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

[Michael Jerris]

This sounds like a nat issue where we are sending a BYE but its not  
getting to the other end.  Check the sip trace and see if that is the  
case.

Mike

On Dec 23, 2008, at 5:09 AM, Dmitry Mordovin wrote:

> 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 
>> > 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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>>
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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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