[asterisk-users] migration
My client wants to use my service that I will host. It is an IVR application. I have the solution worked out on the server side. I will port his current POTS line phone number to a DID service where I can control it via SIP. Question relates to his current phones. Forgive me as I am new. Does he need his current phones? How will they ring if I port the number? Should I simply have him remove the phones and I can send the calls to his cell phones? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] migration
Re: [asterisk-users] migration Of geese or ducks? (A more descriptive subject will yield better replies.) On Sat, 27 Mar 2010, Thomas Perron wrote: > My client wants to use my service that I will host. It is an IVR > application. I have the solution worked out on the server side. I will > port his current POTS line phone number to a DID service where I can > control it via SIP. > > Question relates to his current phones. Forgive me as I am new. Does he > need his current phones? How will they ring if I port the number? > Should I simply have him remove the phones and I can send the calls to > his cell phones? If his "current POTS line number" rings his "current phones," and you port the POTS line number to a SIP provider, what would you expect to "ring" them? You could dial his cell phones or any other endpoint. You could connect the existing phones to your Asterisk server using ATAs or channel banks or ??? What is in the best interest of your client? There isn't enough information present to offer a reasonable opinion. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Migration from Mantis to JIRA
Greetings, A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be: https://issues.asterisk.org/jira/ Mantis will still be available for some time, but will be read-only. If you have an account on Mantis, you will be able to log in to JIRA using the same username. All of your history will have been migrated. This account can also be used on wiki.asterisk.org. IMPORTANT NOTE: You will have to click the "forgot my password" link to reset your password before you can log in, though. It is not possible to migrate passwords from one to the other as they use a different hashing algorithm. For more information about how to use JIRA, see the JIRA user's guide: http://confluence.atlassian.com/display/JIRA042/JIRA+User%27s+Guide If you run into any problems after the migration has taken place, please report them in the "JIRA Help" project. If you would rather report something via email, email espiceland at digium dot com and me. Thanks, [1] http://lists.digium.com/pipermail/asterisk-dev/2011-May/049088.html -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from Mantis to JIRA
We use Jira at work. I hate it. Hope you have a better experience than I've had! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Bryant Sent: Wednesday, June 01, 2011 7:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Migration from Mantis to JIRA Greetings, A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be: https://issues.asterisk.org/jira/ Mantis will still be available for some time, but will be read-only. If you have an account on Mantis, you will be able to log in to JIRA using the same username. All of your history will have been migrated. This account can also be used on wiki.asterisk.org. IMPORTANT NOTE: You will have to click the "forgot my password" link to reset your password before you can log in, though. It is not possible to migrate passwords from one to the other as they use a different hashing algorithm. For more information about how to use JIRA, see the JIRA user's guide: http://confluence.atlassian.com/display/JIRA042/JIRA+User%27s+Guide If you run into any problems after the migration has taken place, please report them in the "JIRA Help" project. If you would rather report something via email, email espiceland at digium dot com and me. Thanks, [1] http://lists.digium.com/pipermail/asterisk-dev/2011-May/049088.html -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from Mantis to JIRA
On 06/02/2011 06:46 AM, Terry Brummell wrote: > We use Jira at work. I hate it. Hope you have a better experience than > I've had! We've been using it for years internally to Digium. We've been happy with it. -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Migration from 1.2 to 1.8 in production
Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since asterisk 1.8 still in beta process. Thanks, S. Patel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Wednesday, November 03, 2010 9:14 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Migration from 1.2 to 1.8 in production Hello Everyone, We are running asterisk 1.2.x version in production environment since last 5 year and we have no issue at all, But now time to upgrade. and i heard about 1.8 which has introduce many features. I am wondering should I use asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? I would like if you suggest me which version would be good for production since asterisk 1.8 still in beta process. Thanks, S. Patel 1.8 will introduce many features and is the "supported standard", which will be important to you since you are on a "5 year upgrade" plan. It also has more "opportunities" than the 1.4 version since it is under active development and 1.4 is in a "patch only" state. If immediate stability is your goal, you may want to stick with 1.4. If I were going to bite the bullet on 1.6, I'd jump straight to 1.8 since there is no "end-of-life" advantage. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
Am 03.11.10 15:14, schrieb satish patel: > > Hello Everyone, > > We are running asterisk 1.2.x version in production environment since last 5 > year and we have no issue at all, But now time to upgrade. and i heard about > 1.8 which has introduce many features. I am wondering should I use asterisk > 1.8 in production ? or should I go with 1.4 or 1.6 stable version? > > I would like if you suggest me which version would be good for production > since asterisk 1.8 still in beta process. > > Thanks, > S. Patel > Hello Patel, it hardly depends on how many users and concurrent calls you have in your system cause i have recognized 1.2 can handle much more peers than 1.6 or 1.8. maybe you should try to setup a test server and first try it with your setup and some load tests if everything is working as you expect. best regards stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
Thanks for reply, I believe we have around 300 SIP phone register on asterisk and we have 2 T1 line. Roughly i would say max concurrent number 20/30 Max. My only concern is stability after whatever version migration. I believe 1.8 is new and it's just coming out form egg so quite worry about stability. So I have two choice 1.4 and 1.6 stable version. is there anyone who is using 1.8 in production? I am quite impressed with 1.8 features though Thanks, S. Patel > Date: Wed, 3 Nov 2010 15:42:21 +0100 > From: s...@sil.at > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Migration from 1.2 to 1.8 in production > > Am 03.11.10 15:14, schrieb satish patel: > > > > Hello Everyone, > > > > We are running asterisk 1.2.x version in production environment since last > > 5 year and we have no issue at all, But now time to upgrade. and i heard > > about 1.8 which has introduce many features. I am wondering should I use > > asterisk 1.8 in production ? or should I go with 1.4 or 1.6 stable version? > > > > I would like if you suggest me which version would be good for production > > since asterisk 1.8 still in beta process. > > > > Thanks, > > S. Patel > > > > Hello Patel, > > it hardly depends on how many users and concurrent calls you have in > your system cause i have recognized 1.2 can handle much more peers than > 1.6 or 1.8. > > maybe you should try to setup a test server and first try it with your > setup and some load tests if everything is working as you expect. > > best regards > > stefan > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
I have used 1.4 & 1.6. I am testing 1.8 for production and it is looking very good. I am making some changes to accommodate some minor dialplan changes from 1.6. Our 1.4 is very solid 1.6 has some issues with DTMF issues when used with Sonus on the back end. 1.8 is looking very good and we hope to go production before the end of the year. If you have to change righ now are you using custom dialplan code? If you are I would roll the dice and go for 1.8 this will give you the longest life span. If not there is no real big hit for stepping from 1.4 to 1.8. The other issue is if you want really detailed logging for call records the CEL method in 1.8 is the way to go. You will need to be able to boil the data down but it is there. I have seen a few kinks in the current version but it looks like they will be worked out with some incremental updates. Our hope is to be fully 1.8 on all of our backbone production units by the end of Jan 2011 with our first unit by December 2010. I would shy away of 1.6.x based on our experience. Our 1.6.x boxes will move before our 1.4.x boxes. Thanks Bryant From: "Tilghman Lesher" Sent: Wednesday, November 03, 2010 11:26 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Migration from 1.2 to 1.8 in production On Wednesday 03 November 2010 09:32:10 Danny Nicholas wrote: > satish patel wrote: > > We are running asterisk 1.2.x version in production environment since > > last 5 year and we have no issue at all, But now time to upgrade. and i > > heard about 1.8 which has introduce many features. I am wondering > > should I use asterisk 1.8 in production ? or should I go with 1.4 or > > 1.6 stable version? > > > > I would like if you suggest me which version would be good for > > production since asterisk 1.8 still in beta process. > > 1.8 will introduce many features and is the "supported standard", which > will be important to you since you are on a "5 year upgrade" plan. It > also has more "opportunities" than the 1.4 version since it is under > active development and 1.4 is in a "patch only" state. This is not the case. Both 1.8 and 1.4 are in the same state right now. The only difference in support level is that 1.4's EOL is much sooner than the EOL for 1.8. 1.6.2 will EOL at approximately the same time as 1.4. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for the most up-to-date schedule. > If immediate > stability is your goal, you may want to stick with 1.4. If I were > going to bite the bullet on 1.6, I'd jump straight to 1.8 since there > is no "end-of-life" advantage. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
If 1.2 is working fine without any problem then why do you need to upgrade to any newer version? I would suggest don't do it. If you really want to do it just for the sake of doing it, upgrade to 1.4 only, which is the most stable and well tested version of asterisk. Upgrading always causes hickups in the new system, and effects quality of service to the customers. As they say, if its not broken, don't fix it. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com On 2010-11-03 11:30 AM, "Tilghman Lesher" wrote: On Wednesday 03 November 2010 09:32:10 Danny Nicholas wrote: > satish patel wrote: > > We are running asterisk 1.2.x version in production environment since > > ... > 1.8 will introduce many features and is the "supported standard", which > will be important to you... This is not the case. Both 1.8 and 1.4 are in the same state right now. The only difference in support level is that 1.4's EOL is much sooner than the EOL for 1.8. 1.6.2 will EOL at approximately the same time as 1.4. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for the most up-to-date schedule. > If immediate > stability is your goal, you may want to stick with 1.4. If I were > going to bite... -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocati... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
Thanks a lots Bryant, I would test 1.8 and see if it work out, Definitely 1.8 going to be rock sooner or later, Let's try 1.8 Currently we are facing some issue with echo in conference call with 1.2 version hopefully it will go away with 1.8 Thanks, S. Patel From: brya...@zktech.com To: asterisk-users@lists.digium.com Date: Wed, 3 Nov 2010 11:44:24 -0400 Subject: Re: [asterisk-users] Migration from 1.2 to 1.8 in production I have used 1.4 & 1.6. I am testing 1.8 for production and it is looking very good. I am making some changes to accommodate some minor dialplan changes from 1.6. Our 1.4 is very solid 1.6 has some issues with DTMF issues when used with Sonus on the back end. 1.8 is looking very good and we hope to go production before the end of the year. If you have to change righ now are you using custom dialplan code? If you are I would roll the dice and go for 1.8 this will give you the longest life span. If not there is no real big hit for stepping from 1.4 to 1.8. The other issue is if you want really detailed logging for call records the CEL method in 1.8 is the way to go. You will need to be able to boil the data down but it is there. I have seen a few kinks in the current version but it looks like they will be worked out with some incremental updates. Our hope is to be fully 1.8 on all of our backbone production units by the end of Jan 2011 with our first unit by December 2010. I would shy away of 1.6.x based on our experience. Our 1.6.x boxes will move before our 1.4.x boxes. Thanks Bryant From: "Tilghman Lesher" Sent: Wednesday, November 03, 2010 11:26 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] Migration from 1.2 to 1.8 in production On Wednesday 03 November 2010 09:32:10 Danny Nicholas wrote: > satish patel wrote: > > We are running asterisk 1.2.x version in production environment since > > last 5 year and we have no issue at all, But now time to upgrade. and i > > heard about 1.8 which has introduce many features. I am wondering > > should I use asterisk 1.8 in production ? or should I go with 1.4 or > > 1.6 stable version? > > > > I would like if you suggest me which version would be good for > > production since asterisk 1.8 still in beta process. > > 1.8 will introduce many features and is the "supported standard", which > will be important to you since you are on a "5 year upgrade" plan. It > also has more "opportunities" than the 1.4 version since it is under > active development and 1.4 is in a "patch only" state. This is not the case. Both 1.8 and 1.4 are in the same state right now. The only difference in support level is that 1.4's EOL is much sooner than the EOL for 1.8. 1.6.2 will EOL at approximately the same time as 1.4. See https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions for the most up-to-date schedule. > If immediate > stability is your goal, you may want to stick with 1.4. If I were > going to bite the bullet on 1.6, I'd jump straight to 1.8 since there > is no "end-of-life" advantage. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
El 03/11/10 10:44, Bryant Zimmerman escribió: I have used 1.4 & 1.6. I am testing 1.8 for production and it is looking very good. I am making some changes to accommodate some minor dialplan changes from 1.6. Our 1.4 is very solid 1.6 has some issues with DTMF issues when used with Sonus on the back end. 1.8 is looking very good and we hope to go production before the end of the year. If you have to change righ now are you using custom dialplan code? If you are I would roll the dice and go for 1.8 this will give you the longest life span. If not there is no real big hit for stepping from 1.4 to 1.8. The other issue is if you want really detailed logging for call records the CEL method in 1.8 is the way to go. You will need to be able to boil the data down but it is there. I have seen a few kinks in the current version but it looks like they will be worked out with some incremental updates. Our hope is to be fully 1.8 on all of our backbone production units by the end of Jan 2011 with our first unit by December 2010. I would shy away of 1.6.x based on our experience. Our 1.6.x boxes will move before our 1.4.x boxes. Thanks Bryant Hi Bryant, Thanks for sharing your experience, it encourages us to try and test 1.8 throughly before we upgrade our 1.4 boxes, jumping the 1.6 step in the upgrade path. We are also concerned about the short support time that is left for 1.4 and 1.6, so 1.8 would be the best in support terms. This goes in concordance for what I think the asterisk team wants, that is, to focus on only one well supported version instead of having to support several parallel branches which mean more work and "cross-fixing" between them. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Migration from 1.2 to 1.8 in production
Hello! http://linuxinnovations.com lists the evolution of the various commands, applications, and other essential parts of Asterisk from version 1.4 until 1.8, so you may find it a good resource for helping you make a decision. --Elliot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Migration to Asterisk - Running off of Merlin Legend system
We currently have a Merlin Legend system. The voicemail is falling apart (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system locked up and refused to take calls; the official solution is to change the system time back to a year with a matching calendar). We are in the process of preparing the network infrastructure to support a VoIP system with Asterisk, but won't be there for a few months. We'd like to go ahead and replace the voicemail system with Asterisk now, and as we're ready, drop the Merlin system. My questions: Right now, the voicemail system (and auto-attendant) are connected to the switch by 4 analog lines. Logic says that these are FXS cards in the switch, like any other extension. The switch handles an incoming call and transfers it to the auto-attendant. How would such a call be identified to be dropped in the appropriate context? When the phone switch fails to reach someone at an extension, it transfers them to the voicemail system. How could these calls be identified as different from an incoming call to the auto-attendant? How is the appropriate mailbox or extension identified? Thanks, Steve ___ Steve Creel[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Migration to Asterisk - Running off of MerlinLegend system
On Thu, 2003-07-03 at 15:11, Steve Creel wrote: > We currently have a Merlin Legend system. The voicemail is falling apart > (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system > locked up and refused to take calls; the official solution is to change > the system time back to a year with a matching calendar). We are in the > process of preparing the network infrastructure to support a VoIP system > with Asterisk, but won't be there for a few months. We'd like to go ahead > and replace the voicemail system with Asterisk now, and as we're ready, > drop the Merlin system. > > My questions: > > Right now, the voicemail system (and auto-attendant) are connected to the > switch by 4 analog lines. Logic says that these are FXS cards in the > switch, like any other extension. The switch handles an incoming call and > transfers it to the auto-attendant. How would such a call be identified > to be dropped in the appropriate context? > > When the phone switch fails to reach someone at an extension, it transfers > them to the voicemail system. How could these calls be identified as > different from an incoming call to the auto-attendant? How is the > appropriate mailbox or extension identified? It is unlikely that you have 4 lines max to send and receive calls to voicemail and auto attendant. It may be some other technology to route TDM to the PC that does voicemail and auto attendant. If this is true, then you won't be able to use those interfaces. -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Migration to Asterisk - Running off of MerlinLegend system
It generally will send dtmf tones over the line. Get a test set/butt set and hook it up and listen to the line when the transfers occur. Will asterisk support reading dtmf before the call is picked up? dave On Thu, 3 Jul 2003, Steve Creel wrote: > We currently have a Merlin Legend system. The voicemail is falling apart > (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system > locked up and refused to take calls; the official solution is to change > the system time back to a year with a matching calendar). We are in the > process of preparing the network infrastructure to support a VoIP system > with Asterisk, but won't be there for a few months. We'd like to go ahead > and replace the voicemail system with Asterisk now, and as we're ready, > drop the Merlin system. > > My questions: > > Right now, the voicemail system (and auto-attendant) are connected to the > switch by 4 analog lines. Logic says that these are FXS cards in the > switch, like any other extension. The switch handles an incoming call and > transfers it to the auto-attendant. How would such a call be identified > to be dropped in the appropriate context? > > When the phone switch fails to reach someone at an extension, it transfers > them to the voicemail system. How could these calls be identified as > different from an incoming call to the auto-attendant? How is the > appropriate mailbox or extension identified? > > > Thanks, > > Steve > > > ___ > Steve Creel[EMAIL PROTECTED] > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Dave Weis "I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations."- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Migration to Asterisk - Running off of MerlinLegend system
On 3 Jul 2003, Steven Critchfield wrote: > On Thu, 2003-07-03 at 15:11, Steve Creel wrote: > > Right now, the voicemail system (and auto-attendant) are connected to the > > switch by 4 analog lines. Logic says that these are FXS cards in the > > switch, like any other extension. The switch handles an incoming call and > > transfers it to the auto-attendant. How would such a call be identified > > to be dropped in the appropriate context? > > > > When the phone switch fails to reach someone at an extension, it transfers > > them to the voicemail system. How could these calls be identified as > > different from an incoming call to the auto-attendant? How is the > > appropriate mailbox or extension identified? > > It is unlikely that you have 4 lines max to send and receive calls to > voicemail and auto attendant. It may be some other technology to route > TDM to the PC that does voicemail and auto attendant. If this is true, > then you won't be able to use those interfaces. That's normal on the legend. You put the 4 station ports in a hunt group and it does coverage for the other extensions. Various configurations are available from 2-12 ports. dave -- Dave Weis "I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations."- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Migration to Asterisk - Running off of Merlin Legend system
At 04:11 PM 7/3/2003 -0400, you wrote: We currently have a Merlin Legend system. The voicemail is falling apart (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system locked up and refused to take calls; the official solution is to change the system time back to a year with a matching calendar). We are in the process of preparing the network infrastructure to support a VoIP system with Asterisk, but won't be there for a few months. We'd like to go ahead and replace the voicemail system with Asterisk now, and as we're ready, drop the Merlin system. My questions: Right now, the voicemail system (and auto-attendant) are connected to the switch by 4 analog lines. Logic says that these are FXS cards in the switch, like any other extension. The switch handles an incoming call and transfers it to the auto-attendant. How would such a call be identified to be dropped in the appropriate context? Unless I'm mistaken, you just name those lines appropriately, then have matching extensions in your extensions.conf, like any other trunk. The difference being, these lines go straight to an auto attend. You could pass the extension to it via a dialed string, if need be. When the phone switch fails to reach someone at an extension, it transfers them to the voicemail system. How could these calls be identified as different from an incoming call to the auto-attendant? How is the appropriate mailbox or extension identified? The dialed extension, or maybe a callerid string.. -denon Thanks, Steve ___ Steve Creel[EMAIL PROTECTED] ___ 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] Migration to Asterisk - Running off of Merlin Legend system
If the voicemail interface really is analog, then the identification is done through DTMF codes. You'll just have to figure out the format of these digits and let Asterisk deal with them as you want. Most systems with analog voicemail ports work just like this. We've interfaced Asterisk with a Norstar system using Norstar's VMI interfaces. The Norstar VMI sends a specific format of DTMF codes (programmable) that indicate the called extension. Fancier systems can indicate the reason for the transfer (busy/unanswered), the caller's number (via CID) and the called extension. -wade > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Steve Creel > Sent: Thursday, July 03, 2003 4:11 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Migration to Asterisk - Running off of Merlin > Legend system > > We currently have a Merlin Legend system. The voicemail is falling apart > (with the transition to a 10 digit timestamp on Sept. 8, 2001, the system > locked up and refused to take calls; the official solution is to change > the system time back to a year with a matching calendar). We are in the > process of preparing the network infrastructure to support a VoIP system > with Asterisk, but won't be there for a few months. We'd like to go ahead > and replace the voicemail system with Asterisk now, and as we're ready, > drop the Merlin system. > > My questions: > > Right now, the voicemail system (and auto-attendant) are connected to the > switch by 4 analog lines. Logic says that these are FXS cards in the > switch, like any other extension. The switch handles an incoming call and > transfers it to the auto-attendant. How would such a call be identified > to be dropped in the appropriate context? > > When the phone switch fails to reach someone at an extension, it transfers > them to the voicemail system. How could these calls be identified as > different from an incoming call to the auto-attendant? How is the > appropriate mailbox or extension identified? > > > Thanks, > > Steve > > > ___ > Steve Creel[EMAIL PROTECTED] > > > > ___ > 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
