[SR-Users] how to play ring tune when callee declines
Jurijs Ivolga
jurijs.ivolga at gmail.com
Fri Sep 22 10:00:49 CEST 2017
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="
> INCOMPATIBLE_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
>>>>>
>>>>
>>>>
>>>>
>>>>
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>>>> 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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