Hello,
Kamailio SIP Server v5.4.4 stable release is out.
This is a maintenance release of the latest stable branch, 5.4, that
includes fixes since the release of v5.4.3. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.4.x. Deployments running previous v5.4.x
versions are strongly recommended to be upgraded to v5.4.4.
For more details about version 5.4.4 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2021/02/kamailio-v5-4-4-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Hello,
I am considering to release Kamailio v5.4.4 sometime next week, likely
on Monday, February 15, 2020. This is the usual heads up notification to
see if anyone is aware of issues not yet reported to bug tracker and if
yes, do it as soon as possible to give them a chance to be fixed.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla
Hello dears
I'm trying to build an IMS testbed based on kamailio and integrate it with the PSTN network, in order to do that I'm looking for AGCF,MGCF open source solution.
Am I able to do the functionality of those nodes based on
Kamailio,Asterisk or any other open source?
thanks and regards
Eyas
I am trying to get a webrtc setup going. Here is what I have
I have asterisk server at 10.123.244.18. The webrtc works internally from the freepbx UCP application As well as the Raspberry phone allocation. This server doesn't have any nat on it all devices are on local / reouted networks.
I have a Proxy server at 10.123.245.30 address. This server is located in AWS and has an elastic IP.
On this server I have ngiinx that will load the raspberry phone up.
What configuration do I need in kamailio and rtpengine to get this working.
If I forward port 8089 in nginx to the /ws side on my asterisk server I can get a call to bridge but with no audio and the call end at 30 seconds when remote. It works internally fine. Likely beccuae the web browser can get to https://10.123.244.18:8089/ws ports fine.
Thanks.
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I did more tests and found that loose_route() does not find socket by
name given in ;sn param:
Feb 12 10:40:59 lab /usr/bin/sip-proxy[16975]: WARNING: rr [loose.c:799]: rr_do_force_send_socket(): no socket found to match second RR (sip:tenantX.teams.tutpro.com:8007;transport=tls;r2=on;sn=ext_tls;lr;n1;savp=avp;pm=0)
In config I have defined:
listen=tls:192.x.x.x:8007 name "ext_tls"
-- Juha
Using a domain name in R-R header may not be such a good idea:
Feb 11 12:12:06 lab /usr/bin/sip-proxy[735]: WARNING: rr [loose.c:799] rr_do_force_send_socket(): no socket found to match second RR (sip:tenantX.teams.tutpro.com:8007;transport=tls;r2=on;pm=0;savp=avp;lr)
Or is there some means to tell K, which FQDN maps to which IP address,
when there are many such FQDNs?
-- Juha
Hi,
In my failover setting using Dispatcher module, I mark the GW if fails to response. OPTIONS are being sent every 30 sec:
failure_route[DISPATCH_FAILOVER]{
if (t_is_canceled()) exit;
if (t_check_status("500") or (t_branch_timeout() and !t_branch_replied())) {
ds_mark_dst("ip");
if (ds_next_dst()) {
t_set_fr(0,1000);
t_on_failure("DISPATCH_FAILOVER");
route(RELAY);
exit;
}
}
}
However, Kamailio still recognize it as active gw ( A.A.A.A is down, no response from OPTIONS being sent)
$ sudo sercmd dispatcher.list
{
SET_NO: 1
SET: {
SET_ID: 1
DEST: {
URI: sip:A.A.A.A:5060
FLAGS: AP
PRIORITY: 0
ATTRS:
}
DEST: {
URI: sip:B.B.B.B:5060
FLAGS: AP
PRIORITY: 0
ATTRS:
}
}
}
Any help would be appreciated.
AL
When request is sent from Kamailio to MS Teams SIP Proxy, the top R-R
URI needs to contain FQDN of Kamailio SIP proxy instead of its IP
address. Document
https://skalatan.de/de/blog/kamailio-sbc-teams
suggest to replace record_route(); call with
record_route_preset("SBC-DNS-DOMAIN:5061;transport=tls", "SBC-IP-ADDR:5060");
That works only in a very simple case where the request came in over UDP
or TCP and SIP Proxy has only one listening address, i.e., SBC-IP-ADDR.
One way to solve the problem might be a new r_r function that would take
FQDN of the top R-R URI as argument or introduction of a pv from where
the current record_route() function would take the FQDN if it has been
set.
Any comments or other solutions?
-- Juha
