18 Sep
2017
18 Sep
'17
3:37 a.m.
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
18 Sep
18 Sep
6:05 a.m.
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(a)asterisk.example.org".org";
route(RELAY);
exit;
}
11:26 p.m.
New subject: how to play ring tune when callee declines
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(a)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(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
19 Sep
19 Sep
12:07 a.m.
New subject: how to play ring tune when callee declines
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(a)163.com>om>:
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(a)pocos.nl> wrote:
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰
wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
20 Sep
20 Sep
3:14 a.m.
New subject: how to play ring tune when callee declines
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
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(a)pocos.nl> wrote:
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
7:32 a.m.
New subject: how to play ring tune when callee declines
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(a)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(a)163.com>om>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
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(a)pocos.nl> wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰
wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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;
$ru="busymessage(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
22 Sep
22 Sep
12:19 a.m.
New subject: how to play ring tune when callee declines
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
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(a)pocos.nl> wrote:
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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;
$ru="busymessage(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users 
2:53 a.m.
New subject: how to play ring tune when callee declines
Hi,
Sip trace from Freeswitch will help, but I think you need to insert
Record-Route, try in following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" + $sel(cfg_get.voicemail.srv_
port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)pocos.nl> wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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;
$ru="busymessage(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
3:11 a.m.
New subject: how to play ring tune when callee declines
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com> wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in
following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
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(a)pocos.nl> wrote:
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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;
$ru="busymessage(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
3:29 a.m.
New subject: how to play ring tune when callee declines
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what
provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i
miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com> wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert
Record-Route, try in following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)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(a)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(a)163.com>om>:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)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;
> $ru="busymessage(a)asterisk.example.org".org";
> route(RELAY);
> exit;
>}
>
>_______________________________________________
>Kamailio (SER) - Users Mailing List
>sr-users(a)lists.kamailio.org
>https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
3:31 a.m.
New subject: how to play ring tune when callee declines
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<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(a)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(a)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(a)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)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)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(a)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(a)163.com>om>:
> 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(a)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;
> > $ru="busymessage(a)asterisk.example.org".org";
> > route(RELAY);
> > exit;
> >}
> >
> >_______________________________________________
> >Kamailio (SER) - Users Mailing List
> >sr-users(a)lists.kamailio.org
> >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
3:54 a.m.
New subject: how to play ring tune when callee declines
Hi guy.
sorry for the confusion. I'll try to reorganize it.
In kamailio.cfg I added
if ($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(a)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(a)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(a)gmail.com> wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback"
data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga(a)gmail.com> wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com> wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in
following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
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(a)pocos.nl> wrote:
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4 a.m.
New subject: how to play ring tune when callee declines
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(a)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(a)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(a)gmail.com> wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback" data="/usr/local/freeswitch/so
unds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga(a)gmail.com>
wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what
provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i
miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com> wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert
Record-Route, try in following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
> Hello Sergey,
> I installed freeswitch, what should i do next?
>
>
>
>
>
> At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
>
>> 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(a)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(a)asterisk.example.org".org";
>> > route(RELAY);
>> > exit;
>> >}
>> >
>> >_______________________________________________
>> >Kamailio (SER) - Users Mailing List
>> >sr-users(a)lists.kamailio.org
>> >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users(a)lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:06 a.m.
New subject: how to play ring tune when callee declines
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(a)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(a)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(a)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(a)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(a)gmail.com> wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback"
data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga(a)gmail.com> wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com> wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in
following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
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(a)pocos.nl> wrote:
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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;
$ru="busymessage(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:14 a.m.
New subject: how to play ring tune when callee declines
Hi,
You need to answer call too...
Try this:
* in freeswitch/conf/dialplan/default.xml*
<extension name="prompt-offline">
<condition field="destination_number"
expression="^prompt-(.+)$">
<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(a)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(a)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(a)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(a)gmail.com> wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback" data="/usr/local/freeswitch/so
unds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000
extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga(a)gmail.com>
wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what
provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did
i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com>
wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert
Record-Route, try in following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
> Hello
> I added below code to let kamailio route invite to freeswitch:
> if ($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(a)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(a)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(a)163.com>om>:
>
>> Hello Sergey,
>> I installed freeswitch, what should i do next?
>>
>>
>>
>>
>>
>> At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
>>
>>> 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(a)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(a)asterisk.example.org".org";
>>> > route(RELAY);
>>> > exit;
>>> >}
>>> >
>>> >_______________________________________________
>>> >Kamailio (SER) - Users Mailing List
>>> >sr-users(a)lists.kamailio.org
>>> >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users(a)lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users(a)lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:25 a.m.
New subject: how to play ring tune when callee declines
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(a)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(a)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(a)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(a)35.202.167.70) Ended
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel
sofia/internal/13112345678(a)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(a)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(a)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(a)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(a)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(a)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(a)gmail.com> wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback"
data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga(a)gmail.com> wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com> wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in
following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
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(a)pocos.nl> wrote:
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:32 a.m.
New subject: how to play ring tune when callee declines
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(a)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(a)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(a)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(a)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(a)35.202.167.70) Ended
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close
Channel sofia/internal/13112345678(a)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(a)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(a)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(a)163.com> wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)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(a)gmail.com> wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback" data="/usr/local/freeswitch/so
unds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000
extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga(a)gmail.com
wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what
provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
> Hello,
> Thanks for the heads up. The siptrace does help.
> Now the FS returns(with or without record_route();):
> SIP/2.0 480 Temporarily Unavailable
> Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
>
> I have generate offline.xml under conf/directory/default. Where did
> i miss?
>
> Thanks
>
>
>
>
>
> At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com>
> wrote:
>
> Hi,
>
> Sip trace from Freeswitch will help, but I think you need to insert
> Record-Route, try in following way:
>
> if ($rU=="12345") {
> if(is_method("INVITE")) {
> record_route();
> $ru = "sip:" + "offline" + "@"
+
> $sel(cfg_get.voicemail.srv_ip)
> + ":" +
> $sel(cfg_get.voicemail.srv_port);
> route(RELAY);
> exit;
> }
> }
>
> With kind regards,
>
> Jurijs
>
> On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
>
>> Hello
>> I added below code to let kamailio route invite to freeswitch:
>> if ($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(a)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(a)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(a)163.com>om>:
>>
>>> Hello Sergey,
>>> I installed freeswitch, what should i do next?
>>>
>>>
>>>
>>>
>>>
>>> At 2017-09-19 12:07:23, "Sergey Safarov"
<s.safarov(a)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(a)163.com>om>:
>>>
>>>> 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(a)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(a)asterisk.example.org".org";
>>>> > route(RELAY);
>>>> > exit;
>>>> >}
>>>> >
>>>> >_______________________________________________
>>>> >Kamailio (SER) - Users Mailing List
>>>> >sr-users(a)lists.kamailio.org
>>>> >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing List
>>>> sr-users(a)lists.kamailio.org
>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users(a)lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users(a)lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:41 a.m.
New subject: how to play ring tune when callee declines
Hi,
Please check this:
http://lists.freeswitch.org/pipermail/freeswitch-dev/2013-November/006889.h…
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(a)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(a)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(a)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(a)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(a)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(a)35.202.167.70) Ended
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close
Channel sofia/internal/13112345678(a)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(a)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(a)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(a)163.com> wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)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(a)gmail.com>
wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback" data="/usr/local/freeswitch/so
unds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000
extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <
jurijs.ivolga(a)gmail.com> wrote:
> Hi,
>
> So, problem is not related to record route but to config of freeswitch.
>
> Not sure what you wrote in mail above, but you need to add code what
> provided Sergey to:
>
> /usr/local/freeswitch/conf/dialplan/default.xml
>
> With kind regards,
>
> Jurijs
>
> On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
>
>> Hello,
>> Thanks for the heads up. The siptrace does help.
>> Now the FS returns(with or without record_route();):
>> SIP/2.0 480 Temporarily Unavailable
>> Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
>>
>> I have generate offline.xml under conf/directory/default. Where
>> did i miss?
>>
>> Thanks
>>
>>
>>
>>
>>
>> At 2017-09-22 14:53:06, "Jurijs Ivolga"
<jurijs.ivolga(a)gmail.com>
>> wrote:
>>
>> Hi,
>>
>> Sip trace from Freeswitch will help, but I think you need to insert
>> Record-Route, try in following way:
>>
>> if ($rU=="12345") {
>> if(is_method("INVITE")) {
>> record_route();
>> $ru = "sip:" + "offline" +
"@" +
>> $sel(cfg_get.voicemail.srv_ip)
>> + ":" +
>> $sel(cfg_get.voicemail.srv_port);
>> route(RELAY);
>> exit;
>> }
>> }
>>
>> With kind regards,
>>
>> Jurijs
>>
>> On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
>>
>>> Hello
>>> I added below code to let kamailio route invite to freeswitch:
>>> if ($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(a)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(a)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(a)163.com>om>:
>>>
>>>> Hello Sergey,
>>>> I installed freeswitch, what should i do next?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> At 2017-09-19 12:07:23, "Sergey Safarov"
<s.safarov(a)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(a)163.com>om>:
>>>>
>>>>> 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(a)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(a)asterisk.example.org".org";
>>>>> > route(RELAY);
>>>>> > exit;
>>>>> >}
>>>>> >
>>>>> >_______________________________________________
>>>>> >Kamailio (SER) - Users Mailing List
>>>>> >sr-users(a)lists.kamailio.org
>>>>> >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Kamailio (SER) - Users Mailing List
>>>>> sr-users(a)lists.kamailio.org
>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing List
>>>> sr-users(a)lists.kamailio.org
>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users(a)lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>>
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users(a)lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:43 a.m.
New subject: how to play ring tune when callee declines
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(a)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(a)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(a)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(a)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(a)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(a)35.202.167.70) Ended
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel
sofia/internal/13112345678(a)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(a)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(a)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(a)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(a)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(a)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(a)gmail.com> wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback"
data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga(a)gmail.com> wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com> wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in
following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
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(a)pocos.nl> wrote:
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:51 a.m.
New subject: how to play ring tune when callee declines
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(a)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(a)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(a)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(a)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(a)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(a)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
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
;party=calling;privacy=off;screen=no
-----------------------------------------------------------
-------------
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1642 Session 1
(sofia/internal/13112345678(a)35.202.167.70) Ended
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close
Channel sofia/internal/13112345678(a)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(a)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(a)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(a)163.com> wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)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(a)gmail.com>
wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback" data="/usr/local/freeswitch/so
unds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000
extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <
jurijs.ivolga(a)gmail.com> wrote:
> Hi,
>
> So, problem is not related to record route but to config of freeswitch.
>
> Not sure what you wrote in mail above, but you need to add code what
> provided Sergey to:
>
> /usr/local/freeswitch/conf/dialplan/default.xml
>
> With kind regards,
>
> Jurijs
>
> On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
>
>> Hello,
>> Thanks for the heads up. The siptrace does help.
>> Now the FS returns(with or without record_route();):
>> SIP/2.0 480 Temporarily Unavailable
>> Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
>>
>> I have generate offline.xml under conf/directory/default. Where
>> did i miss?
>>
>> Thanks
>>
>>
>>
>>
>>
>> At 2017-09-22 14:53:06, "Jurijs Ivolga"
<jurijs.ivolga(a)gmail.com>
>> wrote:
>>
>> Hi,
>>
>> Sip trace from Freeswitch will help, but I think you need to insert
>> Record-Route, try in following way:
>>
>> if ($rU=="12345") {
>> if(is_method("INVITE")) {
>> record_route();
>> $ru = "sip:" + "offline" +
"@" +
>> $sel(cfg_get.voicemail.srv_ip)
>> + ":" +
>> $sel(cfg_get.voicemail.srv_port);
>> route(RELAY);
>> exit;
>> }
>> }
>>
>> With kind regards,
>>
>> Jurijs
>>
>> On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
>>
>>> Hello
>>> I added below code to let kamailio route invite to freeswitch:
>>> if ($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(a)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(a)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(a)163.com>om>:
>>>
>>>> Hello Sergey,
>>>> I installed freeswitch, what should i do next?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> At 2017-09-19 12:07:23, "Sergey Safarov"
<s.safarov(a)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(a)163.com>om>:
>>>>
>>>>> 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(a)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(a)asterisk.example.org".org";
>>>>> > route(RELAY);
>>>>> > exit;
>>>>> >}
>>>>> >
>>>>> >_______________________________________________
>>>>> >Kamailio (SER) - Users Mailing List
>>>>> >sr-users(a)lists.kamailio.org
>>>>> >https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> Kamailio (SER) - Users Mailing List
>>>>> sr-users(a)lists.kamailio.org
>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> Kamailio (SER) - Users Mailing List
>>>> sr-users(a)lists.kamailio.org
>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users(a)lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>>
>>
>>
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users(a)lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
5:23 a.m.
New subject: how to play ring tune when callee declines
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(a)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(a)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(a)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(a)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(a)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(a)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(a)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(a)35.202.167.70) Ended
2017-09-22 08:23:29.847976 [NOTICE] switch_core_session.c:1646 Close Channel
sofia/internal/13112345678(a)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(a)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(a)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(a)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(a)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(a)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(a)gmail.com> wrote:
Hi,
You need to add:
<extension name="prompt-offline">
<condition field="destination_number"
expression="^offline$">
<action application="playback"
data="/usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav"/>
</condition>
</extension>
to conf/dialplan/default.xml
in your code, you had extra line what was sending a call to 1000 extension.
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:29 AM, Jurijs Ivolga <jurijs.ivolga(a)gmail.com> wrote:
Hi,
So, problem is not related to record route but to config of freeswitch.
Not sure what you wrote in mail above, but you need to add code what provided Sergey to:
/usr/local/freeswitch/conf/dialplan/default.xml
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 10:11 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello,
Thanks for the heads up. The siptrace does help.
Now the FS returns(with or without record_route();):
SIP/2.0 480 Temporarily Unavailable
Reason: SIP;cause=606;text="USER_NOT_REGISTERED"
I have generate offline.xml under conf/directory/default. Where did i miss?
Thanks
At 2017-09-22 14:53:06, "Jurijs Ivolga" <jurijs.ivolga(a)gmail.com> wrote:
Hi,
Sip trace from Freeswitch will help, but I think you need to insert Record-Route, try in
following way:
if ($rU=="12345") {
if(is_method("INVITE")) {
record_route();
$ru = "sip:" + "offline" + "@" +
$sel(cfg_get.voicemail.srv_ip)
+ ":" +
$sel(cfg_get.voicemail.srv_port);
route(RELAY);
exit;
}
}
With kind regards,
Jurijs
On Fri, Sep 22, 2017 at 7:19 AM, 赵国杰 <zhaoguojie2010(a)163.com> wrote:
Hello
I added below code to let kamailio route invite to freeswitch:
if ($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(a)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(a)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(a)163.com>om>:
Hello Sergey,
I installed freeswitch, what should i do next?
At 2017-09-19 12:07:23, "Sergey Safarov" <s.safarov(a)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(a)163.com>om>:
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(a)pocos.nl> wrote:
On Mon, Sep 18, 2017 at 03:37:22PM +0800, 赵国杰 wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)asterisk.example.org".org";
route(RELAY);
exit;
}
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
19 Sep
19 Sep
5:13 a.m.
New subject: how to play ring tune when callee declines
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).
2704
days inactive
2708
days old
21 comments
4 participants
participants (4)
-
Daniel Tryba -
Jurijs Ivolga -
Sergey Safarov -
赵国杰