Hi All
I am having a sip client, I am using GNU tls for transport layer security.
I am using Kamailio (TLS enabled) to test all the functionalities.
I did registration over TLS. It was fine.
I tried to make call.
I was able to send INVITE.
Kamaiio trying to connect to called party (which is again my sip endpoint).
That time certificate verification failed.
*Error log*
* 8(14180) ERROR: tls [tls_server.c:1193]: tls_read_f(): TLS
write:error:14090086:SSL routines:ssl3_get_server_certificate:certificate
verify failed 8(14180) ERROR: <core> [tcp_read.c:1326]: tcp_read_req():
ERROR: tcp_read_req: error reading 8(14180) DEBUG: <core>
[tcp_read.c:1481]: release_tcpconn(): releasing con 0x7f962f6b6db8, state
-2, fd=5, id=3*
How do I debug and fix this issue.
Can somebody help me out.
Thanks
Austin
Hello,
Is there an API that can notify Kamailio of media inactivity in the RTPProxy?
The documentation has something called "timeout_socket" . I don't understand how to use it though. Any help will be greatly appreciated.
Thanks
Sharath
Hello.
When I installed kamailio I add users then I can make a call.But When I install asterisk I found the asterisk take over the kamailio and I can register users.But When I make a call.It shows me can not establish the call.How can find the problem?Any logs?Or tools?
Thanks,
zhangkai
Hi,
I am using Kamailio(4.2.5) as my SIP proxy also enabled authentication.
Added two users xyz_0 and xyz_1 to the kamailio database.
While Registration I am getting 401 Unauthorized as shown below
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 192.168.1.107:48601;branch=z9hG4bK858339335
From: <sip:xyz_0@192.168.1.107>;tag=251958751
To: <sip:xyz_0@192.168.1.107 <sip%3Axyz(a)192.168.1.107>
>;tag=ca5a1db24813f1937965921e320b98e4.6232
Call-ID: 1808202998(a)192.168.1.107
CSeq: 2 REGISTER
WWW-Authenticate: Digest realm="192.168.1.107",
nonce="VZJ3hlWSdlr0HS/jGQljEKmY20ZrO0Qz"
Server: kamailio (4.2.5 (x86_64/linux))
Content-Length: 0
The WWW-Authenticate header is not complete here, May I know the reason for
this ?
I am expecting algorithm, domain and other stuff in the WWW-Authenticate
header.
I am not able to proceed further , Please do needful.
Thanks
hello,
is there REST API for kamailio for basic functions like subscribers,
dispatcher, blacklist/whitelist, ACL, dial plan table, ... ?
thank you
--
---------------------------------------
Marek Cervenka
=======================================
I try to send keepalive requests (options) to clients at usrloc
I tried 4 mechanisms
1) nathelper
modparam("nathelper", "sipping_method", "OPTIONS")
modparam("nathelper", "natping_interval", 15)
modparam("nathelper", "ping_nated_only", 0)
modparam("nathelper", "sipping_bflag", 7)
modparam("nathelper", "sipping_from", "sip:pinger@mydomain")
but it didn't send keepalives to noone client
2)nat_traversal
I tried to trigger keepalives with nat_keepalive() when REGISTER coming to
my service
It works fine with UDP packets
modparam("nat_traversal", "keepalive_interval", 15)
modparam("nat_traversal", "keepalive_method", "OPTIONS")
modparam("nat_traversal", "keepalive_from", "sip:pinger@mydomain")
modparam("nat_traversal", "keepalive_state_file",
"/var/run/keepalive_state")
3)So I also tried to do it with WebSocket keepalive mechanism, but it still
not work for (off course I checked only WS devices)
modparam("websocket", "keepalive_mechanism", 15)
modparam("websocket", "keepalive_timeout", 15)
modparam("websocket", "keepalive_interval", 15)
4) I tried to do it with tcpops
Only for ws and tcp (WS is use TCP as i know)
tcp_keepalive_enable("15", "5", "15");
same result...
So I need mechanism to check all devices for keepalive. I think nathelper
works with all protos (As I read at docs)
Can somebody explain to me what I doing wrong?
hello,
is there REST API for kamailio for basic functions like subscribers,
dispatcher, blacklist/whitelist, ACL, dial plan table, ... ?
thank you
--
---------------------------------------
Marek Cervenka
=======================================
Hello,
after some trying with SEMS and Blox as SBC, I changed my idea of SIP infrastructure from
Kamailio (private) -SBC in DMZ (Private & Public) - www (Public)
to
Kamailio (private) - Kamailio with rtpproxy in DMZ (Private & Public) - www (Public).
Now, my questions are: How can I connect 2 Kamailio SIP server together? Which modules I need? Is there somewhere a good howto?
Cheers,
Kai Ohnacker
I was testing a scenario whith parallel forking (multiple registers) where 2
endpoints answer near simultaniously. When phone that was just a bit slower
hangs up, the BYE gets forwarded to the caller and thus the connection is
terminated. This is as far as I know for stateful transactions (t_relay())
I always assumed in this scenario the state machine in kamailio would prevent
this kind of behavior by itself. Is there a way to prevent these messages
being forwarded?
