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Michael Collins wrote:
<blockquote
cite="mid:87f2f3b90812220853o452bbe27w64fa3e3884b47961@mail.gmail.com"
type="cite">Dmitry,<br>
<br>
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:<br>
<br>
Auth all SIP clients from external db - <a moz-do-not-send="true"
href="http://wiki.freeswitch.org/wiki/Mod_xml_curl">http://wiki.freeswitch.org/wiki/Mod_xml_curl</a><br>
Call control - <a moz-do-not-send="true"
href="http://wiki.freeswitch.org/wiki/Event_Socket">http://wiki.freeswitch.org/wiki/Event_Socket</a><br>
LCR - <a moz-do-not-send="true"
href="http://wiki.freeswitch.org/wiki/Mod_lcr">http://wiki.freeswitch.org/wiki/Mod_lcr</a><br>
<br>
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.<br>
<br>
Hope this helps!<br>
-MC<br>
<br>
<div class="gmail_quote">On Sun, Dec 21, 2008 at 11:08 PM, Dmitry
Mordovin <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:d.mordovin@dwide.com">d.mordovin@dwide.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div bgcolor="#ffffff" text="#000099">Hi Friends!<br>
<br>
I wish to do something like:<br>
- Authorize all sip clients in external DB, for example postgresql.<br>
- Control each call session duration, just like prepaid calling card.<br>
- LCR, if one voip provider can't connect call into PSTN, try next
PSTN supplier.<br>
<br>
Please help me, i can't understand what is docs i might to read to know
how to do it.<br>
Could you show me some links and terms. explain it?<br>
<br>
Thank you so much!<br>
<font color="#888888"><br>
DM
</font></div>
<br>
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</blockquote>
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</blockquote>
Hello Michael!<br>
<br>
Thank you so much!<br>
<br>
Now i have another problem,<br>
<br>
<a class="moz-txt-link-abbreviated" href="mailto:1000@example.com">1000@example.com</a> (kphone) calling -> FS -> <a class="moz-txt-link-abbreviated" href="mailto:1001@example.com">1001@example.com</a>
(X-Lite).<br>
<br>
Sometimes ago 1000 and 1001 connected and talking.<br>
When 1001 user hungup, FS recv BYE from 1001, i see (in log) FS doing
some HUNGUPs function, writes log " Sending BYE to
<a class="moz-txt-link-abbreviated" href="mailto:sofia/internal/1000@example.com">sofia/internal/1000@example.com</a>".<br>
<br>
But <a class="moz-txt-link-abbreviated" href="mailto:1000@example.com">1000@example.com</a> Don't receive any packets since receive 200 OK on
connect.<br>
I tried UDP and TCP protocol.<br>
<br>
my dialplan:<br>
<br>
<extension name="PBX Extension"><br>
<condition field="destination_number"
expression="^(1[0-9]{3})$"><br>
<action application="set"
data="continue_on_fail=true"/><br>
<action application="set"
data="hangup_after_bridge=true"/><br>
<action application="bridge"
data="sofia/default/$1%$${domain}"/><br>
</condition><br>
</extension><br>
<br>
<br>
I found discussion about it and bug fix, but now its happens again, or
its my mistake in configuration?<br>
<br>
Thank you all!<br>
<br>
DM<br>
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