[SR-Users] how to play ring tune when callee declines
Jurijs Ivolga
jurijs.ivolga at gmail.com
Fri Sep 22 09:31:51 CEST 2017
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number" expression="^offline$">
<action application="playback" data="/usr/local/freeswitch/so
unds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:
> Hi,
>
> So, problem is not related to record route but to config of freeswitch.
>
> Not sure what you wrote in mail above, but you need to add code what
> provided Sergey to:
>
> /usr/local/freeswitch/conf/dialplan/default.xml
>
> With kind regards,
>
> Jurijs
>
> On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010 at 163.com> wrote:
>
>> Hello,
>> Thanks for the heads up. The siptrace does help.
>> Now the FS returns(with or without record_route();):
>> SIP/2.0 480 Temporarily Unavailable
>> Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
>>
>> I have generate offline.xml under conf/directory/default. Where did i
>> miss?
>>
>> Thanks
>>
>>
>>
>>
>>
>> At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga at gmail.com> wrote:
>>
>> Hi,
>>
>> Sip trace from Freeswitch will help, but I think you need to insert
>> Record-Route, try in following way:
>>
>> if ($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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>>>>> sr-users at lists.kamailio.org
>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>
>>>>
>>>>
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>>
>>
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