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

[Dmitry Mordovin]

Michael Jerris wrote:
> 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 <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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>
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>   
Thanks a lot Michael!

Checked, its a problem with NAT.

Will try to find solution.

DM
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