VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
stotaro at totarotechn... Guest
|
Posted: Thu Jun 12, 2008 10:36 am Post subject: [asterisk-users] Using Asterisk Only as Voice RecordingSolut |
|
|
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 <nasruddin at ncel.com.pk> wrote:
Quote: | 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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <nasruddin at ncel.com.pk>
wrote:
Quote: |
HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have
| fair
Quote: | command over Asterisk up till now and have run it in different
| scenarios
Quote: | such as Call Center Solution, PBX solution.
There is a requirement to use Asterisk only as Voice Recording
| solution in
Quote: | following manner:
Physical POT lines before entering into our native PBX will be
| splitted and
Quote: | 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
Quote: | (either SIP phone or Analog Phone) I should be able to start recording
| the
Quote: | 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
Quote: | 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
Quote: | 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
Quote: | 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
|
|
|
Back to top |
|
|
nasruddin at ncel.com.pk Guest
|
Posted: Thu Jun 12, 2008 11:01 am Post subject: [asterisk-users] Using Asterisk Only as Voice RecordingSolut |
|
|
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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <nasruddin at ncel.com.pk>
wrote:
Quote: |
HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have
| fair
Quote: | command over Asterisk up till now and have run it in different
| scenarios
Quote: | such as Call Center Solution, PBX solution.
There is a requirement to use Asterisk only as Voice Recording
| solution in
Quote: | following manner:
Physical POT lines before entering into our native PBX will be
| splitted and
Quote: | 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
Quote: | (either SIP phone or Analog Phone) I should be able to start recording
| the
Quote: | 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
Quote: | 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
Quote: | 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
Quote: | 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 |
|
Back to top |
|
|
pdhales at optusnet.co... Guest
|
Posted: Thu Jun 12, 2008 11:38 pm Post subject: [asterisk-users] Using Asterisk Only as Voice RecordingSolut |
|
|
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:
Quote: | 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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <nasruddin at ncel.com.pk>
wrote:
Quote: | HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have
| fair
Quote: | command over Asterisk up till now and have run it in different
| scenarios
Quote: | such as Call Center Solution, PBX solution.
There is a requirement to use Asterisk only as Voice Recording
| solution in
Quote: | following manner:
Physical POT lines before entering into our native PBX will be
| splitted and
Quote: | 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
Quote: | (either SIP phone or Analog Phone) I should be able to start recording
| the
Quote: | 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
Quote: | 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
Quote: | 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
Quote: | 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
|
|
|
Back to top |
|
|
stotaro at totarotechn... Guest
|
Posted: Fri Jun 13, 2008 6:30 am Post subject: [asterisk-users] Using Asterisk Only as Voice RecordingSolut |
|
|
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 <nasruddin at ncel.com.pk> wrote:
Quote: | 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 asterisk....what 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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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:
Quote: | Thanks Steve,
How I can use it "Asterisk" as Man In The Middle. Since we have to
| keep
Quote: | 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
Quote: | system generated event regarding OFF-HOOK and ON-HOOK condition
| through
Quote: | 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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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
| <nasruddin at ncel.com.pk>
Quote: | wrote:
Quote: | HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have
| fair
Quote: | command over Asterisk up till now and have run it in different
| scenarios
Quote: | such as Call Center Solution, PBX solution.
There is a requirement to use Asterisk only as Voice Recording
| solution in
Quote: | following manner:
Physical POT lines before entering into our native PBX will be
| splitted and
Quote: | 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
Quote: | (either SIP phone or Analog Phone) I should be able to start
|
| recording
Quote: | the
Quote: | 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
Quote: | 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
Quote: | 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
Quote: | 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
|
|
|
Back to top |
|
|
nasruddin at ncel.com.pk Guest
|
Posted: Fri Jun 13, 2008 7:05 am Post subject: [asterisk-users] Using Asterisk Only as Voice RecordingSolut |
|
|
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 asterisk....what 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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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:
Quote: | Thanks Steve,
How I can use it "Asterisk" as Man In The Middle. Since we have to
| keep
Quote: | 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
Quote: | system generated event regarding OFF-HOOK and ON-HOOK condition
| through
Quote: | 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: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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
| <nasruddin at ncel.com.pk>
Quote: | wrote:
Quote: | HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have
| fair
Quote: | command over Asterisk up till now and have run it in different
| scenarios
Quote: | such as Call Center Solution, PBX solution.
There is a requirement to use Asterisk only as Voice Recording
| solution in
Quote: | following manner:
Physical POT lines before entering into our native PBX will be
| splitted and
Quote: | 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
Quote: | (either SIP phone or Analog Phone) I should be able to start
|
| recording
Quote: | the
Quote: | 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
Quote: | 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
Quote: | 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
Quote: | 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 |
|
Back to top |
|
|
nasruddin at ncel.com.pk Guest
|
Posted: Mon Jun 16, 2008 6:55 am Post subject: [asterisk-users] Using Asterisk Only as Voice RecordingSolut |
|
|
Thanks for the link. I think I will be using this product.
Syed Nasruddin
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] 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 <nasruddin at ncel.com.pk>:
Quote: |
HI,
I am using TDM800P Digium Card with Asterisk 1.4.* version. I have
| fair
Quote: | command over Asterisk up till now and have run it in different
| scenarios
Quote: | such as Call Center Solution, PBX solution.
There is a requirement to use Asterisk only as Voice Recording
| solution in
Quote: | following manner:
Physical POT lines before entering into our native PBX will be
| splitted and
Quote: | 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
Quote: | (either SIP phone or Analog Phone) I should be able to start recording
| the
Quote: | 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 |
|
Back to top |
|
|
gavin.henry at gmail.com Guest
|
Posted: Tue Jun 17, 2008 2:01 pm Post subject: [asterisk-users] Using Asterisk Only as Voice RecordingSolut |
|
|
2008/6/16 Syed Nasruddin <nasruddin at ncel.com.pk>:
Quote: |
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/ |
|
Back to top |
|
|
|
|
|
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|