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 <zhaoguojie2...@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.ivo...@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) > + ":" + > $sel(cfg_get.voicemail.srv_port); > route(RELAY); > exit; > } > } > > With kind regards, > > Jurijs > > On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2...@163.com> wrote: > >> Hello >> I added below code to let kamailio route invite to freeswitch: >> if ($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.safa...@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/3...@example.org"/> >> <action application="playback" data="ivr/ivr-user_busy.wav"/> >> </condition> >> </extension> >> >> >> ср, 20 сент. 2017 г. в 10:14, 赵国杰 <zhaoguojie2...@163.com>: >> >>> Hello Sergey, >>> I installed freeswitch, what should i do next? >>> >>> >>> >>> >>> >>> At 2017-09-19 12:07:23, "Sergey Safarov" <s.safa...@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 <zhaoguojie2...@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.tr...@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="busymess...@asterisk.example.org"; >>>> > route(RELAY); >>>> > exit; >>>> >} >>>> > >>>> >_______________________________________________ >>>> >Kamailio (SER) - Users Mailing List >>>> >sr-users@lists.kamailio.org >>>> >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Kamailio (SER) - Users Mailing List >>>> sr-users@lists.kamailio.org >>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>>> >>> >>> >>> >>> _______________________________________________ >>> Kamailio (SER) - Users Mailing List >>> sr-users@lists.kamailio.org >>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >>> >> >> >> >> >> _______________________________________________ >> Kamailio (SER) - Users Mailing List >> sr-users@lists.kamailio.org >> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users >> >> > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users@lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > >
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