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ivica.lists at googlem...
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 Posted: Fri Jan 23, 2009 12:41 pm Post subject: [Freeswitch-users] Few question regarding move from Asterisk |
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Hi all,
our company have implemented two Asterisk servers to:
- connect two company sites
- transition to IP telephony
- cut down TCO regarding telephony
Our interconnection schema:
--T1/E1 provider1--< >
--T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 >
---SIP provider3---< >
|
SIP trunk
|
< Asterisk2 >--T1/E1/ trunk--< propriety PBX2 >
On each site we have number of IP phones connected to Asterisk and
analog phones connected to propriety PBX.
Features implemented on Asterisk boxes are:
- IVR
- queue
- conference
- DISA
- BLF
- transfer calls
All was more or less good until we had to implement call rating (we
have to keep track cost made on each extension for statistic).
Company policy is that implementation has to be in house. We hit brick
wall because Asterisk have inaccurate CDRs (transfers, forwards,...)
I am looking in FS for last few weeks and it seams to me that it can
replace our Asterisk boxes, and more  .
I am a little confused with XML config but it seams to me that it is
worth of learning.
For most of my question I have found answers in documentation wiki and
on list archive, but I have just a few question still without answer.
I am sorry if the answers are out there but I was to clumsy to find
them. In that case some info or link woud be great.
So here we go:
1) OpenZap is stable enough that it can be used in production? As you
can see we depend on 4 zap trunks.
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msc at freeswitch.org
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| Posted: Fri Jan 23, 2009 3:08 pm Post subject: [Freeswitch-users] Few question regarding move from Asterisk |
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On Fri, Jan 23, 2009 at 3:53 AM, Ivica Samija
<ivica.lists@googlemail.com> wrote:
| Quote: | Hi all,
our company have implemented two Asterisk servers to:
- connect two company sites
- transition to IP telephony
- cut down TCO regarding telephony
Our interconnection schema:
--T1/E1 provider1--< >
--T1/E1 provider2--< Asterisk1 >--T1/E1/ trunk--< propriety PBX1 >
---SIP provider3---< >
|
SIP trunk
|
< Asterisk2 >--T1/E1/ trunk--< propriety PBX2 >
On each site we have number of IP phones connected to Asterisk and
analog phones connected to propriety PBX.
Features implemented on Asterisk boxes are:
- IVR
- queue
- conference
- DISA
- BLF
- transfer calls
All was more or less good until we had to implement call rating (we
have to keep track cost made on each extension for statistic).
Company policy is that implementation has to be in house. We hit brick
wall because Asterisk have inaccurate CDRs (transfers, forwards,...)
I am looking in FS for last few weeks and it seams to me that it can
replace our Asterisk boxes, and more .
I am a little confused with XML config but it seams to me that it is
worth of learning.
For most of my question I have found answers in documentation wiki and
on list archive, but I have just a few question still without answer.
I am sorry if the answers are out there but I was to clumsy to find
them. In that case some info or link woud be great.
So here we go:
1) OpenZap is stable enough that it can be used in production? As you
can see we depend on 4 zap trunks.
|
I would highly recommend doing a test with your provider and your PBX
to see if there are any interop issues. Also, OpenZAP has a PRI issue
where channels don't go back to DOWN (i.e. idle/on-hook) properly.
It's a known issue that is being worked on but it's being done in a
volunteer's spare time so there's no ETA on that. I would recommend
that you hop on IRC and join both #freeswitch and #openzap channels.
User Cypromis is putting some openzap stuff into production next week
so you might want to pick his brain on the issues that he sees.
-MC
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