Is there a document or recommended strategy on how to migrate from an existing PBX system to FS? There are 2 T1-PRI lines used 24/7 (call center) which can't be down for development or testing. The existing system is InterTel. I have a Sangoma A104D available with a tap ready to be installed. I'm thinking (hoping?) that I can start by using it to monitor and/or record calls without interfering with the InterTel.
On Wed, Oct 22, 2008 at 9:50 AM, Thomas Troesch <ttroesch@dslextreme.com>wrote:
Is there a document or recommended strategy on how to migrate from an existing PBX system to FS? There are 2 T1-PRI lines used 24/7 (call center) which can't be down for development or testing. The existing system is InterTel.
Right now the OpenZAP stuff is still in development. I can't recommend putting it in a 24/7 can't-be-down-for-a-second kind of environment, at least not yet. We're getting there... Also, be sure to note which PRI protocol variant (aka "dialect") you are currently using. OZ supports DMS and 5ESS pretty well but NI2 is still a little iffy. Again, we're working on it but there are only so many hours in the day... As for documentation, I'm almost positive that there aren't any docs extant that describe this process. I do know that Yossi N. has migrated from Asterisk to FreeSWITCH and has some experience. His presentation at ClueCon this year is available here: http://files.freeswitch.org/cluecon_2008/Day_03.Presentation_11.Yossi_Neiman... However, I don't think we're at the point of how to rip out a legacy PBX and drop in FS. Not yet, but give us some time... :)
I have a Sangoma A104D available with a tap ready to be installed. I'm thinking (hoping?) that I can start by using it to monitor and/or record calls without interfering with the InterTel.
I am interested in knowing if you can pull this off. I'd love to see this in action. I've got an A104D and I would be very interested in seeing someone get the HI-Z setup nailed down so that you can tap T1 lines. I'm particularly interested in sniffing d-chan traffic but I would also be interested in seeing someone use FS to create a call-recording system. (I know that Tri-Sys in NJ uses YATE + Sangoma for this, so I'm 100% sure that it is possible.) If you feel like being a trailblazer then by all means go for it! Hop on IRC (#freeswitch and #openzap on irc.freenode.net) and pepper us all with questions. The only rules are "be cool" and "document it on the wiki if you learn something that wasn't already there." -MC
However, I don't think we're at the point of how to rip out a legacy PBX and drop in FS. Not yet, but give us some time... :)
Why not? You can simply use a Asterisk box for the Zap stuff. That is how we do it. Works pretty well. Better still use E1/T1 gateways or POTS gateways for analog lines, they are not that expensive anymore (if you factor in the cost of the machine itself) -- Faraz R Khan www.emergen.biz
Thank you for your input. I am clearly naive about what may be required, and I need to get more details on what is currently in use. The environment is not 'can't-be-down-for-a-second' - it has crashed before - but that became an emergency. It is possible to plan for short ( 1/2 hour ) downtimes, but not easily. Thank you for the link - it was interesting. I also listened to some talks by Diana Cionoiu of YATES and Nenad Corbic from Sangoma that I found on you-tube ( from Asterisk-Tag in Germany ). It helped by giving some context for things I'm not familiar with. I'll check in on the IRC when I have some questions. On Wednesday 22 October 2008 15:01:34 Michael Collins wrote:
On Wed, Oct 22, 2008 at 9:50 AM, Thomas Troesch <ttroesch@dslextreme.com>wrote:
Is there a document or recommended strategy on how to migrate from an existing PBX system to FS? There are 2 T1-PRI lines used 24/7 (call center) which can't be down for development or testing. The existing system is InterTel.
Right now the OpenZAP stuff is still in development. I can't recommend putting it in a 24/7 can't-be-down-for-a-second kind of environment, at least not yet. We're getting there... Also, be sure to note which PRI protocol variant (aka "dialect") you are currently using. OZ supports DMS and 5ESS pretty well but NI2 is still a little iffy. Again, we're working on it but there are only so many hours in the day...
As for documentation, I'm almost positive that there aren't any docs extant that describe this process. I do know that Yossi N. has migrated from Asterisk to FreeSWITCH and has some experience. His presentation at ClueCon this year is available here: http://files.freeswitch.org/cluecon_2008/Day_03.Presentation_11.Yossi_Nei man.1500kbps.mp4
However, I don't think we're at the point of how to rip out a legacy PBX and drop in FS. Not yet, but give us some time... :)
I have a Sangoma A104D available with a tap ready to be installed. I'm thinking (hoping?) that I can start by using it to monitor and/or record calls without interfering with the InterTel.
I am interested in knowing if you can pull this off. I'd love to see this in action. I've got an A104D and I would be very interested in seeing someone get the HI-Z setup nailed down so that you can tap T1 lines. I'm particularly interested in sniffing d-chan traffic but I would also be interested in seeing someone use FS to create a call-recording system. (I know that Tri-Sys in NJ uses YATE + Sangoma for this, so I'm 100% sure that it is possible.)
If you feel like being a trailblazer then by all means go for it! Hop on IRC (#freeswitch and #openzap on irc.freenode.net) and pepper us all with questions. The only rules are "be cool" and "document it on the wiki if you learn something that wasn't already there."
-MC
Here are some more videos from the last ClueCon http://files.freeswitch.org/cluecon_2008/ On Thu, Oct 23, 2008 at 5:03 AM, Thomas Troesch <ttroesch@dslextreme.com>wrote:
Thank you for your input. I am clearly naive about what may be required, and I need to get more details on what is currently in use.
The environment is not 'can't-be-down-for-a-second' - it has crashed before - but that became an emergency. It is possible to plan for short ( 1/2 hour ) downtimes, but not easily.
Thank you for the link - it was interesting. I also listened to some talks by Diana Cionoiu of YATES and Nenad Corbic from Sangoma that I found on you-tube ( from Asterisk-Tag in Germany ). It helped by giving some context for things I'm not familiar with.
I'll check in on the IRC when I have some questions.
On Wednesday 22 October 2008 15:01:34 Michael Collins wrote:
On Wed, Oct 22, 2008 at 9:50 AM, Thomas Troesch <ttroesch@dslextreme.com>wrote:
Is there a document or recommended strategy on how to migrate from an existing PBX system to FS? There are 2 T1-PRI lines used 24/7 (call center) which can't be down for development or testing. The existing system is InterTel.
Right now the OpenZAP stuff is still in development. I can't recommend putting it in a 24/7 can't-be-down-for-a-second kind of environment, at least not yet. We're getting there... Also, be sure to note which PRI protocol variant (aka "dialect") you are currently using. OZ supports DMS and 5ESS pretty well but NI2 is still a little iffy. Again, we're working on it but there are only so many hours in the day...
As for documentation, I'm almost positive that there aren't any docs extant that describe this process. I do know that Yossi N. has migrated from Asterisk to FreeSWITCH and has some experience. His presentation at ClueCon this year is available here:
http://files.freeswitch.org/cluecon_2008/Day_03.Presentation_11.Yossi_Nei
man.1500kbps.mp4
However, I don't think we're at the point of how to rip out a legacy PBX and drop in FS. Not yet, but give us some time... :)
I have a Sangoma A104D available with a tap ready to be installed. I'm thinking (hoping?) that I can start by using it to monitor and/or record calls without interfering with the InterTel.
I am interested in knowing if you can pull this off. I'd love to see this in action. I've got an A104D and I would be very interested in seeing someone get the HI-Z setup nailed down so that you can tap T1 lines. I'm particularly interested in sniffing d-chan traffic but I would also be interested in seeing someone use FS to create a call-recording system. (I know that Tri-Sys in NJ uses YATE + Sangoma for this, so I'm 100% sure that it is possible.)
If you feel like being a trailblazer then by all means go for it! Hop on IRC (#freeswitch and #openzap on irc.freenode.net) and pepper us all with questions. The only rules are "be cool" and "document it on the wiki if you learn something that wasn't already there."
-MC
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participants (4)
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Anthony Minessale
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Faraz Khan
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Michael Collins
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Thomas Troesch