Hi All
I have some strange behavior of kamailio with TLS.
I have configured second IP addres on server, added it to tls
listener, and tls.cfg file.
But when I try to connect using
openssl s_client -showcerts -connect 10.1.23.33:5061 -tls1 -state
and
openssl s_client -showcerts -connect 10.1.23.23:5061 -tls1 -state
I see same certificates (sip2 my config samples are bellow)
if I make changes in port number (for ip 10.1.23.33 set port 5091 in
both config parts) - I see correct certificates.
Does anyone have this problem?
Thanks in advance.
----- listen section ----
listen=tls:10.1.23.23:5061
listen=tls:10.1.23.33:5061
----- tls.cfg ------
[server:default]
method = TLSv1+
verify_certificate = no
require_certificate = no
private_key = /etc/kamailio/keys/sip1.key
certificate = /etc/kamailio/keys/sip1.crt
[server:10.1.23.33:5061]
method = TLSv1+
verify_certificate = no
require_certificate = no
private_key = /etc/kamailio/keys/sip1.key
certificate = /etc/kamailio/keys/sip1.crt
[server:10.1.23.23:5061]
method = TLSv1+
verify_certificate = no
require_certificate = no
private_key = /etc/kamailio/keys/sip2.key
certificate = /etc/kamailio/keys/sip2.crt
--
Best regards,
Sergey Basov e-mail: sergey.v.basov(a)gmail.com
Hi!
I have this kind of scenario
Receive an INVITE -> send it to user, but if possible, if I’ll receive
REGISTER with same credentials I’m calling to - drop an INVITE there also,
if call was not answered on first location.
Means create other INVITE branch async on the fly, if possible.
Where at least to look at?
Thanks.
--
Best regards,
Igor
Hello,
I try to compile kamailio in freebsd.
I want to add carrierroute module.
I do :
1) cd /usr/ports/net/kamailio
2) make install
but carrierroute module is not compiled.
Could you help me?
Regards
Abdoul.
Hello,
I try to compile kamailio in freebsd.
I want to add carrierroute module.
I did :
1) cd /usr/ports/net/kamailio/
2) make install
but carrierroute module is not compiled.
Could you help me?
Regards
Abdoul.
I need a sanity check if you don’t mind… I am setting up a stateless proxy for several (separate, not load balanced) Asterisk servers behind Kamailio 4.2 (with domain and dispatcher). All works as expected except when called party hangs up, as Asterisk seems to address the BYE “To” (that is sent to the calling party) to the originating domain, and not the IP of the UA (as it does in other cases). When this happens, stateless Kamailio does not seem to know how to route them, which makes sense. I assume I would need to rewrite the “To” field on BYE message so that it contains the UA’s IP (to match the BYE’s that are routed correctly). The RURI header contains the IP/port of the UA, so I am attempting to use it and here is where I am stuck…
Here is the subst from onreply_route:
if (is_method("BYE")) {
xlog("L_NOTICE", "BYE Detected \n");
subst('/^To:(.*)<sip:(.*)@(.*)>(.*)$/To: <sip:\2@$rd:$rp>\4/ig');
}
Here is the “To”:
To: "Test User 2" <sip:102@zxcv.asdf.qwerty.net>;tag=8AB53900B9646F1DF05330E7FBF846C8
Any obvious glaring mistakes here? Would I then need additional logic to forward the BYE on?
My apologies if this is wordy, or nonsensical...
Hello,
I finally managed to place a call from sipml5 webrtc client to Mercuro IMS
client. The phone rings, and when I answer it sends 200 OK to the sipml5
where as sipml5 send back an ACK message which never passes the originating
PCSCF. The PCSCF says:
8(3640) WARNING: <core> [msg_translator.c:2729]: via_builder(): TCP/TLS
connection (id: 0) for WebSocket could not be found
8(3640) ERROR: <core> [msg_translator.c:1947]:
build_req_buf_from_sip_req(): could not create Via header
8(3640) ERROR: <core> [forward.c:548]: forward_request(): building failed
I doubt that the WebSocket connection is closed, cause when I terminate the
call from Mercuro client a bye request is being sent to the sipml5.
The ACK package:
ACK sip:alice@192.168.0.10:49794;transport=udp SIP/2.
Via: SIP/2.0/WS
df7jal23ls0d.invalid;branch=z9hG4bKvuly7bmxnN4aqM4zZTIS;rport
From: "Bob"<sip:bob@net1.test>;tag=GxzKy1nCMEI1mR0RztrB
To: <sip:alice@net1.test>;tag=18823
Contact: "Bob"<sip:bob@df7jal23ls0d.invalid
;rtcweb-breaker=no;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: 5a500969-d0fa-14d1-7d0e-8605f4356ca6
CSeq: 3887 ACK
Content-Length:
Route: <sip:192.168.0.11:4060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 69
Route: <sip:mo@192.168.0.11:880
;transport=ws;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3>
Route: <sip:mo@192.168.0.11:4060
;r2=on;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.0c3>
Route: <sip:mo@192.168.0.11:6060
;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062>
Route: <sip:mt@192.168.0.11:6060
;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.f062>
Route: <sip:mt@192.168.0.11:4060
;lr=on;ftag=GxzKy1nCMEI1mR0RztrB;did=e82.1c3>
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Have been thinking for quite a while, but couldn't really find a reason why
it wouldn't add the v,a header. A debug 3 level log file is also attached.
Thanks in advance,
Serhat
Hello
I am trying to configure kamailio 4.4.2 with asterisk 13.10.0.
My Os is Debian 7.4
I followed the tutorial :
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb…
The two softwares are installed on my server. So they have the same IP
adress but the port is different like they said in the tutorial. I use too
Mysql to stock users informations
I have 2 databases Kamailio and Asterisk. I use ODBC for communicate
beetwen asterisk server and his database.
However when my sip client (linephone) connects to kamalio, kamailio
doesn't forward information to asterisk. I don't kow the reason. So i can
send message but not call
If someone knows how to configure, please help me.
I join my kamailio configuration file.
Regards.
ahnouxs(a)gmail.com
The core documentation says that in a named onreply_route[], only
provisional replies can be drop()'d. To drop any reply, it is necessary
to use a global onreply_route.
Is there any workaround for this, i.e. so I can drop a 2xx reply from a
specific TM transaction?
The reason is, to know whether to drop it, I need access to either AVPs
or, ideally, dialog variables. Since the global onreply_route is
executed by the core, I presume I won't have access to anything that
persists through TM there.
Thanks!
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Hi,
I am trying to get a simple Edge Proxy - Registrar setup working
behind NAT. Initially I tried various scripts - including the one
mentioned in the Oubound page and Peter Dunkley' example - but got
some errors related to PATH.
Can somebody give me a working script of both proxy and registrar? Any
help would be gladly appreciated.
regards
Ganesh Kumar
--
---
http://www.infinicalls.com
Hi,
If a transaction is currently t_suspended() and has not been
t_continued(), will t_check_trans() still find that transaction for
retransmission-dampening purposes, e.g. if a retransmission of the same
request as the original one is received?
Thanks,
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 (direct) / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
