Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[Freeswitch-users] Few question regarding move from Asterisk to FS - resend


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users
View previous topic :: View next topic  
Author Message
mrene_lists at avgs.ca
Guest





PostPosted: Fri Jan 23, 2009 12:39 pm    Post subject: [Freeswitch-users] Few question regarding move from Asterisk Reply with quote

1) A lot of people use openzap in production environements
2) probably, even if openzap doesnt implement it (which I think it does), you can use call groups to achieve the same results
3) freeswitch has a db application/api that does the same.
4) it sets the "nibble_total_billed" channel variable, which you can use in your cdrs.


Mathieu


On Fri, Jan 23, 2009 at 7:20 AM, Ivica Samija <ivica.lists@googlemail.com (ivica.lists@googlemail.com)> wrote:
Quote:
Last message was incomplete, sorry for that, resending.

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 Smile.
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 would 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.
We use OpenVox T1/E1 cards (D210P and D410P) with wct4xxp module.
We use ccs, hdb3, crc4, loadzone=it in zaptel.conf
We use switchtype=euroisdn, pridialplan=international, pridialplan=unknown,
pri_cpe and pri_net signaling.
Any suggestions are welcome.

2) Is it possible do bridge to Zap group instead of channel
something like Dial(Zap/g4/${EXTEN:1},60,t)?

3) If I understood correct I can talk with SQLite (or other DB) only
through Lua, Javascript,...
there is nothing similar to Set(mobile=${DB(mob/${EXTEN:1})})

4) mod_nibblebill, is it possible to get resulting cost per channel in
some channel variable
and than put value in custom cdr field

Sorry on bad english, not a native speaker.
Best regards,
Ivica

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Back to top
ivica.lists at googlem...
Guest





PostPosted: Fri Jan 23, 2009 12:42 pm    Post subject: [Freeswitch-users] Few question regarding move from Asterisk Reply with quote

Last message was incomplete, sorry for that, resending.

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 Smile.
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 would 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.
We use OpenVox T1/E1 cards (D210P and D410P) with wct4xxp module.
We use ccs, hdb3, crc4, loadzone=it in zaptel.conf
We use switchtype=euroisdn, pridialplan=international, pridialplan=unknown,
pri_cpe and pri_net signaling.
Any suggestions are welcome.

2) Is it possible do bridge to Zap group instead of channel
something like Dial(Zap/g4/${EXTEN:1},60,t)?

3) If I understood correct I can talk with SQLite (or other DB) only
through Lua, Javascript,...
there is nothing similar to Set(mobile=${DB(mob/${EXTEN:1})})

4) mod_nibblebill, is it possible to get resulting cost per channel in
some channel variable
and than put value in custom cdr field

Sorry on bad english, not a native speaker.
Best regards,
Ivica

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Back to top
msc at freeswitch.org
Guest





PostPosted: Fri Jan 23, 2009 3:12 pm    Post subject: [Freeswitch-users] Few question regarding move from Asterisk Reply with quote

FYI, sorry, I responded to the other email first!

Per Mathieu's replies, yes FreeSWITCH can do all of those things that
you mentioned. The key for you will be to unlearn "the Asterisk way"
because much of the way Asterisk does things is a result of working
around inherent limitations in the system. FreeSWITCH is designed to
avoid those limitations wherever possible, so in many cases there are
ways to accomplish what you are doing with Asterisk without copying
what Asterisk does.

My advice to you is to get some test boxes to play with. Try doing
various things with FreeSWITCH to see if it does what you need it to
do. And go slowly. FreeSWITCH does all sorts of things so give
yourself some time to learn it all.

-MC

On Fri, Jan 23, 2009 at 9:39 AM, Mathieu Rene <mrene_lists@avgs.ca> wrote:
Quote:
1) A lot of people use openzap in production environements
2) probably, even if openzap doesnt implement it (which I think it does),
you can use call groups to achieve the same results
3) freeswitch has a db application/api that does the same.
4) it sets the "nibble_total_billed" channel variable, which you can use in
your cdrs.


Mathieu


On Fri, Jan 23, 2009 at 7:20 AM, Ivica Samija <ivica.lists@googlemail.com>
wrote:
Quote:

Last message was incomplete, sorry for that, resending.

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 Smile.
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 would 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.
We use OpenVox T1/E1 cards (D210P and D410P) with wct4xxp module.
We use ccs, hdb3, crc4, loadzone=it in zaptel.conf
We use switchtype=euroisdn, pridialplan=international,
pridialplan=unknown,
pri_cpe and pri_net signaling.
Any suggestions are welcome.

2) Is it possible do bridge to Zap group instead of channel
something like Dial(Zap/g4/${EXTEN:1},60,t)?

3) If I understood correct I can talk with SQLite (or other DB) only
through Lua, Javascript,...
there is nothing similar to Set(mobile=${DB(mob/${EXTEN:1})})

4) mod_nibblebill, is it possible to get resulting cost per channel in
some channel variable
and than put value in custom cdr field

Sorry on bad english, not a native speaker.
Best regards,
Ivica

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> freeSWITCH Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
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

VoiceMeUp - Corporate & Wholesale VoIP Services