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

Jurijs Ivolga jurijs.ivolga at gmail.com
Fri Sep 22 09:29:07 CEST 2017


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
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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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>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
>
>
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