In WebRTC (websocket) the contact header always has invalid URI, so, you
MUST either use path module OR add contact alias for both REGISTER and
SUBSCRIBE requests otherwise any subsequent incoming SIP requests such as
INVITE and NOTIFY won't be routed to endpoint (SIP client).
See websocket module
Check if you have,
# IP authorization and user authentication
> route[AUTH] {
> ...
> # user authenticated - remove auth header
> if(!is_method("REGISTER|PUBLISH"))
> consume_credentials();
> ...
> }
This basically allows only REGISTER and PUBLISH to be authenticated
Setup a counter in redis server shared across all kamailios, add to counter when call is answered and subtract when call hangs up.You can use any existing db that you already have instead of redis as long as it is accessible across all kamailios.--Muhammad Shahzad ShafiTel: +49 176 99 83 10 85Sent
Have a look at RTPEngine usage documentation here,https://kamailio.org/docs/modules/5.6.x/modules/rtpengine.html#idm80It answers all your questions. In short, you can have multiple sets of RTPEngines, each set can have multiple instances and each instance has weightage to define its priority within
That's likely because kamailio does not know where the destination endpoint is. You must either share user location information with kamailio or somehow keep track of registered users on kamailio (dual registration or cascade registration or path header if supported by pbx etc.).--Muhammad Shahzad
This error is seems to come from libstirshaken (
https://github.com/signalwire/libstirshaken/blob/master/include/stir_shaken.h
line 46) and has nothing to do with Kamailio. Please open a bug with
signalwire who owns and maintains this library.
Per my understanding this library is bit old and uses
Seems like you haven't set server IP. You need to SSH to server and then
type "setup" then go to
Network Configuration -> Edit device -> select eth0 (or whatever is your
ethernet device name is)
then uncheck DHCP and set static IP, also make sure set DNS correctly as
well. Then save and exit conf
Removing headers with up 5 characters in name would make SIP message
invalid since many critical headers such as TO, FROM, and VIA etc. will
also be removed. So, even if the regex works, Kamailio will NOT relay this
SIP message.
Thank you.
On Tue, Jun 7, 2022 at 5:00 PM Patrick Karton
wrote:
You need to increase xhttp_pi_buf_size,
https://kamailio.org/docs/modules/5.5.x/modules/xhttp_pi.html#idm193
--
Muhammad Shahzad Shafi
Tel: +49 176 99 83 10 85
On Thu, Oct 28, 2021 at 7:34 PM Abdirahman Osman
wrote:
> Hello Everyone,
>
> I am using the xhttp_pi modules to see the list of rule
Not really, these are PABX features and Kamailio is not such. You would need something like Asterisk or FreeSWITCH which have several options to do such stuff.Question is how would this media server know when to hold, transfer etc. If the endpoint is a dumb sip softphone as you described?Sent from
The dlg_var is string, make sure you cast them to int for duration calculations. Also manually cast var(v_duration) to int as well, because python_exec method treats them as string (this shouldn't be a problem since python vars are typeless but just to be sure).Now about the Null values in v_durati
The problem seems to that the kamailio is trying to route it over UDP port
using WSS transport, which will never work.
You are creating a transport bridge (UDP <-> WSS i.e. stateless transport
<-> stateful transport) as well media bridge (RTP <-> SRTP), so it is going
to be tricky. I strongly disc
Please put dlg_manage immediately after loose_route condition, e.g.
if (loose_route) {
dlg_manage();
...
}
There is no harm in calling it for every SIP method (not just BYE or ACK)
within dialog. It will help for re-invites (call hold / unhold events) etc.
from same buggy UAs as well.
Thank
Ok. Then possibly you have a buggy UA that is tempering with RR. Try
following,
1. Use dialog match mode 1.
2. Add dlg_manage after loose_route.
If problem persists then share sip trace of such call.
Hope this helps.
On Fri, 24 Jul 2020 at 16:33, Ilie Soltanici wrote:
> Yes, I do use the Re
What does the SIP trace at kamailio says? Do you see ACK pass through
kamailio? Are you doing record-route on call because otherwise ACK and/or
BYE may bypass kamailio.
Hope this helps.
On Fri, 24 Jul 2020 at 12:49, Ilie Soltanici wrote:
> Hello,
>
> I'm trying to get CDR working in Kamailio b
The SIP code 503 is tricky in the sense that i can indicate either server
maintenance or server overload. In both cases it can send Retry-After
header and any subsequent requests from same source are ignored for the
duration of Retry-After interval. [1].
Additionally RFC3261 and RFC3263 define tha
ion along those general lines.
>
> That still leaves some questions surrounding the purpose and necessity of
> various settings, however.
>
> — Alex
>
> —
> Sent from mobile, with due apologies for brevity and errors.
>
> On Mar 7, 2020, at 1:28 AM, M S wrote:
&g
Hi,
There are multiple scenarios and possibilities for this behavior, please
check,
1. What kind of sip endpoints you have? Any TCP based transport e.g. TLS or
WSS endpoint? Are they mobile apps (especially iOS devices) with push
notification and "backgrounding" support?
2. How many distinct SIP
You have many problems,
1. You do not seem to have TCP keepalives setup. So idle TCP connections
may be disrupted, closed or reset by any router on the internet, forcing
SIP client to re-register. As a result many more tcp connections are
created by a single SIP end point then expected. You said y
u can disable
"verify_certificate" option in tls module.
Hope this helps.
On Tue, 3 Mar 2020, 08:53 ahmed moghazy, wrote:
> I know that but i want voip program to be used with kamailio using tls i
> you could suggest one , thank you
>
> Sent from my iPhone
>
> On Ma
You don't need to use LUA for the task, you can use any language you are
comfortable with, e.g. perl, python or php. You can even use remote nodejs
and call it from kamailio as http request.
There are tons of example scripts, tutorials and howtos online regarding
how to send push notifications fro
kamailio.
On Mon, 2 Mar 2020, 20:27 ahmed moghazy, wrote:
> I put my own certficate in /res/raw/rootca.pem in linphone
>
> Sent from my iPhone
>
> On Mar 2, 2020, at 7:40 PM, M S wrote:
>
>
> Please read the tls module documentation,
>
>
> https://kamail
p must what what kind of
> parameters and how many parameters we needed ???
>
>
> Sent with ProtonMail <https://protonmail.com> Secure Email.
>
> ‐‐‐ Original Message ‐‐‐
> On Thursday, February 27, 2020 6:38 PM, M S wrote:
>
> Push notification setup is long
Please read the tls module documentation,
https://kamailio.org/docs/modules/devel/modules/tls.html#tls.p.require_certificate
when enabled it requires client to present certificate, but client does not
have one, which is the likely cause of the problem.
Hope this helps.
On Mon, 2 Mar 2020, 14:3
Push notification setup is long and complex but generally has following
process. Don't forget to read documentation of relevant kamailio modules
for proper use.
1. The iOS app must send push notification parameters in SIP request e.g.
SIP Register. In kamailio, when this request is received, you m
You are nesting one problem description into another and it all does not
makes any sense to me.
Anyways, what i understand is that you are trying to engage two RTP proxies
in series from WAN to LAN to WAN again. So you need to create media bridge
at each kamailio instance that routes RTP traffic f
You would need to install kernel module in host machine, only then it will
be available in docker container. You will also need to mark container as
privileged container and enable 1-to-1 NAT for RTP port range between host
and vm (i recommend using iptables for this instead of using docker port
ex
Hi,
I have a use case where I need to send a statefull (or stateless) reply to
non-INVITE SIP transaction from PERL script using app_perl module's
"moduleFunction" method,
https://kamailio.org/docs/modules/5.0.x/modules/app_perl.html#ID-7b0efd72c8ec89bc43dad940590b40c6
It fails with error "funct
. If however
the callee can NOT fix this problem then you will have store the original
Contact header in initial INVITE from caller side somewhere in kamailio
script e.g. htable, and change RURI to this value BEFORE calling loose
route. Which should fix the problem.
Hope this helps.
Thank you.
--
you.
--
M. S
On Tue, Aug 14, 2018 at 7:19 PM, Alex Balashov
wrote:
> One curiosity is that the reinvite does not appear to have the Via from
> the UAC that sent the initial invite. But certainly that would not cause
> this behaviour?
>
> On Tue, Aug 14, 2018 at 12:52:50PM -0400
I have tested it. It does not work. It only stops sip transaction
accounting but not the cdrs.
I think i have to add support in native code for dialog level flag or
variable to allow the customization of cdrs...humm, lets see.
Thank you.
On Sat, 21 Jul 2018 at 17:40, M S wrote:
> Ok, let
t
> you give it a try.
>
> On Sat, Jul 21, 2018 at 2:16 AM M S wrote:
>
>> Humm, isn’t acc log_flag is for transaction logging, rather then cdrs?
>>
>> For cdrs, the acc module has just one relevant module parameter, that is
>> cdr_enable which enables / disable
to FLT_ACC, it should be as simple as
> not setting or resetting the FLT_ACC flag, depending on the script logic.
>
> You will probably need to do that for both INVITEs and BYEs
>
>
> On Fri, Jul 20, 2018 at 2:48 PM M S wrote:
>
>> Hi,
>>
>> I have a kamailio v
Hi,
I have a kamailio v5.0 setup where cdrs are enabled using acc module. Is
there a way to enable / disable cdrs on per call basis in kamailio script?
For example i want to allow cdrs only for offnet calls and disable them for
onnet calls?
May be a dialog flag configurable in module parameters t
Hi,
Let me explain a bit the STUN server in kamailio is for SIP signaling to
help resolve NAT issues related to SIP over UDP for SIP clients. While a
separate / 3rd party STUN server is mostly useful for NAT issues related to
media problems.
Thank you.
On Mon, 7 May 2018 at 10:33, Mojtaba wrote:
Hi,
When kamailio dumps the messages to destination party upon register, do you
see its reply received by kamailio correctly? Most probable cause of the
problem is that kamailio does not receives 200 OK for dumped message from
destination party and assumes message delivery failure, therefore it st
Happy April Fools day or as celebrated in Britain and United States as
“Blame The Russians Day”.
Yesterday I caught a Russian spy in disguise of a 2 years old baby trying
to infiltrate our top secret society of open source telephony. I got
suspicious when i saw him saying very loudly “da”, “da” (R
Hi,
My understanding is that return value of last statement becomes the return
value of named route if none explicitly specified. Therefore i always
explicitly return 1 or exit.
—
Muhammad.
On Tue 20. Feb 2018 at 22:48, Jeff Bilyk wrote:
> Hello,
>
> Running into odd behaviour in Kamailio 5.0.2
First, RFCs related to SIP presence are quite confusing sometimes and often
not fully implemented by presence servers and endpoints.
Secondly, dialog presence event for first call has already completed its
life-cycle i.e. It has been terminated by second publish from Asterisk. You
can not change d
Hi,
Enable core dump and provide backtrace of sigfault. See this for more info,
https://sourceware.org/gdb/onlinedocs/gdb/Backtrace.html
Thank you.
On Fri, Dec 29, 2017 at 12:25 PM, Abdul Basit wrote:
> Adding further from https://stackoverflow.com/questions/2549214/
> interpreting-segfault
Hi,
I don't think you can FAIL a call that is already answered i.e. marked as
SUCCESSFUL. The only way to achieve the functionality you desire is to use
SIP REFER.
On Kamailio you can achieve this using,
https://kamailio.org/docs/modules/4.1.x/modules/dialog.html#idp17465536
On Asterisk you may
Hi,
Here is a simpler easiler solution for your case.
Prerequisits:
1. Media gateways should register for somewhat longer period of time, e.g.
3600 seconds (shorter intervals would work too but longer is better).
2. All media gateways should register to kamailio through same kamailio
listen socke
Hi,
There is a Kamailio sigfault issue #1277 that is also present in current
master branch. I would really appreciate if you can take a look at it
before next release, so i can upgrade my kamailio setup to v5.0 upon next
release.
Thank you.
On Mon, Oct 23, 2017 at 10:58 AM, Daniel-Constantin M
That's great. I was thinking of experimenting a few things with Kamailio in
docker but couldn't find any suitable image to start with. Thank you for
setting it up. I will definitely try it.
Keep up the good work.
--
M. S.
On Mon, Sep 25, 2017 at 7:35 PM, Sergey Safarov wrote:
&g
-Constantin Mierla
wrote:
> Hello,
>
>
> On 02.05.17 13:02, M S wrote:
> > Hi,
> >
> > I have a kamailio v4.4.5 deployment that serves TCP clients for SIP
> > and XCAP requests (without SIP presence). The kamailio's core TCP
> > parameters allow us t
Hi,
I have a kamailio v4.4.5 deployment that serves TCP clients for SIP and
XCAP requests (without SIP presence). The kamailio's core TCP parameters
allow us to set TCP read and write buffer sizes which are set globally for
all TCP connections.
However, i noticed that in general SIP over TCP requ
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