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