[SR-Users] how to play ring tune when callee declines
Jurijs Ivolga
jurijs.ivolga at gmail.com
Fri Sep 22 10:14:37 CEST 2017
Hi,
You need to answer call too...
Try this:
* in freeswitch/conf/dialplan/default.xml*
<extension name="prompt-offline">
<condition field="destination_number" expression="^prompt-(.+)$">
<action application="answer"/>
<action application="playback" data="ivr/ivr-user_busy.wav"/>
</condition>
</extension>
Please send full logs next time, you can remove IP-addresses and other
info, but one line is not really helpful.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:00 AM, Jurijs Ivolga <jurijs.ivolga at gmail.com>
wrote:
> Hi,
>
> You probably don't need record route and you need to remove "<action
> application="bridge" data="user/$1@${domain_name}"/>"
>
> Try in this way:
>
> *In kamailio.cfg* I added if ($rU=="12345") {
> if(is_method("INVITE")) {
> #record_route();
> $ru = "sip:prompt-1000@" +
> $sel(cfg_get.voicemail.srv_ip)
> + ":" +
> $sel(cfg_get.voicemail.srv_port);
> route(RELAY);
> exit;
> }
> }
>
> * in freeswitch/conf/dialplan/default.xml*, i added
> <extension name="prompt-offline">
> <condition field="destination_number" expression="^prompt-(.+)$">
> <action application="playback" data="ivr/ivr-user_busy.wav"/>
> </condition>
> </extension>
>
> Jurijs
>
> On Fri, Sep 22, 2017 at 10:54 AM, 赵国杰 <zhaoguojie2010 at 163.com> wrote:
>
>> Hi guy.
>> sorry for the confusion. I'll try to reorganize it.
>>
>> * In kamailio.cfg* I added
>> if ($rU=="12345") {
>> if(is_method("INVITE")) {
>> #record_route();
>> $ru = "sip:prompt-1000@" +
>> $sel(cfg_get.voicemail.srv_ip)
>> + ":" +
>> $sel(cfg_get.voicemail.srv_port);
>> route(RELAY);
>> exit;
>> }
>> }
>>
>> * in freeswitch/conf/dialplan/default.xml*, i added
>> <extension name="prompt-offline">
>> <condition field="destination_number" expression="^prompt-(.+)$">
>> <action application="bridge" data="user/$1@${domain_name}"/>
>> <action application="playback" data="ivr/ivr-user_busy.wav"/>
>> </condition>
>> </extension>
>>
>> *sofia log:*
>> [NOTICE] switch_channel.c:1077 New Channel sofia/internal/
>> 13112345678 at 35.202.167.70 [848d0dd5-0513-41bd-982c-45a2a886e194]
>> [INFO] mod_dialplan_xml.c:635 Processing 13112345678
>> <13112345678>->prompt-1000 in context public
>> [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/13112345678 at 35.
>> 202.167.70 to XML[prompt-1000 at default]
>> [INFO] mod_dialplan_xml.c:635 Processing 13112345678
>> <13112345678>->prompt-1000 in context default
>> [NOTICE] switch_ivr_originate.c:2759 Cannot create outgoing channel of
>> type [error] cause: [USER_NOT_REGISTERED]
>> [NOTICE] switch_ivr_originate.c:2759 Cannot create outgoing channel of
>> type [user] cause: [USER_NOT_REGISTERED]
>> -----------------------------------------------------------
>> -------------
>> SIP/2.0 480 Temporarily Unavailable
>> ......
>> Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
>>
>> -----------------------------------------------------------
>> -------------
>>
>> However, if i delete:
>> <action application="bridge" data="user/$1@${domain_name}"/>,
>> the FS returns 488 instead of 480. Reason: Q.850;cause=88;text="INCOMPATI
>> BLE_DESTINATION"
>>
>> Thanks
>>
>>
>>
>>
>> At 2017-09-22 15:31:51, "Jurijs Ivolga" <jurijs.ivolga at gmail.com> wrote:
>>
>> 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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> 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
>>>>
>>>>
>>>
>>
>>
>>
>>
>> _______________________________________________
>> 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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