9 Feb
2018
9 Feb
'18
10:57 p.m.
Hi,
Im new to Kamailio so im sorry if some of my questions will sound a bit
wierd :=)
i have a setup like this: (Kamailio version 5.1)
UA > Kamailio > Freeswitch
Kamailio forwards registration to freeswitch without saving location.
I have configured sip over websockets and im able to register to
Freeswitcha via kamailio.
Also, i use add_path_received() so kamailio adds a path header to register
messages
recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809:
------------------------------------------------------------------------
REGISTER sip:192.168.50.60 SIP/2.0
Via: SIP/2.0/UDP
192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS
192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435
Max-Forwards: 69
To: <sip:1001@192.168.50.60>
From: <sip:1001@192.168.50.60>;tag=vq30modpgd
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Authorization: Digest algorithm=MD5, username="1001",
realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90",
uri="sip:192.168.50.60", response="9b592b874474d4ffe5769f2410baceef",
qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001
Contact: <sip:1001@192.0.2.110
;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, 100rel, outbound
User-Agent: SIP.js/0.7.0 BB
Content-Length: 0
Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6>
------------------------------------------------------------------------
send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS
192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435
From: <sip:1001@192.168.50.60>;tag=vq30modpgd
To: <sip:1001@192.168.50.60>;tag=Spya88DKU9gSQ
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600
Date: Fri, 09 Feb 2018 21:38:07 GMT
User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Content-Length: 0
------------------------------------------------------------------------
Once, registered, im trying to originate a call to the registered endpoint
from freeswitch
this results in an invite like this:
send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549:
------------------------------------------------------------------------
INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0
Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB
Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Max-Forwards: 70
From: "" <sip:1002@192.168.50.65>;tag=tZQ393yprj7BK
To: <sip:1001@192.0.2.110;transport=ws>
Call-ID: 8212b700-8885-1236-08b3-000c293ae94e
CSeq: 118770326 INVITE
Contact: <sip:mod_sofia@192.168.50.65:15060>
User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 891
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:1002@192.168.50.65
;party=calling;screen=yes;privacy=off
Freeswitch appends a route header to the invite specifying kamailio as
route and a received with original uri.
well, here is the issue ... im failing to hit the SIP over websocket
client... Kamailio
my RELAY route is like this
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) {
xlog("L_ALERT", "Its a Branch, go manage that !!!
<$ru>\n");
t_on_branch("MANAGE_BRANCH");
}
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) {
xlog("L_ALERT", "its onreply_route, go manage !!!
<$ru>\n");
t_on_reply("MANAGE_REPLY");
}
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) {
xlog("L_ALERT", "Its failure_route, go manage !!!
<$ru>\n");
t_on_failure("MANAGE_FAILURE");
}
}
$fs = "udp:192.168.50.60:5060";
xlog("L_ALERT", "Trying to rely !!! <$ru>\n");
if (!t_relay()) {
xlog("L_ALERT", "failed to rely !!! <$ru>\n");
sl_reply_error();
}
exit;
}
here is a part of the log:
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core>
[core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced
udp:192.168.50.60:5060, to ws:192.0.2.110:5060)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core>
[core/msg_translator.c:424]: clen_builder(): content-length: 916 (916)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core>
[core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for
WebSocket could not be found
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core>
[core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not
create Via header
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm
[t_fwd.c:476]: prepare_new_uac(): could not build request
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm
[t_fwd.c:1735]: t_forward_nonack(): failure to add branches
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm
[t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm
[t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: ***
cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26
n=xlog
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT: <script>:
failed to rely !!! <sip:1001@192.0.2.110;transport=ws>
is that because i set a $fu before t_relay() ?
any idea how to approach the issue ?
13 Feb
13 Feb
11:48 a.m.
Hello,
you have an explicit set of $fs, which is setting the send socket. You
have to do then when receiving from WS, not when going to WS.
In 5.1, you can try to enable:
-
https://www.kamailio.org/docs/modules/stable/modules/path.html#path.p.enabl…
and then let kamailio do all about send socket (remove the assignments
to $fs).
Cheers,
Daniel
On 09.02.18 22:57, Tihomir Culjaga wrote:
Hi,
Im new to Kamailio so im sorry if some of my questions will sound a
bit wierd :=)
i have a setup like this: (Kamailio version 5.1)
UA > Kamailio > Freeswitch
Kamailio forwards registration to freeswitch without saving location.
I have configured sip over websockets and im able to register to
Freeswitcha via kamailio.
Also, i use add_path_received() so kamailio adds a path header to
register messages
recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809:
------------------------------------------------------------------------
REGISTER sip:192.168.50.60 SIP/2.0
Via: SIP/2.0/UDP
192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS
192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435
Max-Forwards: 69
To: <sip:1001@192.168.50.60 <mailto:sip%3A1001@192.168.50.60>>
From: <sip:1001@192.168.50.60
<mailto:sip%3A1001@192.168.50.60>>;tag=vq30modpgd
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Authorization: Digest algorithm=MD5, username="1001",
realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90",
uri="sip:192.168.50.60",
response="9b592b874474d4ffe5769f2410baceef",
qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001
Contact: <sip:1001@192.0.2.110
<mailto:sip%3A1001@192.0.2.110>;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, 100rel, outbound
User-Agent: SIP.js/0.7.0 BB
Content-Length: 0
Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6>
------------------------------------------------------------------------
send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS
192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435
From: <sip:1001@192.168.50.60
<mailto:sip%3A1001@192.168.50.60>>;tag=vq30modpgd
To: <sip:1001@192.168.50.60
<mailto:sip%3A1001@192.168.50.60>>;tag=Spya88DKU9gSQ
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Contact: <sip:1001@192.0.2.110
<mailto:sip%3A1001@192.0.2.110>;transport=ws>;expires=600
Date: Fri, 09 Feb 2018 21:38:07 GMT
User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Content-Length: 0
------------------------------------------------------------------------
Once, registered, im trying to originate a call to the registered
endpoint from freeswitch
this results in an invite like this:
send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549:
------------------------------------------------------------------------
INVITE sip:1001@192.0.2.110
<mailto:sip%3A1001@192.0.2.110>;transport=ws SIP/2.0
Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB
Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Max-Forwards: 70
From: "" <sip:1002@192.168.50.65
<mailto:sip%3A1002@192.168.50.65>>;tag=tZQ393yprj7BK
To: <sip:1001@192.0.2.110 <mailto:sip%3A1001@192.0.2.110>;transport=ws>
Call-ID: 8212b700-8885-1236-08b3-000c293ae94e
CSeq: 118770326 INVITE
Contact: <sip:mod_sofia@192.168.50.65:15060
<http://sip:mod_sofia@192.168.50.65:15060>>
User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 891
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:1002@192.168.50.65
<mailto:sip%3A1002@192.168.50.65>>;party=calling;screen=yes;privacy=off
Freeswitch appends a route header to the invite specifying kamailio as
route and a received with original uri.
well, here is the issue ... im failing to hit the SIP over websocket
client... Kamailio
my RELAY route is like this
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) {
xlog("L_ALERT", "Its a Branch, go manage that
!!! <$ru>\n");
t_on_branch("MANAGE_BRANCH");
}
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) {
xlog("L_ALERT", "its onreply_route, go manage
!!! <$ru>\n");
t_on_reply("MANAGE_REPLY");
}
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) {
xlog("L_ALERT", "Its failure_route, go manage
!!! <$ru>\n");
t_on_failure("MANAGE_FAILURE");
}
}
$fs = "udp:192.168.50.60:5060 <http://192.168.50.60:5060>";
xlog("L_ALERT", "Trying to rely !!! <$ru>\n");
if (!t_relay()) {
xlog("L_ALERT", "failed to rely !!!
<$ru>\n");
sl_reply_error();
}
exit;
}
here is a part of the log:
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING:
<core> [core/forward.c:231]: get_send_socket2(): protocol/port
mismatch (forced udp:192.168.50.60:5060 <http://192.168.50.60:5060>,
to ws:192.0.2.110:5060 <http://192.0.2.110:5060>)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG:
<core> [core/msg_translator.c:424]: clen_builder(): content-length:
916 (916)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING:
<core> [core/msg_translator.c:2765]: via_builder(): TCP/TLS connection
(id: 0) for WebSocket could not be found
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR:
<core> [core/msg_translator.c:1980]: build_req_buf_from_sip_req():
could not create Via header
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm
[t_fwd.c:476]: prepare_new_uac(): could not build request
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm
[t_fwd.c:1735]: t_forward_nonack(): failure to add branches
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm
[t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm
[t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: ***
cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744
a=26 n=xlog
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]:ALERT:
<script>: failed to rely !!! <sip:1001@192.0.2.110
<mailto:sip%3A1001@192.0.2.110>;transport=ws>
is that because i set a $fu before t_relay() ?
any idea how to approach the issue ?
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
12:05 p.m.
Hi Daniel,
thanks for you e-mail... i will try that approach.
in the main time i exploited the "received" parameter in route header sent
by freeswitch and with a bit of manipulation i managed to rout the call
back to WS client :=)
Im will definitely try to enable r2 and come back with the results.
--- snip ---
route[SIPOUT] {
if (!uri==myself) {
xlog("L_INFO", "ROUTE HEADER value: $hdr(Route)\n");
$var(rcv_regex)=
$(hdr(Route){re.subst,/.*received=(.*)/\1/});
$var(ip)= $(var(rcv_regex){s.select,0,~});
$var(port)= $(var(rcv_regex){s.select,1,~});
$var(proto_number)= $(var(rcv_regex){s.select,2,~});
if ($var(proto_number) == "1") {
$var(trans) = "UDP";
} else if ($var(proto_number) == "6" ) {
$var(trans) = "ws";
} else {
# not interested for now
$var(trans) = "TCP";
}
$var(request_route_regex) = $var(ip) + ":" + $var(port) +
";transport=" +$var(trans);
xlog("L_INFO", "ROUTE HEADER - rcv_regex:
$var(rcv_regex)\n");
xlog("L_INFO", "ROUTE HEADER - request route :
$var(request_route_regex)\n");
append_hf("P-hint: outbound\r\n");
$ru="sip:" + $rU + "@"+$var(request_route_regex);
xlog("SCRIPT: Request uri = $ru\n");
route(RELAY);
}
}
--- snip ---
On 13 February 2018 at 11:48, Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
Hello,
you have an explicit set of $fs, which is setting the send socket. You
have to do then when receiving from WS, not when going to WS.
In 5.1, you can try to enable:
- https://www.kamailio.org/docs/modules/stable/modules/path.
html#path.p.enable_r2
and then let kamailio do all about send socket (remove the assignments to
$fs).
Cheers,
Daniel
On 09.02.18 22:57, Tihomir Culjaga wrote:
Hi,
Im new to Kamailio so im sorry if some of my questions will sound a bit
wierd :=)
i have a setup like this: (Kamailio version 5.1)
UA > Kamailio > Freeswitch
Kamailio forwards registration to freeswitch without saving location.
I have configured sip over websockets and im able to register to
Freeswitcha via kamailio.
Also, i use add_path_received() so kamailio adds a path header to register
messages
recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809:
-----------------------------------------------------------
-------------
REGISTER sip:192.168.50.60 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.
0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;
branch=z9hG4bK6931435
Max-Forwards: 69
To: <sip:1001@192.168.50.60>
From: <sip:1001@192.168.50.60>;tag=vq30modpgd
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Authorization: Digest algorithm=MD5, username="1001",
realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90",
uri=
"sip:192.168.50.60", response="9b592b874474d4ffe5769f2410baceef",
qop=auth, cnonce="sf7ff0fp8tm5", nc=00000001
Contact: <sip:1001@192.0.2.110;transport=ws>;reg-id=1;+sip.
instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, 100rel, outbound
User-Agent: SIP.js/0.7.0 BB
Content-Length: 0
Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6>
-----------------------------------------------------------
-------------
send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444:
-----------------------------------------------------------
-------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.
0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS 192.0.2.110;rport=63296;received=192.168.200.77;
branch=z9hG4bK6931435
From: <sip:1001@192.168.50.60>;tag=vq30modpgd
To: <sip:1001@192.168.50.60>;tag=Spya88DKU9gSQ
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600
Date: Fri, 09 Feb 2018 21:38:07 GMT
User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~
b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Content-Length: 0
-----------------------------------------------------------
-------------
Once, registered, im trying to originate a call to the registered endpoint
from freeswitch
this results in an invite like this:
send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549:
-----------------------------------------------------------
-------------
INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0
Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB
Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Max-Forwards: 70
From: "" <sip:1002@192.168.50.65>;tag=tZQ393yprj7BK
To: <sip:1001@192.0.2.110;transport=ws>
Call-ID: 8212b700-8885-1236-08b3-000c293ae94e
CSeq: 118770326 INVITE
Contact: <sip:mod_sofia@192.168.50.65:15060>
User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~
b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 891
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:1002@192.168.50.65>;party=calling;screen=yes;
privacy=off
Freeswitch appends a route header to the invite specifying kamailio as
route and a received with original uri.
well, here is the issue ... im failing to hit the SIP over websocket
client... Kamailio
my RELAY route is like this
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) {
xlog("L_ALERT", "Its a Branch, go manage that !!!
<$ru>\n");
t_on_branch("MANAGE_BRANCH");
}
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) {
xlog("L_ALERT", "its onreply_route, go manage !!!
<$ru>\n");
t_on_reply("MANAGE_REPLY");
}
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) {
xlog("L_ALERT", "Its failure_route, go manage !!!
<$ru>\n");
t_on_failure("MANAGE_FAILURE");
}
}
$fs = "udp:192.168.50.60:5060";
xlog("L_ALERT", "Trying to rely !!! <$ru>\n");
if (!t_relay()) {
xlog("L_ALERT", "failed to rely !!!
<$ru>\n");
sl_reply_error();
}
exit;
}
here is a part of the log:
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core>
[core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced
udp:192.168.50.60:5060, to ws:192.0.2.110:5060)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core>
[core/msg_translator.c:424]: clen_builder(): content-length: 916 (916)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core>
[core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for
WebSocket could not be found
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core>
[core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not
create Via header
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm
[t_fwd.c:476]: prepare_new_uac(): could not build request
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm
[t_fwd.c:1735]: t_forward_nonack(): failure to add branches
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm
[t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error -2 (-2)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm
[t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: ***
cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26
n=xlog
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT:
<script>: failed to rely !!! <sip:1001@192.0.2.110;transport=ws>
is that because i set a $fu before t_relay() ?
any idea how to approach the issue ?
_______________________________________________
Kamailio (SER) - Users Mailing
Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
12:45 p.m.
Hello,
if you set use_received param for path module, then the received address
in Route header is set as dst uri, see:
-
https://www.kamailio.org/docs/modules/stable/modules/path.html#path.p.enabl…
All you need to do is:
loose_route();
t_relay();
Note that loose_route() is dealing with a pre-loaded Route header in
this case, see the readme of rr module for the return code.
Cheers,
Daniel
On 13.02.18 12:05, Tihomir Culjaga wrote:
Hi Daniel,
thanks for you e-mail... i will try that approach.
in the main time i exploited the "received" parameter in route header
sent by freeswitch and with a bit of manipulation i managed to rout
the call back to WS client :=)
Im will definitely try to enable r2 and come back with the results.
--- snip ---
route[SIPOUT] {
if (!uri==myself) {
xlog("L_INFO", "ROUTE HEADER value: $hdr(Route)\n");
$var(rcv_regex)=
$(hdr(Route){re.subst,/.*received=(.*)/\1/});
$var(ip)= $(var(rcv_regex){s.select,0,~});
$var(port)= $(var(rcv_regex){s.select,1,~});
$var(proto_number)= $(var(rcv_regex){s.select,2,~});
if ($var(proto_number) == "1") {
$var(trans) = "UDP";
} else if ($var(proto_number) == "6" ) {
$var(trans) = "ws";
} else {
# not interested for now
$var(trans) = "TCP";
}
$var(request_route_regex) = $var(ip) + ":" +
$var(port) + ";transport=" +$var(trans);
xlog("L_INFO", "ROUTE HEADER - rcv_regex:
$var(rcv_regex)\n");
xlog("L_INFO", "ROUTE HEADER - request route :
$var(request_route_regex)\n");
append_hf("P-hint: outbound\r\n");
$ru="sip:" + $rU + "@"+$var(request_route_regex);
xlog("SCRIPT: Request uri = $ru\n");
route(RELAY);
}
}
--- snip ---
On 13 February 2018 at 11:48, Daniel-Constantin Mierla
<miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote:
Hello,
you have an explicit set of $fs, which is setting the send socket.
You have to do then when receiving from WS, not when going to WS.
In 5.1, you can try to enable:
-
https://www.kamailio.org/docs/modules/stable/modules/path.html#path.p.enabl…
<https://www.kamailio.org/docs/modules/stable/modules/path.html#path.p.enable_r2>
and then let kamailio do all about send socket (remove the
assignments to $fs).
Cheers,
Daniel
On 09.02.18 22:57, Tihomir Culjaga wrote:
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
Hi,
Im new to Kamailio so im sorry if some of my questions will sound
a bit wierd :=)
i have a setup like this: (Kamailio version 5.1)
UA > Kamailio > Freeswitch
Kamailio forwards registration to freeswitch without saving location.
I have configured sip over websockets and im able to register to
Freeswitcha via kamailio.
Also, i use add_path_received() so kamailio adds a path header to
register messages
recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809:
------------------------------------------------------------------------
REGISTER sip:192.168.50.60 SIP/2.0
Via: SIP/2.0/UDP
192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS
192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435
Max-Forwards: 69
To: <sip:1001@192.168.50.60 <mailto:sip%3A1001@192.168.50.60>>
From: <sip:1001@192.168.50.60
<mailto:sip%3A1001@192.168.50.60>>;tag=vq30modpgd
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Authorization: Digest algorithm=MD5, username="1001",
realm="192.168.50.60",
nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90",
uri="sip:192.168.50.60",
response="9b592b874474d4ffe5769f2410baceef", qop=auth,
cnonce="sf7ff0fp8tm5", nc=00000001
Contact: <sip:1001@192.0.2.110
<mailto:sip%3A1001@192.0.2.110>;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, 100rel, outbound
User-Agent: SIP.js/0.7.0 BB
Content-Length: 0
Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6>
------------------------------------------------------------------------
send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS
192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435
From: <sip:1001@192.168.50.60
<mailto:sip%3A1001@192.168.50.60>>;tag=vq30modpgd
To: <sip:1001@192.168.50.60
<mailto:sip%3A1001@192.168.50.60>>;tag=Spya88DKU9gSQ
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Contact: <sip:1001@192.0.2.110
<mailto:sip%3A1001@192.0.2.110>;transport=ws>;expires=600
Date: Fri, 09 Feb 2018 21:38:07 GMT
User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Content-Length: 0
------------------------------------------------------------------------
Once, registered, im trying to originate a call to the registered
endpoint from freeswitch
this results in an invite like this:
send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549:
------------------------------------------------------------------------
INVITE sip:1001@192.0.2.110
<mailto:sip%3A1001@192.0.2.110>;transport=ws SIP/2.0
Via: SIP/2.0/UDP
192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB
Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Max-Forwards: 70
From: "" <sip:1002@192.168.50.65
<mailto:sip%3A1002@192.168.50.65>>;tag=tZQ393yprj7BK
To: <sip:1001@192.0.2.110
<mailto:sip%3A1001@192.0.2.110>;transport=ws>
Call-ID: 8212b700-8885-1236-08b3-000c293ae94e
CSeq: 118770326 INVITE
Contact: <sip:mod_sofia@192.168.50.65:15060
<http://sip:mod_sofia@192.168.50.65:15060>>
User-Agent:
FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO,
UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 891
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:1002@192.168.50.65
<mailto:sip%3A1002@192.168.50.65>>;party=calling;screen=yes;privacy=off
Freeswitch appends a route header to the invite specifying
kamailio as route and a received with original uri.
well, here is the issue ... im failing to hit the SIP over
websocket client... Kamailio
my RELAY route is like this
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) {
xlog("L_ALERT", "Its a Branch, go manage
that !!! <$ru>\n");
t_on_branch("MANAGE_BRANCH");
}
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) {
xlog("L_ALERT", "its onreply_route, go
manage !!! <$ru>\n");
t_on_reply("MANAGE_REPLY");
}
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) {
xlog("L_ALERT", "Its failure_route, go
manage !!! <$ru>\n");
t_on_failure("MANAGE_FAILURE");
}
}
$fs = "udp:192.168.50.60:5060 <http://192.168.50.60:5060>";
xlog("L_ALERT", "Trying to rely !!! <$ru>\n");
if (!t_relay()) {
xlog("L_ALERT", "failed to rely !!!
<$ru>\n");
sl_reply_error();
}
exit;
}
here is a part of the log:
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]:
WARNING: <core> [core/forward.c:231]: get_send_socket2():
protocol/port mismatch (forced udp:192.168.50.60:5060
<http://192.168.50.60:5060>, to ws:192.0.2.110:5060
<http://192.0.2.110:5060>)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG:
<core> [core/msg_translator.c:424]: clen_builder():
content-length: 916 (916)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]:
WARNING: <core> [core/msg_translator.c:2765]: via_builder():
TCP/TLS connection (id: 0) for WebSocket could not be found
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR:
<core> [core/msg_translator.c:1980]:
build_req_buf_from_sip_req(): could not create Via header
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR:
tm [t_fwd.c:476]: prepare_new_uac(): could not build request
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR:
tm [t_fwd.c:1735]: t_forward_nonack(): failure to add branches
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG:
tm [t_funcs.c:334]: t_relay_to(): t_forward_nonack returned error
-2 (-2)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG:
tm [t_funcs.c:352]: t_relay_to(): -2 error reply generation delayed
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR:
*** cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg]
l=744 a=26 n=xlog
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]:ALERT:
<script>: failed to rely !!! <sip:1001@192.0.2.110
<mailto:sip%3A1001@192.0.2.110>;transport=ws>
is that because i set a $fu before t_relay() ?
any idea how to approach the issue ?
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
<https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>
--
Daniel-Constantin Mierla
www.twitter.com/miconda <http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com
<http://www.asipto.com>
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com
<http://www.kamailioworld.com>
1:42 p.m.
Great!!!
That will do.
Thanks.
Sent from my iPhone
On 13 Feb 2018, at 12:45, Daniel-Constantin Mierla
<miconda(a)gmail.com> wrote:
Hello,
if you set use_received param for path module, then the received address in Route header
is set as dst uri, see:
- https://www.kamailio.org/docs/modules/stable/modules/path.html#path.p.enabl…
All you need to do is:
loose_route();
t_relay();
Note that loose_route() is dealing with a pre-loaded Route header in this case, see the
readme of rr module for the return code.
Cheers,
Daniel
On 13.02.18 12:05, Tihomir Culjaga wrote:
Hi Daniel,
thanks for you e-mail... i will try that approach.
in the main time i exploited the "received" parameter in route header sent by
freeswitch and with a bit of manipulation i managed to rout the call back to WS client
:=)
Im will definitely try to enable r2 and come back with the results.
--- snip ---
route[SIPOUT] {
if (!uri==myself) {
xlog("L_INFO", "ROUTE HEADER value: $hdr(Route)\n");
$var(rcv_regex)= $(hdr(Route){re.subst,/.*received=(.*)/\1/});
$var(ip)= $(var(rcv_regex){s.select,0,~});
$var(port)= $(var(rcv_regex){s.select,1,~});
$var(proto_number)= $(var(rcv_regex){s.select,2,~});
if ($var(proto_number) == "1") {
$var(trans) = "UDP";
} else if ($var(proto_number) == "6" ) {
$var(trans) = "ws";
} else {
# not interested for now
$var(trans) = "TCP";
}
$var(request_route_regex) = $var(ip) + ":" + $var(port) +
";transport=" +$var(trans);
xlog("L_INFO", "ROUTE HEADER - rcv_regex:
$var(rcv_regex)\n");
xlog("L_INFO", "ROUTE HEADER - request route :
$var(request_route_regex)\n");
append_hf("P-hint: outbound\r\n");
$ru="sip:" + $rU + "@"+$var(request_route_regex);
xlog("SCRIPT: Request uri = $ru\n");
route(RELAY);
}
}
--- snip ---
On 13 February 2018 at 11:48, Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com
Kamailio World Conference - May 14-16, 2018 - www.kamailioworld.com Hello,
you have an explicit set of $fs, which is setting the send socket. You have to do then
when receiving from WS, not when going to WS.
In 5.1, you can try to enable:
- https://www.kamailio.org/docs/modules/stable/modules/path.html#path.p.enabl…
and then let kamailio do all about send socket (remove the assignments to $fs).
Cheers,
Daniel
On 09.02.18 22:57, Tihomir Culjaga wrote:
Hi,
Im new to Kamailio so im sorry if some of my questions will sound a bit wierd :=)
i have a setup like this: (Kamailio version 5.1)
UA > Kamailio > Freeswitch
Kamailio forwards registration to freeswitch without saving location.
I have configured sip over websockets and im able to register to Freeswitcha via
kamailio.
Also, i use add_path_received() so kamailio adds a path header to register messages
recv 945 bytes from udp/[192.168.50.60]:5060 at 22:38:07.283809:
------------------------------------------------------------------------
REGISTER sip:192.168.50.60 SIP/2.0
Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS
192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435
Max-Forwards: 69
To: <sip:1001@192.168.50.60>
From: <sip:1001@192.168.50.60>;tag=vq30modpgd
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Authorization: Digest algorithm=MD5, username="1001",
realm="192.168.50.60", nonce="744aaa14-4a86-4ce3-b6d6-419b0a34ee90",
uri="sip:192.168.50.60",
response="9b592b874474d4ffe5769f2410baceef", qop=auth,
cnonce="sf7ff0fp8tm5", nc=00000001
Contact:
<sip:1001@192.0.2.110;transport=ws>;reg-id=1;+sip.instance="<urn:uuid:587f4694-7e55-46dc-a858-579a82556a20>";expires=600
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: path, gruu, 100rel, outbound
User-Agent: SIP.js/0.7.0 BB
Content-Length: 0
Path: <sip:192.168.50.60;lr;received=192.168.200.77~63296~6>
------------------------------------------------------------------------
send 719 bytes to udp/[192.168.50.60]:5060 at 22:38:07.329444:
------------------------------------------------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.50.60;branch=z9hG4bKa929.0f66d60c4333f4a3a9d83d41a63295d9.0
Via: SIP/2.0/WSS
192.0.2.110;rport=63296;received=192.168.200.77;branch=z9hG4bK6931435
From: <sip:1001@192.168.50.60>;tag=vq30modpgd
To: <sip:1001@192.168.50.60>;tag=Spya88DKU9gSQ
Call-ID: dlnrna9o4ngn25fb9d1vi2
CSeq: 196 REGISTER
Contact: <sip:1001@192.0.2.110;transport=ws>;expires=600
Date: Fri, 09 Feb 2018 21:38:07 GMT
User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER,
NOTIFY
Supported: timer, path, replaces
Path: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Content-Length: 0
------------------------------------------------------------------------
Once, registered, im trying to originate a call to the registered endpoint from
freeswitch
this results in an invite like this:
send 1757 bytes to udp/[192.168.50.60]:5060 at 22:46:21.252549:
------------------------------------------------------------------------
INVITE sip:1001@192.0.2.110;transport=ws SIP/2.0
Via: SIP/2.0/UDP 192.168.50.65:15060;rport;branch=z9hG4bKXHcX316pDgXZB
Route: <sip:192.168.50.60>;lr;received=192.168.200.77~63296~6
Max-Forwards: 70
From: "" <sip:1002@192.168.50.65>;tag=tZQ393yprj7BK
To: <sip:1001@192.0.2.110;transport=ws>
Call-ID: 8212b700-8885-1236-08b3-000c293ae94e
CSeq: 118770326 INVITE
Contact: <sip:mod_sofia@192.168.50.65:15060>
User-Agent: FreeSWITCH-mod_sofia/1.6.19+git~20171120T163416Z~b1b21d0695~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER,
NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 891
X-FS-Support: update_display,send_info
Remote-Party-ID: <sip:1002@192.168.50.65>;party=calling;screen=yes;privacy=off
Freeswitch appends a route header to the invite specifying
kamailio as route and a received with original uri.
well, here is the issue ... im failing to hit the SIP over websocket client... Kamailio
my RELAY route is like this
route[RELAY] {
# enable additional event routes for forwarded requests
# - serial forking, RTP relaying handling, a.s.o.
if (is_method("INVITE|BYE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("branch_route")) {
xlog("L_ALERT", "Its a Branch, go manage that !!!
<$ru>\n");
t_on_branch("MANAGE_BRANCH");
}
}
if (is_method("INVITE|SUBSCRIBE|UPDATE")) {
if(!t_is_set("onreply_route")) {
xlog("L_ALERT", "its onreply_route, go manage !!!
<$ru>\n");
t_on_reply("MANAGE_REPLY");
}
}
if (is_method("INVITE")) {
if(!t_is_set("failure_route")) {
xlog("L_ALERT", "Its failure_route, go manage !!!
<$ru>\n");
t_on_failure("MANAGE_FAILURE");
}
}
$fs = "udp:192.168.50.60:5060";
xlog("L_ALERT", "Trying to rely !!! <$ru>\n");
if (!t_relay()) {
xlog("L_ALERT", "failed to rely !!!
<$ru>\n");
sl_reply_error();
}
exit;
}
here is a part of the log:
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core>
[core/forward.c:231]: get_send_socket2(): protocol/port mismatch (forced
udp:192.168.50.60:5060, to ws:192.0.2.110:5060)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: <core>
[core/msg_translator.c:424]: clen_builder(): content-length: 916 (916)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: WARNING: <core>
[core/msg_translator.c:2765]: via_builder(): TCP/TLS connection (id: 0) for WebSocket
could not be found
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: <core>
[core/msg_translator.c:1980]: build_req_buf_from_sip_req(): could not create Via header
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:476]:
prepare_new_uac(): could not build request
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: tm [t_fwd.c:1735]:
t_forward_nonack(): failure to add branches
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:334]:
t_relay_to(): t_forward_nonack returned error -2 (-2)
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: DEBUG: tm [t_funcs.c:352]:
t_relay_to(): -2 error reply generation delayed
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ERROR: ***
cfgtrace:request_route=[RELAY] c=[/etc/kamailio/kamailio.cfg] l=744 a=26 n=xlog
Feb 9 22:48:48 incfskamtest01 /usr/sbin/kamailio[14896]: ALERT: <script>: failed
to rely !!! <sip:1001@192.0.2.110;transport=ws>
is that because i set a $fu before t_relay() ?
any idea how to approach the issue ?
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Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - March 5-7, 2018, Berlin - www.asipto.com
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participants (2)
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Daniel-Constantin Mierla -
Tihomir Culjaga