[Freeswitch-dev] FreeSwitch, application or module!?

[Dmitry Mordovin]

Michael Collins wrote:
> Dmitry,
>
> FreeSWITCH can definitely help you with all of this, HOWEVER, there's 
> a lot in this scenario that isn't specifically FS. Let me ask you this 
> question: do you have a programmer that can handle the interfacing 
> necessary with FS? I believe you will need to read up on a few things:
>
> Auth all SIP clients from external db - 
> http://wiki.freeswitch.org/wiki/Mod_xml_curl
> Call control - http://wiki.freeswitch.org/wiki/Event_Socket
> LCR - http://wiki.freeswitch.org/wiki/Mod_lcr
>
> I can tell you that there isn't already a whole package with all of 
> this, but rather just some of the individual components that will need 
> to be put together. Will you also need a billing system? If so you'll 
> need to make sure that you can handle CDRs. You have multiple choices 
> there as well. FS does not support logging directly to a backend 
> database, but it does support generating CDR records that are "INSERT 
> INTO..." commands that can easily be run to autopopulate your CDR 
> table. There's also XML CDRs which allow for extremely specific 
> parsing of call history. The catch is that you need a good XML parser 
> and some business logic to make sure that this information is all 
> useful to you.
>
> Hope this helps!
> -MC
>
> On Sun, Dec 21, 2008 at 11:08 PM, Dmitry Mordovin 
> <d.mordovin at dwide.com <mailto:d.mordovin at dwide.com>> wrote:
>
>     Hi Friends!
>
>     I wish to do something like:
>      - Authorize all sip clients in external DB, for example postgresql.
>      - Control each call session duration, just like prepaid calling card.
>      - LCR, if one voip provider can't connect call into PSTN, try
>     next PSTN supplier.
>
>     Please help me, i can't understand what is docs i might to read to
>     know how to do it.
>     Could you show me some links and terms. explain it?
>
>     Thank you so much!
>
>     DM
>
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Hello Michael!

Thank you so much!

Now i have another problem,

1000 at example.com (kphone) calling -> FS -> 1001 at example.com (X-Lite).

Sometimes ago 1000 and 1001 connected and talking.
When 1001 user hungup, FS recv BYE from 1001, i see (in log) FS doing 
some HUNGUPs function, writes log " Sending BYE to 
sofia/internal/1000 at example.com".

But 1000 at example.com Don't receive any packets since receive 200 OK on 
connect.
I tried UDP and TCP protocol.

my dialplan:

        <extension name="PBX Extension">
            <condition field="destination_number" 
expression="^(1[0-9]{3})$">
                <action application="set" data="continue_on_fail=true"/>
                <action application="set" data="hangup_after_bridge=true"/>
                <action application="bridge" 
data="sofia/default/$1%$${domain}"/>
            </condition>
        </extension>


I found discussion about it and bug fix, but now its happens again, or 
its my mistake in configuration?

Thank you all!

DM
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