Hey there,
I’m Philipp, an IT Systems Engineer from Potsdam, Germany and want to deepen my knowledge in SIP communication on scale.
Currently, my setup is quite simple: A single stateless Asterisk instance, fully managed via ARI, which registers to multiple VoIP providers, only processes incoming calls to play audio files (no outgoing or conference calls).
Now, with growing load on the single Asterisk instance, I would like to have one or two Kamailio instances, which will balance the load of incoming calls to multiple Asterisk instances. On which Asterisk instance they end up is irrelevant for me, as the calls can be processed on every instance.
I looked into some Kamailio and Astricon conference speeches and checked out some tutorials. But still I’m not quite sure where to start. So here are some basic questions which would really help me out to get going:
As I don’t want to have registrations of all my Asterisk instances at the VoIP providers, I should use the UAC module to let Kamailio do the registering to the VoIP providers and register my Asterisk instances to Kamailio, right?
Will Kamailio automatically apply new outbound registrations when added to the database or do I have to trigger that manually?
Should I use one or two instances of Kamailio, how hard is it to configure them fail-safe?
For writing my first config file, should I start blank or from the standard config, what’s best practice?
Thanks in advance, looking forward to your replies!
Philipp
Hi,
I want to see if rtp packets are being relayed through kamailio. I am
attmepting to connect rtpengine with kamailio, but am not sure if it is
working, how can I tell?
I have the kamailio server on a different VM than the clients. In
wireshark, no rtp packets show on the server VM and rtp packets only seem
to be going directly through the client VM.
Rtp engine mentions being unable to find hashtables when attempting to
implement a delete_node command.
Thank you very much for your help,
Faiz
Hi there!
I bumped into this post to perform forwarding of REGISTER requests and then
saving a local cache on 2xx replies from main Registrar:
https://lists.kamailio.org/pipermail/sr-users/2020-October/110779.html
I think I understand all the steps described, but some features I need are
missing:
- How to change the Via headers to perform topology hiding? I understand
TOPOS and TOPOH do not work on these types of messages.
- Change contact header so that registrar responses traverse the Kamailio
box. (Use textops I suppose?)
- Besides, is this approach still the best in comparison to OpenSIPs
mid-registrar module?
Thanks in advance,
--
*Thomás Alimena Del Grande*
Engenharia - Aligera
Tel. 51 3500-0121
I noticed that when jsSIP UA that has registered over wss calls another
SIP UA that has registered over tls, record_route() adds only one Route
URI to outgoing INVITE (example below). This causes BYE to fail.
This issue may be caused by the fact that both UAs register over the
same Kamailio tls listening socket. Still two Route URIs should be
added in the same way as is done when one UA registers over tcp and the
other over udp.
-- Juha
---------------------------------------------------------------------
09:25:43.639833 wss:87.95.73.155:19458 wss:192.27.134.1:5061
INVITE sip:foo@test.tutpro.com SIP/2.0
Via: SIP/2.0/WSS jovobf9n4svm.invalid;branch=z9hG4bK2639638
Max-Forwards: 69
To: <sip:foo@test.tutpro.com>
From: " Test" <sip:test@test.tutpro.com>;tag=f7lpsuo7cb
Call-ID: ovhd9fu1fl1a66nec011
CSeq: 1387 INVITE
Contact: <sip:test@test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5>
...
09:25:43.649462 tls:192.27.134.1:5061 tls:87.95.73.155:19461
INVITE sip:foo-0x793ee87a90@10.158.141.103:38378;transport=tls SIP/2.0
Record-Route: <sip:192.27.134.1:5061;transport=ws;sn=ext_tls;lr;n2;dtlsf=avp;pm=0;ice>
Via: SIP/2.0/TLS 192.27.134.1:5061;branch=z9hG4bK637c.7e15ef5b84ea054cfc8ea3d4e860521f.0
Via: SIP/2.0/WSS jovobf9n4svm.invalid;rport=19458;received=87.95.73.155;branch=z9hG4bK2639638
Max-Forwards: 68
To: <sip:foo@test.tutpro.com>
From: " Test" <sip:test@test.tutpro.com>;tag=f7lpsuo7cb
Call-ID: ovhd9fu1fl1a66nec011
CSeq: 1387 INVITE
Contact: <sip:test@test.tutpro.com;gr=urn:uuid:e7f92a54-2295-4772-abc8-504be07e94c5>
Hello there,
Does anybody have experience installing the latest SEMS-Server (1.6.0) on
debian 10 buster?
I tried to install both 1.6.0. and 1.7-dev and some issues are occurred,
but i installed the version 1.3.1 on debian 8 before.
Any help would be appreciated.
Thanks with best regards
--
--Mojtaba Esfandiari.S
Hi list.
mY provider asks me to insert user = phone in my header. because it
requires to receive my INVITE with user = phone in TO and FROM
However, I add the user = phone in my configuration like this:
$ ru = "sip:" + $ rU + "@" + $ sel (cfg_get.pstn1.gw_ip) + ":"
+
$ sel (cfg_get.pstn1.gw_port) + "; user = phone";
but only the INVITE appears, not in the TO or FROM.
INVITE sip: 09872323232(a)X.Y.X.K: 5060; user = phone SIP / 2.0
From: <sip: XXXXYYYYY(a)172.20.12.4: 5060>; tag = 15552444533876
To: <sip: 09872323232(a)172.20.12.2: 5060>
You can help me solve it.
Thanks.
--
César Matheus
Hello,
the first group of presentations selected for the next Kamailio World
Online, September 1-2, 2021, has been published, aiming to provide a
good knowledge base of using Kamailio for different deployment types:
segregation of networks with an SBC-like role, controlling robots, RTP
streams management with RTPEngine, configuration variables or
transformations, SIP attacks handling and asynchronous SIP routing.
You can find more details at:
* https://kamailioworld.com/k09-online/
The schedule will be completed during the next days, with another group
of interesting sessions and open discussion panels.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Hi,
Couple of week before I have posted feature request -
https://github.com/kamailio/kamailio/issues/2807
As suggested, I am trying to figure out how I can achieve it with async or
sworker modules. Can someone help me to understand how to use those modules
to achieve async connectivity?
Regards,
Miteshkumar Thakkar