What you want there is the 'ctl' module and the 'kamcmd' utility that comes
with Kamailio.
# kamcmd ls
Gets a list of available RPC commands from all loaded modules + core.
Beware that if you put the ctl socket in a nonstandard place, you'll need to
specify the path via the '-s' option to
Awesome note.
How does one fire the RPC command. Whats the interface? Any docs on that?
KD
On Wednesday, May 9, 2018, 10:06:35 PM EDT, Alex Balashov
wrote:
There are a lot of options here.
For source IPs and subnets, the `permissions` module probably works
Update : I piggy backed Freeswitch to Kamailio and have resolved the issue. No
media touching us now. Pros and Cons for doing that I know. But we have it
working.
Thanks for the dialog.
KD
On Wednesday, May 9, 2018, 8:57:49 PM EDT, KamDev Essa
wrote:
Response
There are a lot of options here.
For source IPs and subnets, the `permissions` module probably works
best:
https://kamailio.org/docs/modules/5.1.x/modules/permissions.html
While it can function in a mode where it bangs on your database for
every request, it also supports a caching mode (db_mode
Hi all,
Thanks for your nice help!
I have built Kamailio successfully. I just created a new link to libmysqld.so
in /usr/lib64.
Thanks and Best Regards,
Charlie
From: Sergey Safarov [mailto:s.safa...@gmail.com]
Sent: 2018年5月9日 21:41
To: mico...@gmail.com; Kamailio (SER) - Users Mailing List
Response from the carrier matches your description. Looks like their Inbound
carrier is latching but the outbound carrier is not and yes they recommended
handling the NAT on my end.
That said, whats my options here. Is the native rtpproxy scalable? or is it
better to go with a Freeswitch farm
Hi All,
Does anyone have any thoughts on how I can debug this further, I'm
currently stumped as to what steps to take to see why these messages are
failing authentication.
Any suggestions appreciated.
Thanks
On 03/05/18 11:02, Asgaroth wrote:
Hi All,
I am testing a scenario where we
Re-read the piece of the article related to "RTP latching":
http://blog.csrpswitch.com/server-side-nat-traversal-with-kamailio-the-definitive-guide/
In order for RTP to reach a NAT'd endpoint, all other things being
equal, the other party has to do RTP latching. This is true of both
inbound and
Well then why does Inbound (from carrier to NATed UA) call work. Kam is doing
something clever there. Why not when sending the call out.
KD
On Wednesday, May 9, 2018, 4:34:55 PM EDT, Alex Balashov
wrote:
Oh, the UAs are NAT'd? Yeah, you're going to need
Greetings list,
I need to put presence info of a device into asterisk database through
sqlops module.
Currently, i opt to use a lua script to parse xml body of PUBLISH packet
issued by pua_dialoginfo and insert trying/ringing stats into asterisk
database.
Is there any better way to accomplish
easy. Not one of my domains or source IPs. Unless hackers is ex user they
would never know my list.
KD
On Wednesday, May 9, 2018, 4:53:21 PM EDT, Alex Balashov
wrote:
How would you define "foes" in a programmatic sense? :) That will dictate the
answer.
How would you define "foes" in a programmatic sense? :) That will dictate the
answer.
-- Alex
--
Sent via mobile, please forgive typos and brevity.
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We have noticed pesky SIP hackers already knocking on our POC servers. Kam is
holding the fort by sending out the proverbial 401/407 (snore). I can SQLOPS to
do a dip to check destination domain or Source IP and nix the snore for alien
requests, but wanted to know if there was a native way to
Update. I just checked inbound calls using the "rtpproxy clean" cfg.
Inbound calls from Carrier to UA work fine with bidirectional RTP. Its the
outbound calls from UA to carrier that have a NAT issue.
I am so close :)
KD
On Wednesday, May 9, 2018, 4:29:39 PM EDT, KamDev Essa
Oh, the UAs are NAT'd? Yeah, you're going to need something clever in the
middle that can do the RTP latching, then.
-- Alex
--
Sent via mobile, please forgive typos and brevity.
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That did not help. sip capture shows NATed SDP being sent to carrier.
Obviously carrier will not be able to make head nor tails about it. And carrier
SDP sent to UA is perfect as its not NATed. So UA ===> Carrier is fine. Carrier
> UA is no RTP. Does not look like Kam alone can do RTP
Am Mittwoch, 9. Mai 2018, 18:29:05 CEST schrieb John M:
> I upgraded Kamailio 4.4 to 5.2.0-dev5 and when I tried to start the
> service(p-cscf), I am getting below error. However I am able to start the
> service with 4.4 p-cscf example cfg. Also after upgrading I-CSCF and S-CSCF
> nodes are
If I comment out the modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:7722") as suggested, I do get RTP flowing just in one direction.
From the UA to kam and out to the carrier. I get nothing geting back to the
end point. In other words 1 way RTP. Looks like its Kam RTPProxy or bust.
KD
Correct. So remove all semblance of any RTP proxy and the resulting behaviour
will be exactly what you expect.
On May 9, 2018 2:13:22 PM EDT, KamDev Essa wrote:
>I dont. I want the 2 end points to talk to each other because I am on
>AWS with shaky bandwidth stats. It can
I dont. I want the 2 end points to talk to each other because I am on AWS with
shaky bandwidth stats. It can handle signalingbut not RTP.
However the cfg entry modparam("rtpproxy", "rtpproxy_sock",
"udp:127.0.0.1:7722") gives me to believe (not tcdumped RTP ports yet to prove
it ) that Kam
That depends. Start with a more basic question: why do you need RTP relay in
the middle at all?
On May 9, 2018 1:54:55 PM EDT, KamDev Essa wrote:
>So all calls that kamailio processes using the default cfg file anchor
>RTP on the kamailio server? Is it a best architecture
So all calls that kamailio processes using the default cfg file anchor RTP on
the kamailio server? Is it a best architecture to farm out RTP to Freeswitch?
Or is the Kamailio RTP proxy a better gig?
KD
On Wednesday, May 9, 2018, 1:38:04 PM EDT, Alex Balashov
Ironically, nothing. Kamailio doesn't touch the respective parties' SDP unless
you invoke an RTP relay (or something else like fix_nated_sdp()).
On May 9, 2018 1:03:19 PM EDT, KamDev Essa wrote:
>What cfg files changes do I need to make to get Kamailio to be a
>signally
Hello All,
I am trying to resolve, in my mind, the flow of a WebRTC<=>WebRTC call using
Kamailio and Asterisk.
Each WebRTC client is registered in Kamailio and when I call WebTRC Client1
from WebRTC Client2 what I see is ->
The Invite is sent from Kamailio to Asterisk and then Asterisk is
What cfg files changes do I need to make to get Kamailio to be a signally only
server, yet manipulate the SDP part of the INVITE message to allow remote
parties to send media to each other? In Freeswitch terms "bypassmedia".
KD___
Kamailio (SER) -
Hello list,
I'm looking for to understand a problem have with Kamailio.
The configuration is:
user (behind NAT) -> Kamilio (Public IP) -> user (behind NAT)
When I make a call from user A to user B (both registered) , User B
receive the INVITE correctly but in the ringing SIP message change
Hello,
I upgraded Kamailio 4.4 to 5.2.0-dev5 and when I tried to start the
service(p-cscf), I am getting below error. However I am able to start the
service with 4.4 p-cscf example cfg. Also after upgrading I-CSCF and S-CSCF
nodes are running without any issue.
Any help would be greatly
Hi!
You can to hande it with add_contact_alias but im not sure it will rewrite
transport for you
also if you will store contact as it is on your backend it is a big chance
that it can be unusefull with your SIP service because conract uri is
encrypted and most of b2b servers like asterisk for
Need to install mysql-devel
https://github.com/kamailio/kamailio/blob/master/pkg/kamailio/obs/kamailio.spec#L554
And then apply Daniel recommendations.
ср, 9 мая 2018 г. в 14:42, Daniel-Constantin Mierla :
> Hello,
>
> can you go to src/modules/db_mysql and run:
>
> make
Hello,
i don't think that GStreamer can handle SIP or SDP information.
It is just a ...
" GStreamer can bridge to other multimedia frameworks in order to reuse
existing components (e.g. codecs) and use platform input/output mechanisms:"
And to point at RTP ... some codes are missing like
Hello,
I am using Kamailio as registration server and FreeSwitch for signalling
(RTP packet handling). Can I use GStreamer instead of Freeswitch or
Asterisk?
Thank you.
Regards,
CM
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Hello Ye,
i run MariaDB cluster on different servers.
I installed Kamailio with normal MySQL first and after i redirected to
the MariaDB cluster.
hope it helps :))
Regarts Rainer
Am 09.05.18 um 10:46 schrieb Ye, Charlie (NSB - CN/Qingdao):
Hi Daniel,
Please find my answer as below:
Hi Daniel,
Please find my answer as below:
The operating system: Red Hat Enterprise Linux Server release 7.4 (Maipo).
I didn’t modify the Makefile, only include db_mysql module when made cfg.
Thanks and Best Regards,
Charlie
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent:
Hello,
what operating system are you using?
Have you done any changes to the Makefiles, or is just stock Kamailio?
Cheers,
Daniel
On 09.05.18 10:23, Ye, Charlie (NSB - CN/Qingdao) wrote:
>
> Hi all,
>
>
>
> Does everyone use MariaDB in Kamailio? When I built Kamailio with
> MariaDB, the
Hi all,
Does everyone use MariaDB in Kamailio? When I built Kamailio with MariaDB, the
below error happened. Would anyone give a help? Thanks in advance!
[root@clcmos /usr/local/src/kamailio-devel/kamailio]$ # make all
make -C src/ all
make[1]: Entering directory
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