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

[Michael Collins]

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 at avgs.ca> wrote:
> 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 at googlemail.com>
> wrote:
>>
>> 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 :).
>> 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
>>
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