its done by SIP UPDATE on polycom/aastra or sip INFO packets on snom when the call is bridged.<br>X-lite does not update anything when it receives them.<br><br>That's about it.<br><br><br><div class="gmail_quote">2010/4/4 Nguyễn Mạnh Hùng <span dir="ltr"><<a href="mailto:hungngm@bkav.com.vn">hungngm@bkav.com.vn</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div><font color="#000000" face="Times New Roman" size="3">
<div>Hi Anthony,</div>
<div> </div>
<div>Can you discuss some details in how polycom or snom can do this and x-lite
not.</div>
<div> </div>
<div>If can, I want to edit some open source soffphone like officeSIP to do
this.</div>
<div> </div>
<div>Best Regards.</div>
<div> </div></font><font color="#000000" face="Times New Roman" size="3">
<div style="background: rgb(240, 176, 112) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;"><a href="mailto:anthony.minessale@gmail.com" target="_blank">Anthony Minessale</a> <font color="#0000ff">[<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>]</font> </div>
<div></div></font> </div></div><div><div></div><div class="h5">
<div></div>
<div> </div>We already do it.<br>X-Lite does not support it.<br>If you try
it with a phone like snom or polycom you will see it works just like
that.<br><br><br>
<div class="gmail_quote">2010/4/3 Nguyễn Mạnh Hùng <span dir="ltr"><<a href="mailto:hungngm@bkav.com.vn" target="_blank">hungngm@bkav.com.vn</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div><font color="#000000" face="Times New Roman" size="3">
<div>Hi Seven Du.</div>
<div> </div>
<div>Thanks to yours suggetion.</div>
<div> </div>
<div>I have an ideal, it is: when the call between caller and agent is set,
the caller_id is determined. So, i want to edit code to sent the agent
information (the call_id and call_id_number) which will
be displayed againt in the agent's softphone (as
Xlite..) when the call is happening.</div>
<div> </div>
<div>I read some documents but i still can't determine: It's maybe yes or
maybe to do this and where to do this.</div>
<div> </div>
<div>Can you give me some comments.</div>
<div> </div>
<div>Best Regard.</div>
<div> </div></font><font color="#000000" face="Times New Roman" size="3">
<div style="background: rgb(240, 176, 112) none repeat scroll 0% 0%; -moz-background-clip: border; -moz-background-origin: padding; -moz-background-inline-policy: continuous;"><a href="mailto:dujinfang@gmail.com" target="_blank">Seven Du</a> <font color="#0000ff">[<a href="mailto:dujinfang@gmail.com" target="_blank">dujinfang@gmail.com</a>]</font> </div>
<div></div></font> </div></div>
<div></div>
<div> </div>
<div>As discussed in the list, it's not a freeswitch problem but a reality of
life.
<div><br><br>Think about customer A and B calls in one after another,
then if<br>FreeSWITCH call agent X with caller id A and Y with caller id B,
and<br>angent Y answers before X, then<br><br>1) if bridge Y with A with the
FIFO rule, then the caller id is wrong<br>2) if bridge Y with B, the caller id
is right but it breaks the rule<br>of FIFO - A should be served before B!! And
what even worse is that<br>if X never answer A then A never can be served
which is really<br>unfair!!<br><br>Of course you don't want 1), and you don't
need mod_fifo if you want<br>behavior 2), you just need some dialplan trick or
some simple Lua<br>script I think. Also FreeSWITCH is designed to be easily
extended with<br>almost any languages so feel free to implement
anything.<br><br></div>2010/3/31 Nguyễn Mạnh Hùng :
<div><br>> Hi Mike and Seven Du.<br>> Thanks to yours
help.<br>> I known the mechanism of mod_fifo.<br>>>><a href="http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html" target="_blank">http://lists.freeswitch.org/pipermail/freeswitch-users/2009-November/050175.html</a>.<br>
>
What a pity, It can't solve this problem. I can't use freeswitch for my
call<br>> center.<br>> Hope new version can solve this
!!!<br><br>_______________________________________________<br>FreeSWITCH-users
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<br></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>
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