Hi all
version 5.7.6 Debian 12
I'm having issues with a kamailio server not sending 200 to some OPTIONS
requests.
It does send 200 to some subscribers but not 100% of the time to 100% of subs.
I have enabled debug=3 in that server and this is what I found:
If the log says that it found existing
El Wed, 19 Jun 2024 11:54:05 -0500
Brett Nemeroff escribió:
> What you should expect using this method is blocking while the exec runs
> which could run you out of child processes while they complete. I'd also
> expect to see an "unusual amount of CPU activity" which will be the result
> of the f
El Wed, 19 Jun 2024 19:54:26 -
"sarah.martin--- via sr-users" escribió:
> I am extremely new at this, but trying to set up TLS with a carrier. TLS
> connection is good, Invite goes out, we get the 100 and the 200, but
> subsequent messages (ACK and BYE) are being sent with UDP and I cannot f
El Tue, 18 Jun 2024 13:54:41 -0500
Brett Nemeroff escribió:
> Just want to add that exec is heavy and slow. I would not recommend it.
>
> Is there a reason you want to do this over http-ifying your script and
> using async?
>
>
Not really. The scripts are provided. Maybe I'm against "http all
El Tue, 18 Jun 2024 13:56:06 -0400
Alex Balashov via sr-users escribió:
> Yes, there is. You're best off building an rtimer+mqueue pipeline (or
> streamlined 'async' equivalent) and running your exec() calls in the
> asynchronous route. You can then t_continue() the transaction inside there.
>
Hi all
I've been working with async http client and async db queries in the past but
now I have to execute scripts and store the return values in avps.
Since I have no experience with that, I wonder how to achieve concurrency with
that scenario. Is there a way to exec async or suspend until the
El Thu, 6 Jun 2024 17:48:43 +
Henning Westerholt escribió:
> Hello,
>
> just to double check, you are also setting this two flags then in processing
> the INVITE, right?
>
Yes. I get the failed rows in the database tables "missed_calls"
I just want in a invite-407-invite-486 call to have a
El Thu, 6 Jun 2024 15:40:19 +0200
"Jon Bonilla (Manwe) via sr-users" escribió:
>
> DEBUG: acc [acc_logic.c:721]: tmcb_func(): acc callback called for
> t(0x7f7cec807480) event type 512, reply code 407
>
> DEBUG: acc [acc_logic.c:443]: should_acc_reply(): probing acc
Hi all
I'm trying to make kamailio not to insert a row in missed_calls table when
reply code is 407. But even if I set the "failed_filter" modparam the 407 row
is inserted.
The acc mmodparams are:
modparam("acc", "db_table_acc", "kam_acc")
modparam("acc", "db_flag", FLT_ACC)
modparam("acc", "db
El Sat, 20 Apr 2024 17:28:09 -
"christian.marinelli--- via sr-users" escribió:
> Hi all,
> i have a problem with my Kamailio SIP Server.
> I set up a Kamailio SIP server in a virtual machine on a private network and
> i connect to this with a WireGuard VPN. The problem is that i can connect t
El Mon, 26 Feb 2024 15:39:31 +0100
Benoit Panizzon via sr-users escribió:
>
> The message body is not the issue, this seems to be handled in a
> different memory buffer. The CPE crashed with a 'memory buffer too
> small' when composing the reply to an invite with 5 or more Via and RR
> even wit
El Fri, 19 Jan 2024 12:07:39 +0100
Daniel-Constantin Mierla escribió:
> Hello,
>
> the log message in your first email is about not finding the socket. The
> record route matches the domain, it is just that the socket to be used
> cannot matched by the second Route URI. You can name the sockets
Upgraded to 5.7.4
Then I added "sindominio.net" example domain to corex aliases
Debug Log:
Jan 19 08:14:32 pekedev33
kamailio: DEBUG: [core/modparam.c:135]: set_mod_param_regex(): found
in module corex
[/usr/lib/x86_64-linux-gnu/kamailio/modules/corex.so]
Jan 19 08:14:32 pekedev33 kamailio: D
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_d
El Sat, 13 Jan 2024 19:08:37 +0100
"Jon Bonilla (Manwe) via sr-users" escribió:
It was stupid indeed.
I wasn't seeing that I was receiving 5060 as ruri port.
For future readers: If you have $rp != $null only A and queries are
performed.
cheers,
Jon
--
PekePBX,
Hi
This may be a stupid question but after a couple of hours and some headache I
can't see where the problem is.
I'm trying to use SRV records to relay to different servers as I've done many
many times before. But in this case I'm getting a 478 from kamailio and I see
that it's not trying to reso
Hi
5.6.4
I'm trying to add uacreg logic to my setup. When I add a new entry via rpc I
see that the reg is stored in mem but not inserted in the database for
persistency.
My uac params are like this:
modparam("uac", "reg_db_url",DBURL)
modparam("uac", "reg_timer_interval", 900)
modparam("uac", "
El Thu, 23 Feb 2023 12:59:42 +0100
Daniel-Constantin Mierla escribió:
> You can also check the issuer of the certificate, there should be some
> variable in the config returning that when incoming traffic is over tls
> and the peer has presented a certificate.
>
>
Right
I was checking only in
Hi
Again, fighting with MS Teams but this time it's a kamailio question.
I have MSteams hosts in the dispatcher for OPTIONS and for destination.
Acording to MS you need to try sip.pstnhub.foo then sip2.pstnhub.foo
and sip3.pstnhub.foo
2 | sip:sip.pstnhub.microsoft.com;transport=tls | 0 | 1
2 | s
r that socket creating a custom
ca list? It still would not filter just MS.
In the end I guess I'll get an IP list and filter because opening two /14 nets
seems crazy to me.
>
> > On 20 Feb 2023, at 7:00 pm, Jon Bonilla (Manwe) wrote:
> >
> > Hi
> >
> > S
El Mon, 20 Feb 2023 13:06:30 +
Henning Westerholt escribió:
> Hello,
>
> correct me if I am wrong, but the second group of addresses ("/16") is
> included in the first group ("/14"), right?
>
> Regarding the large IP scope, this is the way Microsoft designed it,
> unfortunately. There is no
Hi
Sorry for the OT but I think here's the place where I an find a lot of Ms teams
integrations
I've been working on MS teams direct routing integration for PekePBX. It works.
I guess I've done it as everybody else, using Henning's guide as base and
extending it for multitenant setup (thanks Henn
El Thu, 28 Jul 2022 22:30:40 +0300
Ovidiu Sas escribió:
> I had a similar setup, with an empty dbtext table. Works like a charm!
>
> -ovidiu
>
>
Hi Ovidiu
I have a couple of performance questions.
First, seems like the module doesn't use it's own timer processes. You can set
it to the seco
El Mon, 1 Aug 2022 10:38:52 +0200
Ihor Olkhovskyi escribió:
> Jon,
>
> You can check one of my old blog posts related to this
>
> https://blog.provoip.org/2017/02/kamailio-limit-cpscpmconcurrent-calls.html
>
> It's based on htable and quite easy to implement
>
>
Hi
Ihor
Yes, sounds intere
El Thu, 28 Jul 2022 18:08:33 +
Henning Westerholt escribió:
> Hi,
>
> pipelimit allows dynamic queues from a DB. But if you want to limit
> concurrent calls, the best option is probably using dialog profiles with the
> dialog modules. For a cps limit you could also just use htable.
>
> Chee
Hi
I'm thinking on having a cps preference for each pstn gw that I can check.
I've started thinking in the pike module but it won't allow me to set custom
cps per peer. seems that the threshold is global.
Ratelimit module needs the queues to be set as modparam, not dynamic AFAICS.
Any hints?
Hi all
I've been checking for adding rcd info to te stir/shaken stuff using secspid
module. I've only found this thread:
https://lists.kamailio.org/pipermail/sr-dev/2021-June/063156.html
I'd like to know if someone has working code snippet using this function to
create/sign the json containing
El Thu, 7 Apr 2022 16:27:37 -0300
Luciano Motti escribió:
>
Seen this before in some providers. You should tell them that they're breaking
SIP.
As a workaround you could try topos module.
--
PekePBX, the multitenant PBX solution
https://pekepbx.com
pgpGOkWTIvuU_.pgp
Description: Firma
El Wed, 19 Jan 2022 16:06:04 +0100
marek escribió:
> hi,
>
> is it possible create log of SIP dialog like
> https://wiki.asterisk.org/wiki/display/AST/Unique+Call-ID+Logging ?
>
> it looks like its possible like this
>
> xlog("L_INFO", "D$dlg(h_id) something to log \n");
>
> but i must c
El Mon, 5 Jul 2021 06:26:12 +
Henning Westerholt escribió:
> Hello Alex,
>
> yes, sipwise maintain a fork of it.
> One related question, the last “official” release 1.6 of sems was in 2015 (at
> least in public github). Are people just using the development version in
> production then, also
El Tue, 22 Jun 2021 15:54:36 +
Peter Manley escribió:
> Hello,
>
> I'm using Kamailio 5.3.2 and am trying to establish a WebRTC connection for
> using SIP through Kamailio on port 8089. I am getting the following error in
> the Syslog:
>
> Jun 22 09:20:38 VRTPENGINE kamailio[43050]: ERROR:
El Tue, 23 Mar 2021 18:43:37 -0300
Vinicius Kwiecien Ruoso escribió:
> using the Path module on Asterisk, so when registering, the path is
> recorded and sent back from Asterisk, Kamailio is also not respecting
> that
you'll need to share your config and logs. This should work in your scenario.
El Tue, 9 Mar 2021 12:32:19 +0100
Daniel-Constantin Mierla escribió:
> The domain in the jsonrpc response refers to the location table --
> somehow internally the name of the table storing the contacts is also
> referred as domain. I just pushed a patch to add table to the response text.
>
> Do
El Tue, 9 Mar 2021 09:32:00 +0100
Daniel-Constantin Mierla escribió:
> Hello,
>
> the error message from kamctl, not from kamailio. You need to set
> SIP_DOMAIN inside kamctlrc file or do: kamctl ul show username@domain,
> where domain can be localhost or anything else. The kamctl is built that
Hi all
Using version 5.4.x here.
I save the registrations with use_domain 0 but I do use the domain module to do
some checks.
If I execute "kamctl ul" show I can see the registrations but if I execute
"kamctl ul show username" I get
"ERROR: domain unknown: use usernames with domain or set d
Sorry for the noise.
I had modparam db_url but no db_mode
--
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https://pekepbx.com
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Description: Firma digital OpenPGP
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El Sat, 17 Oct 2020 12:37:48 -0400
Fred Posner escribió:
> Dump should be the in memory, and show is from the db.
>
> Did you try a reload?
>
Also restarts :)
the logs for a reload:
# kamctl trusted reload
{
"jsonrpc": "2.0",
"error": {
"code": 500,
"message": "Reload failed.
Hi all
Just installed a kamailio 5.4.1 for a simple proxy scenario.
I created the version,address and trusted tables, and added some ips to the
trusted via kamctl.
If I execute "kamctl trusted show" I see the list of servers. But with "kamctl
trusted dump" I receive
{
"jsonrpc": "2.0",
"
El Sun, 11 Oct 2020 20:07:58 +
David VILLAUME escribió:
> Hello,
>
> I try to perform a registration caching in order to have a short interval
> registration on phone <> Kamailio (5 minutes) and a longest interval on the
> leg Kamailio <> registrar(1h).
>
> I’m not so sure about the best w
Hi
when I start kamailio in my dev system it fails to start and segfaults.
Version: 5.4.0
CFG Line: if (registered("kam_location",$tu)) {
Log:
Aug 14 10:20:29 pekedev2 kamailio: CRITICAL: [core/cfg.y:3588]:
yyerror_at(): parse error in config file /etc/kamailio/kamailio.cfg, line 382,
column
Hi all
A customer of mine has some crashes every week. We installed the dbg and
finally generated a core.
Bt full attached.
Kamailio installed from kamailio deb 5.1 repos
kamailio -V
version: kamailio 5.1.7 (x86_64/linux)
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCK
El Thu, 20 Dec 2018 12:11:19 -0500
Alex Balashov escribió:
> Hi John,
>
> Sorry, I accidentally deleted your original e-mail and thus am breaking
> threading; my apologies for that.
>
> As a matter of fact, I wrote an article recently to address the very
> questions you ask:
>
> http://www.eva
El Tue, 2 Oct 2018 22:15:48 +0200
Daniel-Constantin Mierla escribió:
> Even with client on the same host as the server, the port should be
> different. R-URI of ACK has port 5060, like on of the Route headers, so
> I assume only kamailio is using that port.
>
> R-URI for ACK has to be the contac
le 1.15. check_route_param usage
if (check_route_param("nat=yes"))
>
> Cheers,
> Daniel
>
> On 02.10.18 17:27, Jon Bonilla (Manwe) wrote:
> > El Mon, 1 Oct 2018 12:16:28 +0300
> > Igor Olhovskiy escribió:
> >
> >> Hi!
> >>
> >&
8, 1:54 PM +0300, Jon Bonilla (Manwe) ,
> wrote:
> > Hi all. I'm having problems with an indialog ACK request which is not being
> > routed correctly.
> >
> > I call loose_route and print:
> >
> > $du=sip:MYIP:2443;transport=ws;r2=on;lr=on;did=1d1.77
Hi all. I'm having problems with an indialog ACK request which is not being
routed correctly.
I call loose_route and print:
$du=sip:MYIP:2443;transport=ws;r2=on;lr=on;did=1d1.7702;nat=ws
$route_uri=sip:MYIP;r2=on;lr=on;did=1d1.7702;nat=ws
But when I call
check_route_param("nat=ws") I get fals
Hi all
After a migration, a customer is reporting one way audio on incoming calls. A
tcpdump capture shows 65 rtp packets in both directions but rtpproxy log shows
RTP stats: 1 in from callee, 64 in from caller, 65 relayed, 0 dropped
Any thoughts?
cheers,
Jon
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