29 Mar
2013
29 Mar
'13
1:41 a.m.
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket support, and
the users can register using the JsSIP Tryit sample WebRTC application.
They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over
websockets module.
However, after registration, the users can't place an audio call. I see no
ringing on the remote browser. Can anyone help with clues or debug? In
Debug log I can see the websocket ws_frame.c decode the websocket message
into SIP, and I see normal SIP call flow for an INVITE. However, nothing
indicating a call.
I ran 'ngrep -p -w -W byline port 8888' (WS port) and see that I'm getting
an error response to browser UA of "405: Method Not Allowed". I've
isolated it down to the this snippet in the kamailio.cfg for
route[LOCATION]:
$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "TEST: Method Not
Allowed");
exit;
}
}
The switch case is returning -2, for some reason.
Any help in debugging this appreciated.
29 Mar
29 Mar
4:26 a.m.
New subject: Help with SIP over Websocket audio call: 405 Method Not Allowed
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the "Allow:" header in the
REGISTER.
You can check this by looking at the REGISTER request in a trace and by
inspecting the location records stored in Kamailio (use the ul.dump
command in kamctl for this).
You can disable method filtering in the Kamailio registrar module by
ensuring that the "method_filtering" modparam is set to 0 (or just not set
at all as disabled is the default). Doing this should prevent lookup()
ever returning -2.
Regards,
Peter
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket support,
and
the users can register using the JsSIP Tryit sample WebRTC application.
They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over
websockets module.
However, after registration, the users can't place an audio call. I see
no
ringing on the remote browser. Can anyone help with clues or debug? In
Debug log I can see the websocket ws_frame.c decode the websocket message
into SIP, and I see normal SIP call flow for an INVITE. However, nothing
indicating a call.
I ran 'ngrep -p -w -W byline port 8888' (WS port) and see that I'm getting
an error response to browser UA of "405: Method Not Allowed". I've
isolated it down to the this snippet in the kamailio.cfg for
route[LOCATION]:
$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "TEST: Method Not
Allowed");
exit;
}
}
The switch case is returning -2, for some reason.
Any help in debugging this appreciated.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
10:20 p.m.
New subject: Help with SIP over Websocket audio call: 405 Method Not Allowed
Peter,
Thank you. By changing the "method_filtering" modparam to 0 (it was
actually 1), I am now able to make it past this, and the INVITE is
processed over WS transport. However, the audio call is still not
completing.
I am seeing a "180 Ringing" message for a while, followed by a "408
Request
Timeout". Nothing is showing ringing on the remote browser with JsSIP
tryit.
The only clues I can see in /var/log/syslog hae to do with Accounting DB.
I am using MySQL.
Note that I can do SIP User Agent client calling just fine between these
two same users, and using the JsSIP Tryit app I can also do 'chat'
messaging. Just can't do audio call. Here is /var/log/syslog:
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: NOTICE: acc
[acc.c:275]: ACC: call missed:
timestamp=1364588167;method=INVITE;from_tag=v71r89q4si;to_tag=mnspl1i563;call_id=8rcjevfvgid74ep1s8rc;code=408;reason=Request
Timeout;src_user=brad;src_domain=xxx.net
;src_ip=172.10.200.149;dst_ouser=joe;dst_user=tmgcpvap;dst_domain=7dq4kria04ks.invalid
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: db_mysql
[km_dbase.c:122]: driver error on query: Unknown column 'src_user' in
'field list'
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: <core>
[db_query.c:235]: error while submitting query
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: acc [acc.c:404]:
failed to insert into database
On Thu, Mar 28, 2013 at 9:26 PM, Peter Dunkley <
peter.dunkley(a)crocodile-rcs.com> wrote:
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the "Allow:" header in the
REGISTER.
You can check this by looking at the REGISTER request in a trace and by
inspecting the location records stored in Kamailio (use the ul.dump
command in kamctl for this).
You can disable method filtering in the Kamailio registrar module by
ensuring that the "method_filtering" modparam is set to 0 (or just not set
at all as disabled is the default). Doing this should prevent lookup()
ever returning -2.
Regards,
Peter
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket support,
and
the users can register using the JsSIP Tryit sample WebRTC application.
They can do 'chat' feature of JsSIP Tryit using kamailio 4.0 sip over
websockets module.
However, after registration, the users can't place an audio call. I see
no
ringing on the remote browser. Can anyone help with clues or debug? In
Debug log I can see the websocket ws_frame.c decode the websocket message
into SIP, and I see normal SIP call flow for an INVITE. However, nothing
indicating a call.
I ran 'ngrep -p -w -W byline port 8888' (WS port) and see that I'm
getting
an error response to browser UA of "405:
Method Not Allowed". I've
isolated it down to the this snippet in the kamailio.cfg for
route[LOCATION]:
$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "TEST: Method Not
Allowed");
exit;
}
}
The switch case is returning -2, for some reason.
Any help in debugging this appreciated.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
30 Mar
30 Mar
12:03 a.m.
New subject: Help with SIP over Websocket audio call: 405 Method Not Allowed
Hi,
There is a much simpler WebSocket Kamailio configuration file in the
examples directory in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples…
It doesn't have accounting or any of the other advanced stuff from the
example you provided a link to.
If you can get calls to work with the simple configuration then you know
your problems aren't related to WebSockets, and instead are related to
something else within the configuration.
Regards,
Peter
Peter,
Thank you. By changing the "method_filtering" modparam to 0 (it was
actually 1), I am now able to make it past this, and the INVITE is
processed over WS transport. However, the audio call is still not
completing.
I am seeing a "180 Ringing" message for a while, followed by a "408
Request
Timeout". Nothing is showing ringing on the remote browser with JsSIP
tryit.
The only clues I can see in /var/log/syslog hae to do with Accounting DB.
I am using MySQL.
Note that I can do SIP User Agent client calling just fine between these
two same users, and using the JsSIP Tryit app I can also do 'chat'
messaging. Just can't do audio call. Here is /var/log/syslog:
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: NOTICE: acc
[acc.c:275]: ACC: call missed:
timestamp=1364588167;method=INVITE;from_tag=v71r89q4si;to_tag=mnspl1i563;call_id=8rcjevfvgid74ep1s8rc;code=408;reason=Request
Timeout;src_user=brad;src_domain=xxx.net
;src_ip=172.10.200.149;dst_ouser=joe;dst_user=tmgcpvap;dst_domain=7dq4kria04ks.invalid
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: db_mysql
[km_dbase.c:122]: driver error on query: Unknown column 'src_user' in
'field list'
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: <core>
[db_query.c:235]: error while submitting query
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: acc
[acc.c:404]:
failed to insert into database
On Thu, Mar 28, 2013 at 9:26 PM, Peter Dunkley <
peter.dunkley(a)crocodile-rcs.com> wrote:
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the "Allow:" header in the
REGISTER.
You can check this by looking at the REGISTER request in a trace and by
inspecting the location records stored in Kamailio (use the ul.dump
command in kamctl for this).
You can disable method filtering in the Kamailio registrar module by
ensuring that the "method_filtering" modparam is set to 0 (or just not
set
at all as disabled is the default). Doing this should prevent lookup()
ever returning -2.
Regards,
Peter
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket
support,
and
the users can register using the JsSIP Tryit sample WebRTC
application.
They can do 'chat' feature of JsSIP
Tryit using kamailio 4.0 sip over
websockets module.
However, after registration, the users can't place an audio call. I
see
no
ringing on the remote browser. Can anyone help with clues or debug?
In
Debug log I can see the websocket ws_frame.c
decode the websocket
message
into SIP, and I see normal SIP call flow for an
INVITE. However,
nothing
indicating a call.
I ran 'ngrep -p -w -W byline port 8888' (WS port) and see that I'm
getting
an error response to browser UA of "405:
Method Not Allowed". I've
isolated it down to the this snippet in the kamailio.cfg for
route[LOCATION]:
$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "TEST: Method Not
Allowed");
exit;
}
}
The switch case is returning -2, for some reason.
Any help in debugging this appreciated.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
12:05 a.m.
New subject: Help with SIP over Websocket audio call: 405 Method Not Allowed
Sorry,
Wrong link. The correct one is:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples…
Peter
Hi,
There is a much simpler WebSocket Kamailio configuration file in the
examples directory in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples…
It doesn't have accounting or any of the other advanced stuff from the
example you provided a link to.
If you can get calls to work with the simple configuration then you know
your problems aren't related to WebSockets, and instead are related to
something else within the configuration.
Regards,
Peter
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
Peter,
Thank you. By changing the "method_filtering" modparam to 0 (it was
actually 1), I am now able to make it past this, and the INVITE is
processed over WS transport. However, the audio call is still not
completing.
I am seeing a "180 Ringing" message for a while, followed by a "408
Request
Timeout". Nothing is showing ringing on the remote browser with JsSIP
tryit.
The only clues I can see in /var/log/syslog hae to do with Accounting
DB.
I am using MySQL.
Note that I can do SIP User Agent client calling just fine between these
two same users, and using the JsSIP Tryit app I can also do 'chat'
messaging. Just can't do audio call. Here is /var/log/syslog:
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: NOTICE: acc
[acc.c:275]: ACC: call missed:
timestamp=1364588167;method=INVITE;from_tag=v71r89q4si;to_tag=mnspl1i563;call_id=8rcjevfvgid74ep1s8rc;code=408;reason=Request
Timeout;src_user=brad;src_domain=xxx.net
;src_ip=172.10.200.149;dst_ouser=joe;dst_user=tmgcpvap;dst_domain=7dq4kria04ks.invalid
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: db_mysql
[km_dbase.c:122]: driver error on query: Unknown column 'src_user' in
'field list'
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: <core>
[db_query.c:235]: error while submitting query
Mar 29 20:16:07 ace /usr/local/sbin/kamailio[5928]: ERROR: acc
[acc.c:404]:
failed to insert into database
On Thu, Mar 28, 2013 at 9:26 PM, Peter Dunkley <
peter.dunkley(a)crocodile-rcs.com> wrote:
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the "Allow:" header in the
REGISTER.
You can check this by looking at the REGISTER request in a trace and by
inspecting the location records stored in Kamailio (use the ul.dump
command in kamctl for this).
You can disable method filtering in the Kamailio registrar module by
ensuring that the "method_filtering" modparam is set to 0 (or just not
set
at all as disabled is the default). Doing this should prevent lookup()
ever returning -2.
Regards,
Peter
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Hi,
New to Kamailio. I have my Kamailio 4.0 server with websocket
support,
and
the users can register using the JsSIP Tryit sample WebRTC
application.
They can do 'chat' feature of JsSIP
Tryit using kamailio 4.0 sip
over
websockets module.
However, after registration, the users can't place an audio call. I
see
no
ringing on the remote browser. Can anyone help with clues or debug?
In
Debug log I can see the websocket ws_frame.c
decode the websocket
message
into SIP, and I see normal SIP call flow for an
INVITE. However,
nothing
indicating a call.
I ran 'ngrep -p -w -W byline port 8888' (WS port) and see that I'm
getting
an error response to browser UA of "405:
Method Not Allowed". I've
isolated it down to the this snippet in the kamailio.cfg for
route[LOCATION]:
$avp(oexten) = $rU;
if (!lookup("location")) {
$var(rc) = $rc;
route(TOVOICEMAIL);
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "TEST: Method Not
Allowed");
exit;
}
}
The switch case is returning -2, for some reason.
Any help in debugging this appreciated.
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
10:09 a.m.
New subject: Help with SIP over Websocket audio call: 405 Method Not Allowed
Peter Dunkley writes:
There is a much simpler WebSocket Kamailio
configuration file in the
examples directory in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples…
the link to websocket example is this:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples…
my config is based on that but also handles gruu, which is supported by
tryit client.
-- juha
4255
days inactive
4257
days old
5 comments
3 participants
participants (3)
-
Brad Johns -
Juha Heinanen -
Peter Dunkley