Jurijs

On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga@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@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@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@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@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@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@163.com>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov@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@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@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@asterisk.example.org";
>  route(RELAY);
>  exit;
>}
>
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