<div dir="ltr"><div>Hi,<br><br></div>Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:<br><br>if ($rU=="12345") {<div> if(is_method("INVITE")) {</div><div> record_route()<span class="gmail-pl-c"><span class="gmail-pl-c">;</span></span></div><div> $ru = "sip:" + "offline" + "@" + $sel(cfg_get.voicemail.srv_ip)</div><div> + ":" + $sel(cfg_get.voicemail.srv_<wbr>port);</div><div> route(RELAY);</div><div> exit;</div><div> }</div><div> }</div><div><br></div><div>With kind regards,<br></div></div><div class="gmail_extra"><br clear="all"><div><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr">Jurijs<br></div></div></div>
<br><div class="gmail_quote">On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <span dir="ltr"><<a href="mailto:zhaoguojie2010@163.com" target="_blank">zhaoguojie2010@163.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="line-height:1.7;color:#000000;font-size:14px;font-family:Arial"><div style="zoom:1">Hello <div style="clear:both"></div></div><div style="zoom:1"> I added below code to let kamailio route invite to freeswitch:</div><div style="zoom:1"> if ($rU=="12345") {</div><div style="zoom:1"> if(is_method("INVITE")) {</div><div style="zoom:1"> $ru = "sip:" + "offline" + "@" + $sel(cfg_get.voicemail.srv_ip)</div><div style="zoom:1"> + ":" + $sel(cfg_get.voicemail.srv_<wbr>port);</div><div style="zoom:1"> route(RELAY);</div><div style="zoom:1"> exit;</div><div style="zoom:1"> }</div><div style="zoom:1"> }</div><div style="zoom:1"><br></div><div style="zoom:1"> in freeswitch dialplan/default.xml, i added</div><div style="zoom:1"> <extension name="prompt-offline"></div><div style="zoom:1"> <condition field="destination_number" expression="^offline$"></div><div style="zoom:1"> <action application="bridge" data="user/1000@${domain_name}<wbr>"/></div><div style="zoom:1"> <action application="playback" data="/usr/local/freeswitch/<wbr>sounds/music/8000/suite-<wbr>espanola-op-47-leyenda.wav"/></div><div style="zoom:1"> </condition></div><div style="zoom:1"> </extension></div><div style="zoom:1"><br></div><div style="zoom:1">when i dialed 12345 on sip client, I can see the invite package to freeswitch, and that's it. No package coming back from freeswitch. Eventually, the sip client timeout. I</div><div style="zoom:1">was hoping that when i dial 12345, "suite-espanola-op-47-leyenda.<wbr>wav" will be played. What did i do wrong?</div><div style="zoom:1"><br></div><div style="zoom:1">Thanks</div><div><div class="h5"><div id="m_-5606544910387386882divNeteaseMailCard"></div><br>At 2017-09-20 19:32:14, "Sergey Safarov" <<a href="mailto:s.safarov@gmail.com" target="_blank">s.safarov@gmail.com</a>> wrote:<br> <blockquote id="m_-5606544910387386882isReplyContent" style="PADDING-LEFT:1ex;MARGIN:0px 0px 0px 0.8ex;BORDER-LEFT:#ccc 1px solid"><div dir="ltr">You can add this example to dialplan and make test<div><br></div><div><div><font face="monospace"> <extension name="call_user"></font></div><div><font face="monospace"> <condition></font></div><div><font face="monospace"> <action application="set" data="continue_on_fail=NORMAL_<wbr>TEMPORARY_FAILURE,USER_BUSY,<wbr>NO_ROUTE_DESTINATION,<wbr>SUBSCRIBER_ABSENT"/></font></div><div><font face="monospace"> <action application="bridge" data="user/<a href="mailto:3000@example.org" target="_blank">3000@example.org</a>"/></font></div><div><font face="monospace"> <action application="playback" data="ivr/ivr-user_busy.wav"/></font></div><div><font face="monospace"> </condition></font></div><div><font face="monospace"> </extension></font></div></div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr">ср, 20 сент. 2017 г. в 10:14, 赵国杰 <<a href="mailto:zhaoguojie2010@163.com" target="_blank">zhaoguojie2010@163.com</a>>:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="line-height:1.7;color:#000000;font-size:14px;font-family:Arial"><div>Hello Sergey,</div><div> I installed freeswitch, what should i do next?</div></div><div style="line-height:1.7;color:#000000;font-size:14px;font-family:Arial"><br><br><br><div style="zoom:1"><br><div style="clear:both"></div></div><div id="m_-5606544910387386882m_2085783873288216747divNeteaseMailCard"></div><br>At 2017-09-19 12:07:23, "Sergey Safarov" <<a href="mailto:s.safarov@gmail.com" target="_blank">s.safarov@gmail.com</a>> wrote:<br> <blockquote id="m_-5606544910387386882m_2085783873288216747isReplyContent" style="PADDING-LEFT:1ex;MARGIN:0px 0px 0px 0.8ex;BORDER-LEFT:#ccc 1px solid"><p dir="ltr">This can be implemenred using freeswitch.<br>
Ping me directly after you install freeswith on linux and configure ssh remote access</p>
<br><div class="gmail_quote"><div dir="ltr">вт, 19 сент. 2017 г., 6:27 赵国杰 <<a href="mailto:zhaoguojie2010@163.com" target="_blank">zhaoguojie2010@163.com</a>>:<br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="line-height:1.7;color:#000000;font-size:14px;font-family:Arial"><div>Thanks Daniel,</div><div> I've done some digging, and from Andrew Prokop's blog, it says this envolves early midia. Usually this is done by reply a 183 to the caller with media ip and port in the SDP. This makes sense but i still have no idea how to generate 183 response with embedded SDP.</div></div><div style="line-height:1.7;color:#000000;font-size:14px;font-family:Arial"><br><div style="zoom:1"><br><div style="clear:both"></div></div><div id="m_-5606544910387386882m_2085783873288216747m_4331241580758290794divNeteaseMailCard"></div><br><pre><br>At 2017-09-18 18:05:46, "Daniel Tryba" <<a href="mailto:d.tryba@pocos.nl" target="_blank">d.tryba@pocos.nl</a>> wrote:
>On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
>> I want the caller to play a short audio(like "the number your are calling is busy") when the callee declines the call. How can i do that?
>
>You need to check for the status codes in a failure route and then
>somehow generate audio somewhere, which is out of the scope of kamailio
>(maybe rtpproxy can do this, otherwise use something like asterisk):
>
>failure_route[MANAGE_FAILURE] {
>if (t_check_status("486"))
>{
> $du=null;
> $ru="<a href="mailto:busymessage@asterisk.example.org" target="_blank">busymessage@asterisk.<wbr>example.org</a>";
> route(RELAY);
> exit;
>}
>
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