This was it for 2011, final timeout for a fruitful year! Many thanks to
the community for keeping things moving forward!
A happy and prosperous 2012 to all Kamailio and SER friends!!!
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hi all,
Someone know if the kamailio 1.5.0 has problem to create and release tcp
connetctions after some time running? In my SIP network I have six IP
telephones and one gateway registered over TLS, after some time (one day)
this devices have problems to register and make calls. I noticed through
the log a freezing of kamailio when it try release the tcp connections.
Below is part of the log where I notice this behavior:
*Dec 28 09:46:46 vts74 /home2/local/kamailio/sbin/kamailio[8951]:
DBG:core:io_watch_add: io_watch_add(0x816cc80, 25, 2, 0xb61e6558), fd_no=20
Dec 28 09:46:46 vts74 /home2/local/kamailio/sbin/kamailio[8951]:
DBG:core:handle_tcp_child: cmd CONN_RELEASE 0xb61e6558 refcnt= 0
Dec 28 09:47:11 vts74 /home2/local/kamailio/sbin/kamailio[8940]:
DBG:usrloc:nodb_timer: Binding '2503','sips:2503@192.168.174.178;transport=tls'
has expired
Dec 28 09:47:26 vts74 /home2/local/kamailio/sbin/kamailio[8951]:
DBG:core:handle_tcpconn_ev: data available on 0xb621d710 30
Dec 28 09:47:26 vts74 /home2/local/kamailio/sbin/kamailio[8951]:
DBG:core:io_watch_del: io_watch_del (0x816cc80, 30, -1, 0x0) fd_no=21 called
*
I know that upgrade to kamailio 3xx is recommended and I'm working to do
it, but before, I have to understand this behavior.
Cheers
Hi,
What is the Pseudo-variables for "Status-Line" filed of SIP response
messages (2xx, 3xx,4xx,5xx,6xx)? i.e., is there a Pseudo-variables to
display the SIP response code?
Thanks,
R
Hi,
I have two gateways pinged by kamailio. The both are AP (active/probing),
when I cut the one gateway off, it becomes IP(inactive/probing) but the
event_route is not fired up. Am I missing something ?
event_route[dispatcher:dst-down] {
xlog("L_ERR", "Destination down: $rm $ru ($du)\n");
}
Mino
Hi!
Relatively new here, so forgive my probably newbie question..
I understand Kamailio has the Dispatcher-module for load-balancing. OpenSIPS has both a Dispatcher-module and a LB-module, which seems to offer improved functionality. Is there also something like the LB-module in Kamailio?
Cheers,
Ron
I have Kamailio 3.2.0 between two asterisk servers, after the call set, one
of the kamailio send the OK from the INVITE and the return ACK of that
message was discarded. This makes asterisk hangup the call after 5 secs.
It's that right?
OK message:
U 172.25.249.15:5060 -> 172.25.249.14:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 172.25.249.14:5060;branch=z9hG4bK09fc3de6;rport=5060.
From: "asterisk" <sip:asterisk@172.25.249.14>;tag=as6411602a.
To: <sip:775008@172.25.249.15:5060>;tag=as55ab3180.
Call-ID: 547225391b7828402ecaa03e1dab5a86(a)172.25.249.14.
CSeq: 102 INVITE.
Server: Asterisk PBX 1.8.7.1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH.
Supported: replaces, timer.
Contact: <sip:775008@172.25.249.15:5080>.
Remote-Party-ID: "Eus Test" <sip:3999@172.25.249.14
>;party=called;privacy=off;screen=no.
Content-Type: application/sdp.
Content-Length: 285.
.
v=0.
o=root 2045590031 2045590031 IN IP4 172.25.249.15.
s=Asterisk PBX 1.8.7.1.
c=IN IP4 172.25.249.15.
t=0 0.
m=audio 11922 RTP/AVP 0 3 8 101.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
Discarded ACK:
U 172.25.249.14:5060 -> 172.25.249.15:5060
ACK sip:775008@172.25.249.15:5080 SIP/2.0.
Via: SIP/2.0/UDP 172.25.249.14:5060;branch=z9hG4bK6ea5aff6;rport.
From: "asterisk" <sip:asterisk@172.25.249.14>;tag=as6411602a.
To: <sip:775008@172.25.249.15:5060>;tag=as55ab3180.
Contact: <sip:asterisk@172.25.249.14>.
Call-ID: 547225391b7828402ecaa03e1dab5a86(a)172.25.249.14.
CSeq: 102 ACK.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Remote-Party-ID: "asterisk" <sip:asterisk@172.25.249.14>.
Content-Length: 0.
.
Kamailio's configuration where the ACK message it's being discarded:
if ( is_method("ACK") ) {
if ( t_check_trans() ) {
# no loose-route, but stateful ACK;
# must be an ACK after a 487
# or e.g. 404 from upstream server
t_relay();
exit;
} else {
# ACK without matching transaction
... ignore and discard
exit;
}
}
It would be ok if I relay the ack even if it didn't match any transaction??
Any help would be appreciated.
Regards,
Lucas
I want to take the opportunity and give proper credits to everyone
around the project, any bit of contribution makes it evolve better and
better year by year. This includes also the involvement in community, we
don't need only developers to improve. There is no reason to stay aside
in an open source project, become active and play a role in its evolution.
We've just started the second decade of development, but there is always
space to innovate communications. With SER and Kamailio, we pioneered
over the years many solutions in unified communication systems, the
plans are the same, be ahead of the market with new features while
preserving the robustness.
Merry Christmas and great winter holidays to all Kamailians and SERians!!!
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.comhttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hi,
I am Isam; I have bought used Juniper vf4000, without any (manuals,
datasheets, disc, etc.). So maybe you something like that or you has an idea
how to configure it. At least to reset the system, because I don't have
login or password.
Kind Regards,
Isam Awadallah
