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

[Jurijs Ivolga]

Hi,

So, problem is not related to record route but to config of freeswitch.

Not sure what you wrote in mail above, but you need to add code what
provided Sergey to:

/usr/local/freeswitch/conf/dialplan/default.xml

With kind regards,

Jurijs

On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010 at 163.com> wrote:

> 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/so
>> unds/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
>>>>
>>>>
>>>>
>>>>
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>>>>
>>>
>>>
>>>
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>>
>>
>>
>>
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>
>
>
>
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