Hi guys, 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?
Best Regards Jesse
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; }
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; }
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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; }
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
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; }
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
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; $ru="busymessage@asterisk.example.org"; route(RELAY); exit; }
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello I added below code to let kamailio route invite to freeswitch: if ($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; $ru="busymessage@asterisk.example.org"; route(RELAY); exit; }
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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; $ru="busymessage@asterisk.example.org"; route(RELAY); exit; }
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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; $ru="busymessage@asterisk.example.org"; route(RELAY); exit; }
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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; $ru="busymessage@asterisk.example.org"; route(RELAY); exit; }
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <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; $ru="busymessage@asterisk.example.org"; route(RELAY); exit; }
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Hi guy. sorry for the confusion. I'll try to reorganize it.
In kamailio.cfg I added if ($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@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@35.202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga jurijs.ivolga@gmail.com wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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; }
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
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@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@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@35. 202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/so unds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga jurijs.ivolga@gmail.com wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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; >} > >_______________________________________________ >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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Hello, I did remove record_route() and bridge. the FS returns 488 instead of 480. Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION"
Thanks
At 2017-09-22 16:00:49, "Jurijs Ivolga" jurijs.ivolga@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@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@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@35.202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga jurijs.ivolga@gmail.com wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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; $ru="busymessage@asterisk.example.org"; route(RELAY); exit; }
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hi,
You need to answer call too...
Try this:
* in freeswitch/conf/dialplan/default.xml* <extension name="prompt-offline"> <condition field="destination_number" expression="^prompt-(.+)$">
<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@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@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@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@35. 202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/so unds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga jurijs.ivolga@gmail.com wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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; > >} > > > >_______________________________________________ > >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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Hello, No luck. Still the same. Here goes the full log, sorry if it's a little overwhelming
------------------------------------------------------------------------ INVITE sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70 Contact: sip:13112345678@175.100.202.254:33189;transport=TLS;ob;alias=175.100.202.254~33189~3 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_HWNXT-24/r2457 Content-Type: application/sdp Content-Length: 515
v=0 o=- 3715057398 3715057398 IN IP4 35.185.130.154 s=pjmedia c=IN IP4 35.185.130.154 t=0 0 m=audio 40026 RTP/AVP 9 8 0 106 101 c=IN IP4 35.185.130.154 a=rtcp:40027 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:106 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BrFCcbuKqPea6vy8L9Imh6dqhorYovx1RdXKlLsP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:9iwM/BpOGlSBK115waMNkpamPBj6prelcsjywL+M a=nortpproxy:yes ------------------------------------------------------------------------ send 664 bytes to udp/[10.240.0.90]:5060 at 08:23:19.816105: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ca9207aa32~64bit Content-Length: 0
------------------------------------------------------------------------ 2017-09-22 08:23:19.787973 [NOTICE] switch_channel.c:1077 New Channel sofia/internal/13112345678@35.202.167.70 [df38887c-8832-42f5-828d-ac89eb6ccf78] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context public 2017-09-22 08:23:19.787973 [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/13112345678@35.202.167.70 to XML[prompt-1000@default] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context default 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Once changed type 'reloadxml' at the console. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:29.847976 [ERR] switch_core_media.c:3547 a=crypto in RTP/AVP, refer to rfc3711 2017-09-22 08:23:29.847976 [NOTICE] switch_channel.c:3752 Hangup sofia/internal/13112345678@35.202.167.70 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] send 1081 bytes to udp/[10.240.0.90]:5060 at 08:23:29.858628: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Max-Forwards: 68 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ca9207aa32~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "prompt-1000" sip:prompt-1000@35.202.167.70;party=calling;privacy=off;screen=no
------------------------------------------------------------------------ 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1642 Session 1 (sofia/internal/13112345678@35.202.167.70) Ended 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel sofia/internal/13112345678@35.202.167.70 [CS_DESTROY] recv 365 bytes from udp/[10.240.0.90]:5060 at 08:23:29.859597: ------------------------------------------------------------------------ ACK sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 ACK Content-Length: 0
------------------------------------------------------------------------
At 2017-09-22 16:14:37, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
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@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@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@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@35.202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga jurijs.ivolga@gmail.com wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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; }
Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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_______________________________________________ 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
Hi,
1) You need to change default password!!!!!!!!!!!! *"Open /usr/local/freeswitch/conf/**vars.xml and change the default_password."*
2) You are calling into Freeswitch with encryption on and probably of this your call is failing, maybe you can try first to try without SRTP and if it works, then you can try to make it work with SRTP
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:25 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, No luck. Still the same. Here goes the full log, sorry if it's a little overwhelming
INVITE sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4. 2c5c86a459371d838623651e8f5b6984.0;i=1 Via: SIP/2.0/TLS 10.60.208.121:43603;received= 175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U 0w0JRcTLD9Y;alias Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag= MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70 Contact: sip:13112345678@175.100.202.254:33189;transport=TLS;ob; alias=175.100.202.254~33189~3 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_HWNXT-24/r2457 Content-Type: application/sdp Content-Length: 515
v=0 o=- 3715057398 3715057398 IN IP4 35.185.130.154 s=pjmedia c=IN IP4 35.185.130.154 t=0 0 m=audio 40026 RTP/AVP 9 8 0 106 101 c=IN IP4 35.185.130.154 a=rtcp:40027 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:106 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline: BrFCcbuKqPea6vy8L9Imh6dqhorYovx1RdXKlLsP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:9iwM/ BpOGlSBK115waMNkpamPBj6prelcsjywL+M a=nortpproxy:yes
send 664 bytes to udp/[10.240.0.90]:5060 at 08:23:19.816105:
SIP/2.0 100 Trying Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4. 2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received= 175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U 0w0JRcTLD9Y;alias Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes From: sip:13112345678@35.202.167.70;tag= MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ ca9207aa32~64bit Content-Length: 0
2017-09-22 08:23:19.787973 [NOTICE] switch_channel.c:1077 New Channel sofia/internal/13112345678@35.202.167.70 [df38887c-8832-42f5-828d- ac89eb6ccf78] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context public 2017-09-22 08:23:19.787973 [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/13112345678@35.202.167.70 to XML[prompt-1000@default] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context default 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Once changed type 'reloadxml' at the console. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:29.847976 [ERR] switch_core_media.c:3547 a=crypto in RTP/AVP, refer to rfc3711 2017-09-22 08:23:29.847976 [NOTICE] switch_channel.c:3752 Hangup sofia/internal/13112345678@35.202.167.70 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] send 1081 bytes to udp/[10.240.0.90]:5060 at 08:23:29.858628:
SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4. 2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received= 175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U 0w0JRcTLD9Y;alias Max-Forwards: 68 From: sip:13112345678@35.202.167.70;tag= MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ ca9207aa32~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "prompt-1000" <sip:prompt-1000@35.202.167.70
;party=calling;privacy=off;screen=no
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1642 Session 1 (sofia/internal/13112345678@35.202.167.70) Ended 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel sofia/internal/13112345678@35.202.167.70 [CS_DESTROY] recv 365 bytes from udp/[10.240.0.90]:5060 at 08:23:29.859597:
ACK sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4. 2c5c86a459371d838623651e8f5b6984.0;i=1 Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag= MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 ACK Content-Length: 0
At 2017-09-22 16:14:37, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
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@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@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@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@35. 202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/so unds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga@gmail.com
wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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/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@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; >> >} >> > >> >_______________________________________________ >> >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
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
Hi,
Please check this:
http://lists.freeswitch.org/pipermail/freeswitch-dev/2013-November/006889.ht...
Probably you need to set
rtp_allow_crypto_in_avp=true in vars.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:32 AM, Jurijs Ivolga jurijs.ivolga@gmail.com wrote:
Hi,
- You need to change default password!!!!!!!!!!!!
*"Open /usr/local/freeswitch/conf/**vars.xml and change the default_password."*
- You are calling into Freeswitch with encryption on and probably of this
your call is failing, maybe you can try first to try without SRTP and if it works, then you can try to make it work with SRTP
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:25 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, No luck. Still the same. Here goes the full log, sorry if it's a little overwhelming
INVITE sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Via: SIP/2.0/TLS 10.60.208.121:43603;received=1 75.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380 xv2U0w0JRcTLD9Y;alias Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5 QazXW6BB To: sip:12345@35.202.167.70 Contact: sip:13112345678@175.100.202.254:33189;transport=TLS;ob;alia s=175.100.202.254~33189~3 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_HWNXT-24/r2457 Content-Type: application/sdp Content-Length: 515
v=0 o=- 3715057398 3715057398 IN IP4 35.185.130.154 s=pjmedia c=IN IP4 35.185.130.154 t=0 0 m=audio 40026 RTP/AVP 9 8 0 106 101 c=IN IP4 35.185.130.154 a=rtcp:40027 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:106 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BrFCcbuKqPea6vy8L9Imh6d qhorYovx1RdXKlLsP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:9iwM/BpOGlSBK115waMNkpa mPBj6prelcsjywL+M a=nortpproxy:yes
send 664 bytes to udp/[10.240.0.90]:5060 at 08:23:19.816105:
SIP/2.0 100 Trying Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=1 75.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380 xv2U0w0JRcTLD9Y;alias Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5 QazXW6BB To: sip:12345@35.202.167.70 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+gi t~20160205T175853Z~ca9207aa32~64bit Content-Length: 0
2017-09-22 08:23:19.787973 [NOTICE] switch_channel.c:1077 New Channel sofia/internal/13112345678@35.202.167.70 [df38887c-8832-42f5-828d-ac89e b6ccf78] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context public 2017-09-22 08:23:19.787973 [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/13112345678@35.202.167.70 to XML[prompt-1000@default] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context default 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Once changed type 'reloadxml' at the console. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:29.847976 [ERR] switch_core_media.c:3547 a=crypto in RTP/AVP, refer to rfc3711 2017-09-22 08:23:29.847976 [NOTICE] switch_channel.c:3752 Hangup sofia/internal/13112345678@35.202.167.70 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] send 1081 bytes to udp/[10.240.0.90]:5060 at 08:23:29.858628:
SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=1 75.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380 xv2U0w0JRcTLD9Y;alias Max-Forwards: 68 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5 QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+gi t~20160205T175853Z~ca9207aa32~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "prompt-1000" <sip:prompt-1000@35.202.167.70
;party=calling;privacy=off;screen=no
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1642 Session 1 (sofia/internal/13112345678@35.202.167.70) Ended 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel sofia/internal/13112345678@35.202.167.70 [CS_DESTROY] recv 365 bytes from udp/[10.240.0.90]:5060 at 08:23:29.859597:
ACK sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5 QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 ACK Content-Length: 0
At 2017-09-22 16:14:37, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
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@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@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@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@35. 202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/so unds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga < jurijs.ivolga@gmail.com> wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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/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@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; >>> >} >>> > >>> >_______________________________________________ >>> >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
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
Thanks man, I didn't explicitly set srtp in kamailio nor freeswitch, how do i turn it off?
At 2017-09-22 16:32:10, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
1) You need to change default password!!!!!!!!!!!! "Open /usr/local/freeswitch/conf/vars.xml and change the default_password."
2) You are calling into Freeswitch with encryption on and probably of this your call is failing, maybe you can try first to try without SRTP and if it works, then you can try to make it work with SRTP
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:25 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, No luck. Still the same. Here goes the full log, sorry if it's a little overwhelming
------------------------------------------------------------------------ INVITE sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70 Contact: sip:13112345678@175.100.202.254:33189;transport=TLS;ob;alias=175.100.202.254~33189~3 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_HWNXT-24/r2457 Content-Type: application/sdp Content-Length: 515
v=0 o=- 3715057398 3715057398 IN IP4 35.185.130.154 s=pjmedia c=IN IP4 35.185.130.154 t=0 0 m=audio 40026 RTP/AVP 9 8 0 106 101 c=IN IP4 35.185.130.154 a=rtcp:40027 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:106 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BrFCcbuKqPea6vy8L9Imh6dqhorYovx1RdXKlLsP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:9iwM/BpOGlSBK115waMNkpamPBj6prelcsjywL+M a=nortpproxy:yes ------------------------------------------------------------------------ send 664 bytes to udp/[10.240.0.90]:5060 at 08:23:19.816105: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ca9207aa32~64bit Content-Length: 0
------------------------------------------------------------------------ 2017-09-22 08:23:19.787973 [NOTICE] switch_channel.c:1077 New Channel sofia/internal/13112345678@35.202.167.70 [df38887c-8832-42f5-828d-ac89eb6ccf78] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context public 2017-09-22 08:23:19.787973 [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/13112345678@35.202.167.70 to XML[prompt-1000@default] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context default 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Once changed type 'reloadxml' at the console. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:29.847976 [ERR] switch_core_media.c:3547 a=crypto in RTP/AVP, refer to rfc3711 2017-09-22 08:23:29.847976 [NOTICE] switch_channel.c:3752 Hangup sofia/internal/13112345678@35.202.167.70 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] send 1081 bytes to udp/[10.240.0.90]:5060 at 08:23:29.858628: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Max-Forwards: 68 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ca9207aa32~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "prompt-1000" sip:prompt-1000@35.202.167.70;party=calling;privacy=off;screen=no
------------------------------------------------------------------------ 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1642 Session 1 (sofia/internal/13112345678@35.202.167.70) Ended 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel sofia/internal/13112345678@35.202.167.70 [CS_DESTROY] recv 365 bytes from udp/[10.240.0.90]:5060 at 08:23:29.859597: ------------------------------------------------------------------------ ACK sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 ACK Content-Length: 0
------------------------------------------------------------------------
At 2017-09-22 16:14:37, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
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@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@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@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@35.202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga jurijs.ivolga@gmail.com wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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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Hi,
First try to set variable in vars.xml, as I sent if didn't help, you can try to turn encryption off on your CSipSimple
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:43 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Thanks man, I didn't explicitly set srtp in kamailio nor freeswitch, how do i turn it off?
At 2017-09-22 16:32:10, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
- You need to change default password!!!!!!!!!!!!
*"Open /usr/local/freeswitch/conf/**vars.xml and change the default_password."*
- You are calling into Freeswitch with encryption on and probably of this
your call is failing, maybe you can try first to try without SRTP and if it works, then you can try to make it work with SRTP
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:25 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, No luck. Still the same. Here goes the full log, sorry if it's a little overwhelming
INVITE sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Via: SIP/2.0/TLS 10.60.208.121:43603;received=1 75.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380 xv2U0w0JRcTLD9Y;alias Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5 QazXW6BB To: sip:12345@35.202.167.70 Contact: sip:13112345678@175.100.202.254:33189;transport=TLS;ob;alia s=175.100.202.254~33189~3 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_HWNXT-24/r2457 Content-Type: application/sdp Content-Length: 515
v=0 o=- 3715057398 3715057398 IN IP4 35.185.130.154 s=pjmedia c=IN IP4 35.185.130.154 t=0 0 m=audio 40026 RTP/AVP 9 8 0 106 101 c=IN IP4 35.185.130.154 a=rtcp:40027 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:106 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BrFCcbuKqPea6vy8L9Imh6d qhorYovx1RdXKlLsP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:9iwM/BpOGlSBK115waMNkpa mPBj6prelcsjywL+M a=nortpproxy:yes
send 664 bytes to udp/[10.240.0.90]:5060 at 08:23:19.816105:
SIP/2.0 100 Trying Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=1 75.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380 xv2U0w0JRcTLD9Y;alias Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5 QazXW6BB To: sip:12345@35.202.167.70 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+gi t~20160205T175853Z~ca9207aa32~64bit Content-Length: 0
2017-09-22 08:23:19.787973 [NOTICE] switch_channel.c:1077 New Channel sofia/internal/13112345678@35.202.167.70 [df38887c-8832-42f5-828d-ac89e b6ccf78] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context public 2017-09-22 08:23:19.787973 [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/13112345678@35.202.167.70 to XML[prompt-1000@default] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context default 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Once changed type 'reloadxml' at the console. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:29.847976 [ERR] switch_core_media.c:3547 a=crypto in RTP/AVP, refer to rfc3711 2017-09-22 08:23:29.847976 [NOTICE] switch_channel.c:3752 Hangup sofia/internal/13112345678@35.202.167.70 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] send 1081 bytes to udp/[10.240.0.90]:5060 at 08:23:29.858628:
SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=1 75.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380 xv2U0w0JRcTLD9Y;alias Max-Forwards: 68 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5 QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+gi t~20160205T175853Z~ca9207aa32~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "prompt-1000" <sip:prompt-1000@35.202.167.70
;party=calling;privacy=off;screen=no
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1642 Session 1 (sofia/internal/13112345678@35.202.167.70) Ended 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel sofia/internal/13112345678@35.202.167.70 [CS_DESTROY] recv 365 bytes from udp/[10.240.0.90]:5060 at 08:23:29.859597:
ACK sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG 4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5 QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 ACK Content-Length: 0
At 2017-09-22 16:14:37, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
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@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@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@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@35. 202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/so unds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga < jurijs.ivolga@gmail.com> wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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/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@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; >>> >} >>> > >>> >_______________________________________________ >>> >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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Hello Jurijs, It worked. You saved my day. Can't thank you enough
Best Regards, Jesse
At 2017-09-22 16:51:32, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
First try to set variable in vars.xml, as I sent if didn't help, you can try to turn encryption off on your CSipSimple
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:43 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Thanks man, I didn't explicitly set srtp in kamailio nor freeswitch, how do i turn it off?
At 2017-09-22 16:32:10, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
1) You need to change default password!!!!!!!!!!!! "Open /usr/local/freeswitch/conf/vars.xml and change the default_password."
2) You are calling into Freeswitch with encryption on and probably of this your call is failing, maybe you can try first to try without SRTP and if it works, then you can try to make it work with SRTP
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 11:25 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, No luck. Still the same. Here goes the full log, sorry if it's a little overwhelming
------------------------------------------------------------------------ INVITE sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70 Contact: sip:13112345678@175.100.202.254:33189;transport=TLS;ob;alias=175.100.202.254~33189~3 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: CSipSimple_HWNXT-24/r2457 Content-Type: application/sdp Content-Length: 515
v=0 o=- 3715057398 3715057398 IN IP4 35.185.130.154 s=pjmedia c=IN IP4 35.185.130.154 t=0 0 m=audio 40026 RTP/AVP 9 8 0 106 101 c=IN IP4 35.185.130.154 a=rtcp:40027 a=sendrecv a=rtpmap:9 G722/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:106 speex/16000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:BrFCcbuKqPea6vy8L9Imh6dqhorYovx1RdXKlLsP a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:9iwM/BpOGlSBK115waMNkpamPBj6prelcsjywL+M a=nortpproxy:yes ------------------------------------------------------------------------ send 664 bytes to udp/[10.240.0.90]:5060 at 08:23:19.816105: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Record-Route: sip:35.202.167.70:5060;r2=on;lr;nat=yes Record-Route: sip:35.202.167.70:5061;transport=tls;r2=on;lr;nat=yes From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70 Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ca9207aa32~64bit Content-Length: 0
------------------------------------------------------------------------ 2017-09-22 08:23:19.787973 [NOTICE] switch_channel.c:1077 New Channel sofia/internal/13112345678@35.202.167.70 [df38887c-8832-42f5-828d-ac89eb6ccf78] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context public 2017-09-22 08:23:19.787973 [NOTICE] switch_ivr.c:1863 Transfer sofia/internal/13112345678@35.202.167.70 to XML[prompt-1000@default] 2017-09-22 08:23:19.787973 [INFO] mod_dialplan_xml.c:635 Processing 13112345678 <13112345678>->prompt-1000 in context default 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 Once changed type 'reloadxml' at the console. 2017-09-22 08:23:19.787973 [CRIT] mod_dptools.c:1670 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-09-22 08:23:29.847976 [ERR] switch_core_media.c:3547 a=crypto in RTP/AVP, refer to rfc3711 2017-09-22 08:23:29.847976 [NOTICE] switch_channel.c:3752 Hangup sofia/internal/13112345678@35.202.167.70 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] send 1081 bytes to udp/[10.240.0.90]:5060 at 08:23:29.858628: ------------------------------------------------------------------------ SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1;received=10.240.0.90 Via: SIP/2.0/TLS 10.60.208.121:43603;received=175.100.202.254;rport=33189;branch=z9hG4bKPj4dLalct0388uwB380xv2U0w0JRcTLD9Y;alias Max-Forwards: 68 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 INVITE User-Agent: FreeSWITCH-mod_sofia/1.4.26+git~20160205T175853Z~ca9207aa32~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Remote-Party-ID: "prompt-1000" sip:prompt-1000@35.202.167.70;party=calling;privacy=off;screen=no
------------------------------------------------------------------------ 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1642 Session 1 (sofia/internal/13112345678@35.202.167.70) Ended 2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel sofia/internal/13112345678@35.202.167.70 [CS_DESTROY] recv 365 bytes from udp/[10.240.0.90]:5060 at 08:23:29.859597: ------------------------------------------------------------------------ ACK sip:prompt-1000@10.240.0.90:5095 SIP/2.0 Via: SIP/2.0/UDP 35.202.167.70:5060;branch=z9hG4bK04d4.2c5c86a459371d838623651e8f5b6984.0;i=1 Max-Forwards: 69 From: sip:13112345678@35.202.167.70;tag=MW06SkJdOZIiqvT4T9DFn0X5QazXW6BB To: sip:12345@35.202.167.70;tag=3N0c8m5X06NBj Call-ID: 8-vmGoxxMxWlb.xQ-VwWyCQ-xnqxNwVe CSeq: 21643 ACK Content-Length: 0
------------------------------------------------------------------------
At 2017-09-22 16:14:37, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
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@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@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@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@35.202.167.70 to XML[prompt-1000@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@gmail.com wrote:
Hi,
You need to add:
<extension name="prompt-offline"> <condition field="destination_number" expression="^offline$"> <action application="playback" data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/> </condition> </extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga jurijs.ivolga@gmail.com wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 zhaoguojie2010@163.com wrote:
Hello, Thanks for the heads up. The siptrace does help. Now the FS returns(with or without record_route();): SIP/2.0 480 Temporarily Unavailable Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" jurijs.ivolga@gmail.com wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in following way:
if ($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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On Tue, Sep 19, 2017 at 11:26:46AM +0800, 赵国杰 wrote:
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.
Just use asterisk/freeswitch/sems/whatever to playback media and you don't have to touch SDP in kamailio to do this. In order to avoid answering the call you have to generate the audio without a 200 OK, so your option is 183 Progress, which asterisk will do for you with Playback(foo,noanswer).
-
Daniel Tryba -
Jurijs Ivolga -
Sergey Safarov -
赵国杰