Hi list,
I have been looking at the recording daemon in RTPengine with Kamailio
controlling this via rtpengine_manage() flags: "record-call=on" in the
initial Invite, or 200OK based on direction.
I tested and this works brilliantly for the full calls, but when looking at
pause/resume the module documentation is bare and I am a bit confused.
It looks like the ngcp docs and Opensips module docs (for RTPengine)
describe the pause/resume as just another start/stop flag.
Use case is simple, all calls are recorded, but we want the option to
pause, then resume once any private or personal information exchange has
completed.
My question is also simple - Is this the recommended way to pause and then
resume recordings or have I missed something?
The Kamailio documentation doesn't explicitly say this (
https://www.kamailio.org/docs/modules/stable/modules/rtpengine.html#rtpengi…)
so just looking for advice if this is the best route to go.
Thanks,
John.
Hello,
On 20.01.25 08:25, Daniel-Constantin Mierla via sr-dev wrote:
> Hello,
>
> I think we should branch it on Wednesday, Jan 22, 2025, no matter on
> what stage we are with cmake support. The old-Makefiles should be kept
> anyhow (as they are or in a special folder with an easy way to recover
> them), because there are many bits and pieces that can be discovered
> later when more users.
>
> Because there were no many bugs reported to the C code specific to 6.0,
> we can aim releasing on January 29, 2025, with old-Makefile still to be
> used for tasks that are not yet covered by cmake.
shall we still aim for releasing v6.0.0 on this upcoming Wednesday,
January 29, 2025?
We the old-Makefiles kept, it is doable, the wiki pages needs to be
created for this release series, which I can do tomorrow, as I waited to
see if how it should be the recommended method for compilation/installation.
If the feeling is that we should wait, then probably the target date has
to be shifted somewhere in the week starting on February 10, 2025,
because Fosdem follows this weekend and then some countries have a week
holiday break.
If no other opinions and change of decision, the release will be done on
Wednesday, with content using the old-Makefiles (to reuse existing
tutorials) and reference to cmake variant when it is the case.
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, May 12-13, 2025, Berlin -- kamailioworld.com
Hello,
just wondering if there is any way to define 2 constants as :
# sets quarantine period the URL is set, in secs, after number of
HTTP_RETRIES retries
#!define SOME_VAR_1 60
# sets the configuration values for HTable module parameter
#!define SOME_VAR_2 "some_configration_value" + SOME_VAR_1
I couldn't find anything in docs, and made several different ways without
success.
Any suggestions?
Atenciosamente / Kind Regards / Cordialement / Un saludo,
*Sérgio Charrua*
Hello everyone!
I was wondering if it is possible to get information about a packet that is
being possessed by the worker without increasing log level?
I have an installation with Kamailio that is configured with two
interfaces. Once in a while Kamailio stops accepting requests on one of the
interfaces. It works fine for requests from other one. I think that worker
processes might be locked while processing some requests. Enabling debug
log is not an option because the traffic is quite high and the log file
will not just be enormous, some lines will be missing. It looks like syslog
will skip some of the log entries.
So is it possible to check what packet worker process is currently
processing? Or do I need to add some log entries with lower log level to
trace requests manually?
Thank you!
Hi List
I have some CPE which require special handling like removing 100rel and
update from the supported methods because they misbehave.
Previously, I was using reg_fetch_contacts and then appending the
branches manually and stacking the UA in an AVP.
With the branch index this allowed me to get the AVP containing the UA
to which the call was branched in the branch route and perform
filtering of methods based on UA, but it was complicated.
Now I use the intended function lookup_to_dset() but now I miss how to
access the register contact UA from within a branch route.
Is there an intended way to access that information?
Mit freundlichen Grüssen
-Benoît Panizzon-
--
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______________________________________________________
Zurlindenstrasse 29 Tel +41 61 826 93 00
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Hi folks,
Trying to hook up a new endpoint but am having an issue. Kamailio is in front of an Asterisk box.
They send an INVITE, we send 100, then 200 OK. However, when they send their ACK, the RURI is not set to the Contact of the 200, instead it is<number>@<proxy ip>. This causes the ACK to get routed to the proxy itself and the call fails.
82.160.190.100:5060 = Kamailio IP Port
198.201.240.242:5080 = Asterisk IP/Port
182.33.174.10:5060 = Provider endpoint
The far endpoint say they cannot fix the RURI - should I be able to handle this ACK below? My understanding is the ACK's RURI should be the Contact of the 200 OK.
200 OK sent from us to the Provider (Contact shows correct URI)
=========================================================
2025/01/15 17:14:49.470634 82.160.190.100:5060 -> 182.33.174.10:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 182.33.174.10;branch=z9hG4bKefc7.ad07f2d3.0
Via: SIP/2.0/UDP 182.33.174.39:5060;received=182.33.174.39;branch=z9hG4bK5bbf9591;rport=5060
Record-Route: <sip:1800715080@82.160.190.100:5061;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes>
Record-Route: <sip:1800715080@82.160.190.100;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes>
Record-Route: <sip:182.33.174.10;lr;ftag=as0b42eef3>
From: "0737965510" <sip:0737966610@182.33.174.39>;tag=as0b42eef3
To: <sip:1800715080@82.160.190.100>;tag=as4133584e
Call-ID: 480dc17b3f88516f364778dc4b2528da@182.33.174.39:5060
CSeq: 102 INVITE
Server: Asterisk PBX 18.9.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
*Contact: <sip:1800715080@198.201.240.242:5080>*
Content-Type: application/sdp
Content-Length: 314 ,
ACK Reply from Provider to Us (RURI points to Kamailio ip:port now
===========================================================
2025/01/15 17:14:49.478119 182.33.174.10:5060 -> 82.160.190.100:5060
*ACK sip:1800715080@82.160.190.100:5060 *SIP/2.0
Via: SIP/2.0/UDP 182.33.174.10;branch=z9hG4bKefc7.ad07f2d3.2
Via: SIP/2.0/UDP 182.33.174.39:5060;received=182.33.174.39;branch=z9hG4bK55656469;rport=5060
Route: <sip:1800715080@82.160.190.100;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes>,sip:1800715080@82.160.190.100:5061;r2=on;lr=on;ftag=as0b42eef3;did=f2e.ade2;nat=yes>
Max-Forwards: 69
From: "0737965510" <sip:0737966610@182.33.174.39>;tag=as0b42eef3
To: <sip:1800715080@82.160.190.100>;tag=as4133584e
Contact: <sip:0737965510@182.33.174.39:5060>
Call-ID: 480dc17b3f88516f364778dc4b2528da@182.33.174.39:5060
CSeq: 102 ACK
Thanks
-Barry
Hello,
Kamailio SIP Server v5.8.5 stable release is out.
This is a maintenance release of the latest stable branch, 5.8, that
includes fixes since the release of v5.8.4. There is no change to
database schema or configuration language structure that you have to do
on previous installations of v5.8.x. Deployments running previous v5.8.x
versions are strongly recommended to be upgraded to v5.8.5.
For more details about version 5.8.5 (including links and guidelines to
download the tarball or from GIT repository), visit:
* https://www.kamailio.org/w/2025/01/kamailio-v5-8-5-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 World Conference, May 12-13, 2025, Berlin -- kamailioworld.com
Hi,
I need to be able to pull in a header provided by a webrtc endpoint so I can use it in my request_route. I understand that I will need to use shm/htable to make it visible but I am unable to get the xhttp header using $header(x-p-push), or any header by name.
In my event_route[xhttp:request] I have the following:
if ($hdr(Upgrade) =~ "websocket" && $hdr(Connection) =~ "Upgrade" && $rm =~ "GET") {
....
xlog("L_INFO", "BF x-p-push header is $hdr(x-p-push) Contact is $hdr(Contact)\n");
}
The log shows both x-p-push and Contact header as null. sipdump shows the headers present in the request.
Am I missing something? Running Kamailio 5.8.4
Thanks
-Barry
Hello,
I am considering to release Kamailio v5.8.5 (out of branch 5.8) on
Thursday, Jan 23, 2025. If anyone is aware of issues not yet on the bug
tracker, report them there asap in order to have a better chance to be
fixed.
Cheers,
Daniel
--
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com