Hello,
a short note to inform that the Call For Presentations is now open for
Kamailio World 2024. Everyone is welcome to submit proposal for
presentations to share the knowledge about Kamailio or Real Time
Communication services, security, high availability, scalability, etc.
Submission form and more details are available at:
- https://www.kamailioworld.com/k2024/call-for-speakers/
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com
Hello.
I have installed Kamailio version devel on Ubuntu 22.04 and use the mysql database on the same machine.
when i run
systemctl status kamailio.service
● kamailio.service - LSB: Start the Kamailio SIP proxy server
Loaded: loaded (/etc/init.d/kamailio; generated)
Active: active (exited) since Sat 2024-01-20 11:35:09 +0330; 21min ago
Docs: man:systemd-sysv-generator(8)
CPU: 48ms
/usr/local/sbin/kamailio[173481]: ERROR: usrloc [dlist.c:846]: register_udomain(): failed to open database connection
/usr/local/sbin/kamailio[173481]: ERROR: registrar [registrar.c:767]: domain_fixup(): failed to register domain
/usr/local/sbin/kamailio[173481]: ERROR: [core/route.c:1193]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:717
/usr/local/sbin/kamailio[173481]: ERROR: [core/rvalue.c:3818]: fix_rval_expr(): failure in cfg at line: 717 col: 22
/usr/local/sbin/kamailio[173481]: ERROR: [core/rvalue.c:3818]: fix_rval_expr(): failure in cfg at line: 717 col: 22
/usr/local/sbin/kamailio[173481]: ERROR: [core/route.c:1193]: fix_actions(): fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:720
/usr/local/sbin/kamailio[173481]: INFO: [core/sctp_core.c:53]: sctp_core_destroy(): SCTP API not initialized
kamailio[173463]: * already running
kamailio[173463]: ...done.
systemd[1]: Started LSB: Start the Kamailio SIP proxy server.
In /usr/local/etc/kamailio/kamctlrc, I uncomment these lines:
SIP_DOMAIN=ims.mnc55.mcc999.farazNET.org
DBENGINE=MYSQL
DBHOST=localhost
in /usr/local/etc/kamailio/kamailio.cfg i apply this changes:
auto_aliases=no
alias="ims.mnc55.mcc999.farazNET.org"
listen=udp:192.168.75.185:5060
listen=tcp:192.168.75.185:5060
how can i solved it?
Hi
I'm having an issue with the double rr for mst direct routing.
I'm using subdomains like 23234.mydomain.com with a wildcard certificate and I
set corex modparam alias_subdomains to mydomain.com
But loose_route doesn't seem to detect the rr as myself
I get the message:
rr [loose.c:804]: rr_do_force_send_socket(): no socket found to match second RR
(sip:20141.mydomain.com:5063;transport=tls;ftag=as7107d0fe;lr;r2=on;vsf=AAAAAAAAAAADCQMaABhMS1NYRh0cRllbR0RRQG91cC5lcw--;nat=yes;pet=mst)
Regarding my setup is more or less like this:
kamailio.local.cfg:
#!substdef "!MSTDOMAIN!mydomain.com!g"
kamailio.cfg:
modparam("corex", "alias_subdomains", "MSTDOMAIN")
But in the cli:
kamctl rpc corex.list_aliases
{
"jsonrpc": "2.0",
"result": [
],
"id": 1319303
}
I don't remember to have this issue when I was developing the mst integration.
Either I missed it or can it be a regression? kamailio version is 5.6.4 at the
moment.
Any ideas?
cheers
Jon
--
PekePBX, the multitenant PBX solution
https://pekepbx.com
Hello!
Thanks in advance for any help you can provide.
I have a simple setup here and am struggling a bit with getting a final
working solution.
This is just a single direction flow from a SIP provider inbound to servers
running HMPelements with CTI integration (sip endpoint). Essentially a
different DN / dial number would go to a different backend server. There
is no registration involved.
The kamailio server therefore needs to sit between the SIP provider and the
backend servers, it will be a NAT situation where private RFC1918 addresses
will be used on the backend after the server receives it with public IP
information.
So that's it. What is the simplest way to accomplish this?
If it would help - here's what ive done so far:
For the Kamailio server, I have an AWS EC2 instance with a public IP
bound/assigned to the main private IP. During troubleshooting, I've added
another private IP to the NIC so that I could throw up another listener.
I was thinking I'd need to accept the SIP on the first listener with the
"advertise" directive. The second listener would not have an advertise
directive so perhaps could be used to talk to the backends.
I defined the WITH NAT and rtp proxy settings (pointing to the first
listener below). The RTP Proxy daemon also knows about the y.y.y.y public
address below.
In the kamailio.cfg, I have a listener setup :
listen=udp:172.24.40.28:5060 advertise y.y.y.y:5060
listen=udp:172.24.40.26:5060
then threw in a route(CMS) in the request route and created the following
route block:
route[CMS] {
#check URI matches phone xxxxxxxxxx
if (uri=~"^sip:\+1xxxxxxxxxx@") {
rewritehostport("10.12.18.193:5060");
forward(uri:host, uri:port);
exit;
};
}
I get the SIP delivered to the backend host! but the INVITE headers are
stil pointing to public IPs and the SDP info is not adjusted to be a
private IP as well - as follows:
z.z.z.z = sip provider , y.y.y.y = public ip (aws ec2 ip). xxxxxxxxx =
calling phone, ######## = called phone (matches routeCMS above)
INVITE sip:+1##########@10.12.18.193:5060 SIP/2.0
Record-Route: <sip:y.y.y.y;lr>
Via: SIP/2.0/UDP
y.y.y.y:5060;branch=z9hG4bK28b7.56b42fdeec9f10cd13030b3711b16f4d.0
Via: SIP/2.0/UDP
z.z.z.z:11000;received=z.z.z.z;rport=11000;branch=z9hG4bKQ2yQKgv7NtKQF
Max-Forwards: 48
From: "" <sip:+1xxxxxxxxxx@z.z.z.z;isup-oli=0>;tag=tUF15gaNg6N6j
To: <sip:+1##########@y.y.y.y>
Call-ID: 9ee7e073-c7fc-46ac-a839-f3c77eaed903
CSeq: 78244364 INVITE
Contact: <sip:mod_sofia@z.z.z.z:11000>
User-Agent: 2600Hz
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER,
NOTIFY
Supported: path, replaces
Allow-Events: talk, hold, conference, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 278
X-FS-Core-UUID: 4e0b2928-d913-42c4-9809-f77a36d281be
X-FS-Support: update_display,send_info
P-Asserted-Identity: "" <sip:+1xxxxxxxxx@z.z.z.z>
v=0
o=FreeSWITCH 1705572686 1705572687 IN IP4 z.z.z.z
Thanks much!
Hi,
Yes I've tried using append_to_reply() in several routes
(tm:branch-failure, sl:local-response, tm:local-response, onreply_route,
failure_route) but it was not applied.
It works fine if a response is received or if it is generated by the
configuration script but not if this response is locally generated by a
kamailio module.
Regards,
Frédéric Gaisnon
> Custom headers to the reply can be added using append_to_reply() function
from textops module. Have you tried that?
>
> Regards,
>
> Fred Posner
> p: +1 (352) 664-3733
> > On Jan 18, 2024, at 2:07 AM, frédéric Gaisnon via sr-users <
sr-users(a)lists.kamailio.org> wrote:
> >
> > Hello,
> > I want to add a SIP custom header on a SIP response generated locally
(on the 408 generated when fr_inv_timeout is expired for example)
> > I've tried functions append_hf and append_to_reply in
event_route[tm:branch-failure:XXX] or event_route [tm:local-response]
without any success.
> > Do you have any suggestions to accomplish this ?
> > Regards,
> > Frédéric Gaisnon
Hi List
Imagine the following situation:
Two Kamailio registrar proxies handling the same domain. Location
Information synced via DMQ.
So if a device registers we don't know on which of the two registrar
proxies.
Core proxy is dispatching calls to either one of those registrars.
Core also performs AOR database lookup and adds an X- header telling
which AOR the call is destined to.
Registrar performs lookup of that AOR in the location database and
routes the call to destination.
Problem: The call could be routed to the the registrar which does
not hold the active socket to that client.
Solution: reg_fetch_contacts, cycle the contacts and append_branch if
the contact has a local socket.
Next Problem: I use a failure_route to sent the call to voicemail, if
not registered, busy etc. This would trigger, if we have no
local socket for the registration which is on the other
registrar. So I have to find a way to send the call to the
other registrar.
I guess I can append_branch a branch pointing to the URI of the
other registrar. This would then succeed in reaching the
device. Also using the failure_route would work this way if
we get a remote error from behind the other registrar or no
registration at all.
Next Problem: I use a branch trigger to remove undesired header I do
not want to send to the customer. Now imagine that AOR has two
contacts, one on registered on each registrar proxy.
So I end up with two destinations added with append_branch to
the destination set.
One points to the device of the locally registered customer
device, the other one points to the other registrar where to
find the other device.
I use a branch route trigger to remove the X- Headers towards
the customer.
But I need to X- Header containing the destination AOR to still
be sent to the other registrar.
Is there a way to trigger a branch route for only some of the
destinations added with append_branch? Or to trigger different
branch routes for different destinations in one set added with
append_branch?
Or am I doing this completely wrong and there is an easier way? ;-)
Mit freundlichen Grüssen
-Benoît Panizzon-
--
I m p r o W a r e A G - Leiter Commerce Kunden
______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
CH-4133 Pratteln Fax +41 61 826 93 01
Schweiz Web http://www.imp.ch
______________________________________________________
Sup guyz! On debian 12 i have kamailio 5.6.3 and when I try to use db_mysql module I always get an error: kamailio[2842268]: ERROR: db_mysql [km_my_con.c:163]: db_mysql_new_connection(): driver error: Can't connect to local server through socket '/run/mysqld/mysqld.sock'
I have no idea where it finds this socket cuz I don’t have it and don’t want to. Where I can change it?
Hello,
I want to add a SIP custom header on a SIP response generated locally (on
the 408 generated when fr_inv_timeout is expired for example)
I've tried functions append_hf and append_to_reply in
event_route[tm:branch-failure:XXX] or event_route [tm:local-response]
without any success.
Do you have any suggestions to accomplish this ?
Regards,
Frédéric Gaisnon
Hello,
Kamailio SIP Server v5.7.4 stable release is out.
This is a maintenance release of the latest stable branch, 5.7, that
includes fixes since the release of v5.7.3. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.7.x. Deployments running previous v5.7.x
versions are strongly recommended to be upgraded to v5.7.4.
For more details about version 5.7.4 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2024/01/kamailio-v5-7-4-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Many thanks to all contributing and using Kamailio!
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Kamailio Advanced Training, February 20-22, 2024 -- asipto.com
Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com