I just found something odd in the Debian package for kamailio-als-modules
and I'm looking for someone to verify it for me.
I verified that I'm using Kamailio 4.2
# Kamailio -v
version: kamailio 4.2.0 (x86_64/linux) 36e695
the docs
http://kamailio.org/docs/modules/4.2.x/modules/tls.html#tls.p.tls_method
show that I should be able to use "*TLSv1+" however Kamailio crashes on
startup with *
[tls_mod.c:287]: mod_init(): Invalid tls_method parameter value
I have verified that TLSv1.1 and TLSv1.2 both work, but I can not use
TLSv1.1+ or TLSv1+
I also tried TLSv1.0+ ( I found it somewhere on the web ) but this doesn't
work either.
I used strings against tls.so to see what was available, and this is what I
found.
strings ~/tls.so | grep TLSv1
TLSv1_client_method
TLSv1_server_method
TLSv1_method
TLSv1_1_client_method
TLSv1_1_server_method
TLSv1_1_method
TLSv1_2_client_method
TLSv1_2_server_method
TLSv1_2_method
TLSv1
TLS method used (TLSv1, SSLv3, SSLv2, SSLv23)
TLSv1.1
TLSv1.2
Any idea why this is differs from the docs, or what I'm doing wrong ?
--
Sincerely
Jay
Hello,
Kamailio SIP Server v4.2.5 stable release is out.
This is a maintenance release of the latest stable branch, 4.2, that
includes fixes since release of v4.2.5. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.2.x. Deployments running previous v4.x.x versions
are strongly recommended to be upgraded to v4.2.5.
For more details about version 4.2.5 (including links and guidelines to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2015/05/kamailio-v4-2-5-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
Hi,
I'm working on a setup, where I have to support late SDP. It is working
right now, but I have to use an explicit rtpproxy_offer() when processing
the reply to an INVITE without SDP.
Is there a way to have rtpproxy_manage() handle those calls automatically?
I was thinking of setting a flag in the request which is used by the
rtpproxy_manage() function later.
Best Regards,
Sebastian
So im logging lots of information to Syslog to assist our helpdesk in
diagnosing customer issues.
Im seeing LOTS of user agents that re-use the sip Call-ID ( especially in
registers ).
What im wanting to be able to do is link a log message ( that I log at some
point in my dialplan ) to a specific SIP packet ( viewed from voipmonitor,
HOMER, wireshark ).
I cant REALLY see anything in the SIP to uniquely identify a packet,
however if kamailio provided the UDP packet checksum in a pseudo variable
then we could combine that with the sip call-id to hopefully do what im
after.
does anyone else have any ideas about how do what im after ?
--
Sincerely
Jay
Hello,
I am considering to release a new maintenance version from latest stable
branch - to be v4.2.5 - next week, most likely on Tuesday (May 19) or
Wednesday (May 20). If there is anything important to get in and not
discussed yet, bring the topic on development mailing list.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, May 27-29, 2015
Berlin, Germany - http://www.kamailioworld.com
i have configured msilo module successful but there isn't any voice with
msilo. i debug and compare the call flow with and without msilo. with msilo
in call flow it didn't follow to relay and consequently didnt activate
rtpproxy befor the call is answered. due to which during call there isn't
any voice. please help
--
View this message in context: http://sip-router.1086192.n5.nabble.com/NO-VOICE-AFTER-MSILO-tp137430.html
Sent from the Users mailing list archive at Nabble.com.
Hello,
I experienced a strange issue with some of VoIP accounts.
When the INVITE comes into MANAGE_FAILURE, after timeout, the config
identifies, with "dialplan", the right Asterisk instance that should handle
the call for voicemail.
This part is okay, and results in a new INVITE with the Request-URI formed
with the right domain (eg. sip: <sip:%3cNUMBER%3e@asterisk3>
<NUMBER>@asterisk3). Then, the request goes to RELAY. Here is the issue:
sometimes, the request is forwarded to the IP of the UA (the one initially
contacted) instead of the IP of Asterisk.
I can't figure out the difference between a succeeded call and a failed one.
If someone has an idea. Here is the config that handles the VoiceMail:
failure_route[MANAGE_FAILURE] {
[.]
if (isflagset(24)) {
$avp(s:inv_timeout) = "5";
t_set_fr($avp(s:inv_timeout)*1000);
if
(avp_db_load("$to/username","$avp(s:vm_uri)/usr_vm")) {
resetflag(24);
avp_pushto("$ruri","$avp(s:vm_uri)");
# Dynamic routing
if
(avp_db_load("$ruri/username","$avp(s:client)/usr_fai")) {
if
(dp_translate("2","$avp(s:client)/$avp(s:dest)") == 1) {
$ru = "sip:" +
$rU + "@" + $avp(s:dest);
} else {
# Load default
voicemail
$avp(s:client) =
"DEFAULT_VM";
dp_translate("2","$avp(s:client)/$avp(s:dest)");
$ru = "sip:" +
$rU + "@" + $avp(s:dest);
};
} else {
# Load default voicemail
$avp(s:client) =
"DEFAULT_VM";
dp_translate("2","$avp(s:client)/$avp(s:dest)");
$ru = "sip:" + $rU + "@" +
$avp(s:dest);
}
} else {
xlog("L_WARN","time=[$Tf] call id=[$ci]
call seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's
method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message
id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], User have no mail
box\n");
exit;
};
prefix("710");
xlog("L_WARN","time=[$Tf] call id=[$ci] call
seq=[$cs] contact header=[$ct] from uri=[$fu] from tag=[$ft] request's
method=[$rm] request's uri=[$ru] to uri=[$tu] to tag=[$tt] sip message
id=[$mi] process id=[$pp] ip source=[$si] flags=[$mf], failure route to
Voice Mail\n");
route(RELAY);
exit;
}
Regards,
Igor.
