[SR-Users] how to play ring tune when callee declines
Jurijs Ivolga
jurijs.ivolga at gmail.com
Fri Sep 22 08:53:06 CEST 2017
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
>>>
>>
>>
>>
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>> sr-users at lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
>
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