Hello,
Is there any reason the CentOS 6 RPMs don't build utils.so or
http_client.so?
Looking back to 4.3 it appears it was being built:
kamailio-utils-4.3.6-1.1.x86_64.rpm
However it is no longer available in 4.4 nor 5.0 RPMs for either CentOS
6 or 7.
Thanks,
--
Trevor Peirce
AcroVoice Solutions Inc
Hi all,
What is the quickest way to update Kamailio tables? I wanted to check out
the SIPTRACE module and I noticed the table is not there.
Thanks.
--
Andy Chen
Hello there,
Some times dns server fails and my kamailio server start replying
"Unresolvable destination (478/SL)".
The propose of my email is to know if is there any possibility to change
this reply (478) to for example one of 5XX replies.
I have modparam("sl", "default_code", 503) but in situations where dns
fails, kamailio doesn't reply 503.
Thank you for your support
Regards
José Seabra
Hi i have kamailio 4.2, dialog module with parametr
modparam("dialog", "dlg_flag", 5)
modparam("dialog", "enable_stats", 1)
modparam("dialog", "profiles_with_value", "caller")
modparam("dialog", "dlg_match_mode", 1)
modparam("dialog", "default_timeout", 3600)
modparam("dialog", "timeout_noreset", 1)
modparam("dialog", "send_bye", 1)
modparam("dialog", "timeout_avp", "$avp(i:4242)")
modparam("dialog", "db_url", DBURL)
modparam("dialog", "db_mode", 0)
in event_route[tm:local-request] i use
if(is_known_dlg()) {
xlog("L_INFO","DIALOG IS KNOWN $dlg(h_entry)::$dlg(h_id)) I
}
And when call drop by timeout, sometimes in log i see
/usr/local/sbin/kamailio[21144]: INFO: <script>: DIALOG IS KNOWN <null>::<null>
But usually this string in log file i see as "DIALOG IS KNOWN 3452::3334"
Why it may be? when the $dlg(h_entry)::$dlg(h_id) is not known, I can not track $DLG_lifetime.
Different between unsuccessful and successful call i dont see
Hi all
I'm looking for ways to push stats/alarms from kamailio to some external destination other than using the SNMPstats module.
Any ideas?
Thanks
Christoph
The information contained in this e-mail message is privileged and confidential and is for the exclusive use of the addressee. The person who receives this message and who is not the addressee, one of his employees or an agent entitled to hand it over to the addressee, is informed that he may not use, disclose or reproduce the contents thereof, and is kindly asked to notify the sender and delete the e-mail immediately.
Hi Team,
Question:
Just wanted to clarify regarding SIP URI in webrtc (over sip) connection.
e.g. if we have a scenario where Kamailio is hosted with websocket support.
Websocket URI is used to send packets to to wss/ws address and the SIP URI
goes with it. I have tested it with any SIP URI including any domain and it
works if I authenticate that domain, regardless that domain is real or not.
// client side SIP URI config
var configuration = {
uri : 'sip:alice@example.com',
};
Can we use any central domain for sip uri similar to realm concept and send
traffic to multiple kamailio servers through websocket LB.
dummy.domain - for ws/wss SIP URI
kamailio-a.domain.comkamailio-b.domain.com
Thanks in advance.
Regards,
Jade
Hi.
I'm just starting with Kamailio - I've been to a few presentations about it,
at Fosdem and KamailioWorld, but never used it so far.
I'm pretty familiar with Asterisk, and I want to use them in combination.
My first use case is to handle SIP registrations (both as a client, registering
to a remote SIP server, and as a server, accepting registrations from
handsets) in Kamailio instead of Asterisk, and still be able to route calls
to/from the registered accounts through the Asterisk dialplan.
I suspect this is a pretty common starting point for people who've used
Asterisk and want to add Kamailio to the setup?
So far I've been unsuccessful finding any Howto or Tutorial explaining how to
do this - I can't help feeling I must simply be looking in the wrong place or
in the wrong way.
Searching for topics like "Asterisk Kamailio Getting Started" I find:
http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf
but with 12 chapters and 85 pages I'm not sure which bits are the essential
parts to do what I need.
I've also found http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-
asterisk-11.3.0-astdb but this seems to focus on a database backend, which I
guess I'll want to use in due course, but doesn't seem the simplest place to
start just to learn how Asterisk & Kamailio work together?
So, please can someone suggest a beginner's guide to introducing Kamailio into
an Asterisk setup, keeping things as simple as possible to start with, so I
can learn about Kamailio one bit at a time (since there do seem to be a lot of
bits to it) from a familiar environment of using it with Asterisk?
Thanks in advance,
Antony.
--
What do you get when you cross a joke with a rhetorical question?
Please reply to the list;
please *don't* CC me.
Let's suppose that i have two machines with installed asterisk and one with
kamailio. I would like have routing based on sip domain. for domain
sip.domain1.com send sip signalling to asterisk#1 server and for
sip.domain2.com send to asterisk#2 server. What i should use? Domain module
or what ?
Tomasz
Kamailio v5.0.0 is out – it comes with 6 new modules and a consistent
set of improvements touching more than 50 existing modules.
You can read detailed release notes at:
* https://www.kamailio.org/w/kamailio-v5-0-0-release-notes/
Many thanks to all developers and community members that made possible
this release.
A consistent effort was directed to restructuring the source code tree
to meet more modern patterns; introducing the KEMI framework that allows
writing SIP routing blocks in other embedded interpreters such as Lua,
JavaScript or Python; removal of obsoleted MI control mechanism, being
replaced by RPC.
Enjoy Kamailio v5.0.0!
Thank you for flying Kamailio!
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference 2017 - http://www.kamailioworld.com