[SR-Users] how to play ring tune when callee declines

[赵国杰]

Hello,
    Thanks for the heads up. The siptrace does help.
    Now the FS returns(with or without record_route();): 
      SIP/2.0 480 Temporarily Unavailable
      Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
    
   I have generate offline.xml under conf/directory/default. Where did i miss?


Thanks







At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga at gmail.com> wrote:

Hi,


Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:

if ($rU=="12345") {
                if(is_method("INVITE")) {
                        record_route();
                        $ru = "sip:" + "offline" + "@" + $sel(cfg_get.voicemail.srv_ip)
                                        + ":" + $sel(cfg_get.voicemail.srv_port);
                        route(RELAY);
                        exit;
                }
        }


With kind regards,



Jurijs



On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010 at 163.com> wrote:

Hello 
    I added below code to let kamailio route invite to freeswitch:
    if ($rU=="12345") {
                if(is_method("INVITE")) {
                        $ru = "sip:" + "offline" + "@" + $sel(cfg_get.voicemail.srv_ip)
                                        + ":" + $sel(cfg_get.voicemail.srv_port);
                        route(RELAY);
                        exit;
                }
        }


      in freeswitch dialplan/default.xml, i added
     <extension name="prompt-offline">
      <condition field="destination_number" expression="^offline$">
        <action application="bridge" data="user/1000@${domain_name}"/>
        <action application="playback" data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/>
      </condition>
    </extension>


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
was hoping that when i dial 12345, "suite-espanola-op-47-leyenda.wav" will be played. What did i do wrong?


Thanks

At 2017-09-20 19:32:14, "Sergey Safarov" <s.safarov at gmail.com> wrote:

You can add this example to dialplan and make test


    <extension name="call_user">
      <condition>
        <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ROUTE_DESTINATION,SUBSCRIBER_ABSENT"/>
        <action application="bridge" data="user/3000 at example.org"/>
        <action application="playback" data="ivr/ivr-user_busy.wav"/>
      </condition>
    </extension>




ср, 20 сент. 2017 г. в 10:14, 赵国杰 <zhaoguojie2010 at 163.com>:

Hello Sergey,
     I installed freeswitch, what should i do next?







At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov at gmail.com> wrote:


This can be implemenred using freeswitch.
Ping me directly after you install freeswith on linux and configure ssh remote access



вт, 19 сент. 2017 г., 6:27 赵国杰 <zhaoguojie2010 at 163.com>:

Thanks Daniel,
    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.







At 2017-09-18 18:05:46, "Daniel Tryba" <d.tryba at pocos.nl> 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="busymessage at asterisk.example.org";
>  route(RELAY);
>  exit;
>}
>
>_______________________________________________
>Kamailio (SER) - Users Mailing List
>sr-users at lists.kamailio.org
>https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users




 

_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users at lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users





 

_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users at lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users





 


_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users at lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users



-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.kamailio.org/pipermail/sr-users/attachments/20170922/add7c259/attachment.html>