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
Hello,
I am using dsiprouter as a public facing server for pass through
registrations to a fusion pbx box. This setup works great and has for a
long time now. The issue I am having is that we are switching soft phone
clients to copier. Zopier uses a push server. When an incoming call comes
into the freeswitch server, it fires off an invite to the Zopier registered
client. For some reason ONLY in this scenario Kamailio is doing an auth
challenge to Freeswitch preventing to call to be sent to the soft phone.
How can I stop this from happening? Other soft phones and hard phones work
great with inbound calls except for Zopier.
thanks,