Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
2008/6/16 Syed Nasruddin [EMAIL PROTECTED]: Thanks for the link. I think I will be using this product. It's very, very good. You can hook it up to MySQL instead of sqlite if needed, just e-mail support. -- http://www.suretecsystems.com/services/openldap/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Thanks for the link. I think I will be using this product. Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Henry Sent: Saturday, June 14, 2008 1:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. 2008/6/12 Syed Nasruddin [EMAIL PROTECTED]: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Our clients use this for E1 Pri: http://www.voicetronix.com/logger.htm Not sure if there is a analogue solution. -- http://www.suretecsystems.com/services/openldap/ ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Dear PaulH, I have 5 PSTN Lines going into my legacy PBX. There is an active IVR present on legacy PBX which the client wants to keep. So what I have to do is: 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine. 2. Insert All those PSTN directly to my 5-Port FXO. 3. Take out 5-FXS Port lines and insert them into my legacy PBX. 4. Since as I mentioned previously that my client wants to keep its IVR intact on its Legacy system so I will not be handling IVR in my Asterisk Dialplan. 5. when the call arrives at asteriskwhat should I do?? Should I simply call Dial(FXS channel) or something else. Kindly provide some info regarding Step 5. Thanks Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, June 13, 2008 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. Basically, you run the phone lines into the asterisk box, then out of the Asterisk system into the PABX. This works reasonably well, and gives you the option to migrate to a full asterisk setup in the future. PaulH Syed Nasruddin wrote: Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- 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 ___ -- 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 ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Step five: Profit ;-) I am not going to write your dialplan for you but here is a clue. http://www.voip-info.org/wiki/view/Asterisk+legacy+integration Of those various setups, you can extract what you need. Thanks, Steve T On Fri, Jun 13, 2008 at 8:05 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: Dear PaulH, I have 5 PSTN Lines going into my legacy PBX. There is an active IVR present on legacy PBX which the client wants to keep. So what I have to do is: 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine. 2. Insert All those PSTN directly to my 5-Port FXO. 3. Take out 5-FXS Port lines and insert them into my legacy PBX. 4. Since as I mentioned previously that my client wants to keep its IVR intact on its Legacy system so I will not be handling IVR in my Asterisk Dialplan. 5. when the call arrives at asteriskwhat should I do?? Should I simply call Dial(FXS channel) or something else. Kindly provide some info regarding Step 5. Thanks Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, June 13, 2008 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. Basically, you run the phone lines into the asterisk box, then out of the Asterisk system into the PABX. This works reasonably well, and gives you the option to migrate to a full asterisk setup in the future. PaulH Syed Nasruddin wrote: Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- 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 ___ -- 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 ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Thanks Steve, Sure, although I would have loved to see a pre-config dialplan:. Thanks for the tip. I think it will help me through. Best Regards Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, June 13, 2008 4:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. Step five: Profit ;-) I am not going to write your dialplan for you but here is a clue. http://www.voip-info.org/wiki/view/Asterisk+legacy+integration Of those various setups, you can extract what you need. Thanks, Steve T On Fri, Jun 13, 2008 at 8:05 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: Dear PaulH, I have 5 PSTN Lines going into my legacy PBX. There is an active IVR present on legacy PBX which the client wants to keep. So what I have to do is: 1. Install 10 Port i.e 5-FXO/5-FXS card on asterisk machine. 2. Insert All those PSTN directly to my 5-Port FXO. 3. Take out 5-FXS Port lines and insert them into my legacy PBX. 4. Since as I mentioned previously that my client wants to keep its IVR intact on its Legacy system so I will not be handling IVR in my Asterisk Dialplan. 5. when the call arrives at asteriskwhat should I do?? Should I simply call Dial(FXS channel) or something else. Kindly provide some info regarding Step 5. Thanks Syed Nasruddin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Friday, June 13, 2008 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. Basically, you run the phone lines into the asterisk box, then out of the Asterisk system into the PABX. This works reasonably well, and gives you the option to migrate to a full asterisk setup in the future. PaulH Syed Nasruddin wrote: Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
You will need exactly two times the number of ports that your legacy system has. Asterisk takes the call on _.,1,DAHDI, starts monitor and dials out the second DAHDI port to your legacy system. It is about ten lines in extensions.conf. Thanks, Steve T On Thu, Jun 12, 2008 at 12:01 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- 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 ___ -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution.
Basically, you run the phone lines into the asterisk box, then out of the Asterisk system into the PABX. This works reasonably well, and gives you the option to migrate to a full asterisk setup in the future. PaulH Syed Nasruddin wrote: Thanks Steve, How I can use it Asterisk as Man In The Middle. Since we have to keep our Native PBX intact and functioning but only thing it doesn't handle is Voice Recording. I thought if I can get some Channel Variable or some system generated event regarding OFF-HOOK and ON-HOOK condition through Asterisk I will easily handle this requirement. It will be a great help if you can elaborate how I can use asterisk as man-in-the-middle configuration along with my current PBX. Thanks a lot for your prompt response Syed Nasruddin (CISSP) Assistant Manager - Systems National Commodity Exchange Limited 9th Floor, PIC Towers 32-A Lalazar Drive M.T. Khan Road Karachi Phone: 111623623 ext 217 Fax: 5611263 Web: www.ncel.com.pk -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, June 12, 2008 7:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using Asterisk Only as Voice RecordingSolution. On Thu, Jun 12, 2008 at 11:16 AM, Syed Nasruddin [EMAIL PROTECTED] wrote: HI, I am using TDM800P Digium Card with Asterisk 1.4.* version. I have fair command over Asterisk up till now and have run it in different scenarios such as Call Center Solution, PBX solution. There is a requirement to use Asterisk only as Voice Recording solution in following manner: Physical POT lines before entering into our native PBX will be splitted and one of each of those lines will also enter into our Asterisk System. Once the call is routed by our native PBX and recipient picks up the phone (either SIP phone or Analog Phone) I should be able to start recording the call. When the call ends, the recording should stop. Problem being faced by me is this that I am able to catch the call in my diaplan and initialize MixMonitor but since my diaplan never detects OFF-HOOK/ON_HOOK condition it continue to wait and finally hang-up while in actual the call is running through our PBX. Is there any channel variable or any other mechanism by which I can accomplish this task? Since i will not be using any Dial() or similar application I will be needing some kind of OFF-HOOK trigger/Event in my dialplan. Your help will be highly appreciated. regards Syed Nasruddin It may not be possible to do this in parallel the way you are trying now. In series should be a simple task. Just pass the call through Asterisk as the man in the middle, the dialplan will be very simple. Thanks, Steve T ___ -- 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 ___ -- 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 ___ -- 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
