<div dir="ltr">You can put what should be the To username for the call as part of the dialstring (something cool I found recently)<div><br></div><div>So say you have your contact set to:</div><div><br></div><div>sofia/profile/user%domain</div>
<div><br></div><div>add a caret(^) followed by the username you want in the To: header in the outgoing invite, for example:</div><div><br></div><div>sofia/profile/user%domain^443300100960</div><div><br></div><div>If I could remember where I found that I'd cite it.</div>
<div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On 30 November 2013 18:50, Nikita Tarasov <span dir="ltr"><<a href="mailto:luckyraul@gmx.com" target="_blank">luckyraul@gmx.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div>I already have tried mod_callcenter
with queue.<br>
<br>
All worked perfectly except only one thing - calls was routed
with "call center server" peer number(if it was registered as
agent). So I can't keep original destination number. <br>
Also I have tried to make mod_callcenter agent concact via sofia
external gateway. Nevertheless my <span lang="en"><span>attempts </span></span>to
change destination number for restoring calls via channel's sip
header variables ended with security issue.<br>
<br>
Really helpfull will be xml diaplan code, if it can be done
another way.<br>
<br>
Thanks.<br>
<br>
<br>
30.11.2013 22:34, Ali Pey пишет:<br>
</div><div><div class="h5">
<blockquote type="cite">
<div dir="ltr">Why don't you try to use Freeswitch queue feature?
When calls are routed to freeswitch, freeswitch can queue them
up and then route them to your call center server when you call
center is reachable.
<div>
<br>
</div>
<div>Freeswitch can ping (Options) your call center server to
detect if it's reachable or not. <br>
<div><br>
</div>
<div>Regards,</div>
<div>Ali Pey<br>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">
On Sat, Nov 30, 2013 at 2:31 AM, Nikita Tarasov <span dir="ltr"><<a href="mailto:luckyraul@gmx.com" target="_blank">luckyraul@gmx.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div style="line-height:initial">
<div>Let me rephrase. I am trying to
make redundant our priopretary call center server,
which doesn't have own capabilities to do it. For
that I am using opensips for detection of failure,
and for traffic routing. freeswitch is for
temporary replacement, while that software is
down. I need to make on freeswitch waiting buffer
and release it to recovered server with keeping
destination number. </div>
<div>
<br style="display:initial">
</div>
<div>
</div>
<table style="border-spacing:0px" width="100%">
<tbody>
<tr>
<td colspan="2" style="text-align:initial;font-size:initial">
<div>
<div><b>From: </b>Ali Pey</div>
<div><b>Sent: </b>Saturday, 30 November
2013 07:48</div>
<div><b>To: </b>FreeSWITCH Users Help</div>
<div><b>Reply To: </b>FreeSWITCH Users
Help</div>
<div><b>Subject: </b>Re:
[Freeswitch-users] Disastery Plan</div>
</div>
</td>
</tr>
</tbody>
</table>
<div>
<div><br>
<div>
<div dir="ltr">Hello,
<div>
<br>
</div>
<div>It's not quite clear what you are
asking. Are you trying to have redundant
opensips servers or redundant freeswitch
servers?<br>
<div class="gmail_extra">What do you mean
by ONE sip server? Everything is a sip
server.</div>
<div class="gmail_extra"><br>
</div>
<div class="gmail_extra">Opensips is quite
capable of detecting if a server is down
and re-route the calls to a redundant
server. That's what the load balancer
module does.</div>
<div class="gmail_extra">
<br>
</div>
<div class="gmail_extra">Regards,</div>
<div class="gmail_extra">Ali Pey</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On Thu, Nov 28,
2013 at 6:21 AM, Nikita Tarasov <span dir="ltr"><<a href="mailto:luckyraul@gmx.com" target="_blank">luckyraul@gmx.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><span style="font-family:Verdana"><span style="font-size:12px">
<p class="MsoNormal"><span style="font-family:Verdana"><span style="font-size:12px"><span>Hi
guys. </span></span></span></p>
<p class="MsoNormal"><span style="font-family:Verdana"><span style="font-size:12px"> </span></span></p>
<p class="MsoNormal"><span style="font-family:Verdana"><span style="font-size:12px"><span>We
have
loadbalancer(opensips)
and a huge amount of
incoming calls going to
ONE sip server, which is
unscalable and
priopretary. </span></span></span></p>
<p class="MsoNormal"><span style="font-family:Verdana"><span style="font-size:12px"><span>I
need to create some kind
of disastery plan for
this server service is
down or on high load. </span></span></span></p>
<p class="MsoNormal">
<span style="font-family:Verdana"><span style="font-size:12px"><span>The
main idea of this plan
is, when disastery
happends, to balance all
incoming calls to
freeswitch and keep them
with music, while main
server is down.</span></span></span></p>
<p class="MsoNormal"><span style="font-family:Verdana"><span style="font-size:12px"><span>On
recovery calls should be
forwarded to main server
with the same
destination number as
they was.</span></span></span></p>
<p class="MsoNormal">
<span style="font-family:Verdana"><span style="font-size:12px"><span>I
have tryed to make it
with mod_callcenter. It
was done by placing
calls in queue and when
main server is on and
re-registered on
freeswitch, calls was
forwarded. </span>But
without destination
number.</span></span></p>
<p class="MsoNormal"><span style="font-family:Verdana"><span style="font-size:12px"><span>And
I can't copy destination
number to forwarded
calls.</span></span></span></p>
<br>
<span style="font-family:Verdana"><span style="font-size:12px">Can
you help me?</span></span></span></span>
<br>
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<pre>_________________________________________________________________________
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