SS phone card system, where the shop prepays an overall amount for SS international calling access. Then you have to time each call as it is SS occurring, debiting each account, and the master account, in real-time. This SS can be a bit complex when you have 20 or 30 calls going at one time. You SS have to cut them off promptly when the money runs out (big problem). And SS you have to provide call detail and charges to them at the end of each call, SS using their own retail tariff. SS To add to the complexity, each country has a different tariff from the long SS distance carrier, and within the country, major cities often have special SS rates per minute. Mobiles have a different tariff too. Phone card SS platforms usually include a least-cost routing system which chooses a SS carrier real time based on the call. Tariffs change weekly and must be SS updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics, plasticity, and form. The greatest scientists are always artists as well. Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] Asterisk PBX Billing
thanks a lot! can you tell me where can i find more info about the CDR? probably this will be the better way to give to the company a summary with all the phone traffic :) Angelo Thursday, July 3, 2003, 4:37:32 PM, you wrote: KCC There is a CDR (Call Detail Record) which is accessible in two different KCC ways. The first is via a simple comma delimited file which can be parsed KCC and fed into whatever database that you want. The second way is to dump KCC the CDR directly into MySQL, and extract accordingly. So the only trick KCC there is to create a database for billing and create a relationship that KCC will extract from the CDR database. KCC Kim C. Callis KCC -Original Message- KCC From: [EMAIL PROTECTED] KCC [mailto:[EMAIL PROTECTED] On Behalf Of Angelo KCC Sampietro KCC Sent: Wednesday, July 02, 2003 7:06 AM KCC To: Scott Stingel KCC Cc: [EMAIL PROTECTED] KCC Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing KCC i think that the problem could be something more easy: KCC it is possible inside asterisk to log all che calls of all the users KCC and know the timing and the number called for each call? KCC if it is possible to do that, could be possible to make a program KCC that takes this files and generate the costs reading the log KCC informations... KCC so for me the real question is: there is a log of all the phone call KCC that are made by asterisk? KCC Angelo KCC Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. KCC It's at SS least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a KCC pre-paid SS phone card system, where the shop prepays an overall amount for SS international calling access. Then you have to time each call as it KCC is SS occurring, debiting each account, and the master account, in KCC real-time. This SS can be a bit complex when you have 20 or 30 calls going at one time. KCC You SS have to cut them off promptly when the money runs out (big problem). KCC And SS you have to provide call detail and charges to them at the end of KCC each call, SS using their own retail tariff. SS To add to the complexity, each country has a different tariff from KCC the long SS distance carrier, and within the country, major cities often have KCC special SS rates per minute. Mobiles have a different tariff too. Phone card SS platforms usually include a least-cost routing system which chooses KCC a SS carrier real time based on the call. Tariffs change weekly and must KCC be SS updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I KCC can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics, plasticity, and form. The greatest scientists are always artists as well. Albert Einstein ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re[4]: [Asterisk-Users] Asterisk PBX Billing
The mysql schema is available in the doc/cdr_mysql.txt file (from the asterisk source dir) James On Thu, 3 Jul 2003, Kim C. Callis wrote: You can find the comma delimited file at /var/log/asterisk/cdr-csv or if you are looking to do some easy querying on a database, you need to create a schema that I am sure someone on the channel has defined somewhere. At that point you clean up the /etc/asterisk/cdr_mysql.conf file to point to the appropriate database and authentication information. Kim C. Callis -Original Message- From: Angelo Sampietro [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 02, 2003 8:07 AM To: Kim C. Callis Cc: [EMAIL PROTECTED] Subject: Re[4]: [Asterisk-Users] Asterisk PBX Billing thanks a lot! can you tell me where can i find more info about the CDR? probably this will be the better way to give to the company a summary with all the phone traffic :) Angelo Thursday, July 3, 2003, 4:37:32 PM, you wrote: KCC There is a CDR (Call Detail Record) which is accessible in two different KCC ways. The first is via a simple comma delimited file which can be parsed KCC and fed into whatever database that you want. The second way is to dump KCC the CDR directly into MySQL, and extract accordingly. So the only trick KCC there is to create a database for billing and create a relationship that KCC will extract from the CDR database. KCC Kim C. Callis KCC -Original Message- KCC From: [EMAIL PROTECTED] KCC [mailto:[EMAIL PROTECTED] On Behalf Of Angelo KCC Sampietro KCC Sent: Wednesday, July 02, 2003 7:06 AM KCC To: Scott Stingel KCC Cc: [EMAIL PROTECTED] KCC Subject: Re[2]: [Asterisk-Users] Asterisk PBX Billing KCC i think that the problem could be something more easy: KCC it is possible inside asterisk to log all che calls of all the users KCC and know the timing and the number called for each call? KCC if it is possible to do that, could be possible to make a program KCC that takes this files and generate the costs reading the log KCC informations... KCC so for me the real question is: there is a log of all the phone call KCC that are made by asterisk? KCC Angelo KCC Wednesday, July 2, 2003, 3:28:22 PM, you wrote: SS Shepherd- SS Having designed one of these in the past (in a higher level voice SS environment), I can tell you that this is not a small undertaking. KCC It's at SS least as much an SQL job as a voice task. SS Usually the way to accomplish this is to establish more-or-less a KCC pre-paid SS phone card system, where the shop prepays an overall amount for SS international calling access. Then you have to time each call as it KCC is SS occurring, debiting each account, and the master account, in KCC real-time. This SS can be a bit complex when you have 20 or 30 calls going at one time. KCC You SS have to cut them off promptly when the money runs out (big problem). KCC And SS you have to provide call detail and charges to them at the end of KCC each call, SS using their own retail tariff. SS To add to the complexity, each country has a different tariff from KCC the long SS distance carrier, and within the country, major cities often have KCC special SS rates per minute. Mobiles have a different tariff too. Phone card SS platforms usually include a least-cost routing system which chooses KCC a SS carrier real time based on the call. Tariffs change weekly and must KCC be SS updated in the system. SS Anyway, I'm just scratching the surface! I'll write more when I KCC can! SS Cheers SS Scott Stingel SS Scott M. Stingel SS Emerging Voice Technology Inc. SS Email: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] SS URL:www.evtmedia.com http://www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of shepherd fungayi Sent: Wednesday, July 02, 2003 10:49 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk PBX Billing Hi I would like to use the Asterisk PBX as part of a phone shop system instead of the usual PBX plus PC. How can I do the the billing in a way that is convinient to the phone shop attendant? Regards Shepherd _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users SS ___ SS Asterisk-Users mailing list SS [EMAIL PROTECTED] SS http://lists.digium.com/mailman/listinfo/asterisk-users -- Angelo Sampietro IT Manager ARC Interactive After a certain high level of technical skill is achieved, Science and art tend to coalesce in esthetics
