Hi, I have have Kamailio as SBC one-end is SRTP and other end is RTP
So I m using
rtpengine_manage("replace-origin replace-session-connection DTLS=passive OSRTP-offer ICE=force SRTP")rtpengine_manage("replace-origin replace-session-connection DTLS=off SDES-off ICE=remove RTP")
1-Asterisk(RTP) -----2-Kamailio/RTPEngine ------ 3- Carrier with SRTP
I detect source IPs and to manage this all.
When call is made from 1 to 3 all works and during call 3 send re-Invite to Hold Call .. We need to call rtpengine_manage and also when on 200OK reply of re-invite from 1.
How shall I detect that 1- if this is Re-Invite (has_totag() )and 2- its from Callee.
Please advise.
Hi;
I want to remove all plain text usernames an passwords from kamailio.cfg
file. Like modparam("auth_db", "db_url", "dbdriver://username:password
@dbhost/dbname")
or this modparam("sqlops","sqlcon","ca=>dbdriver://username:password
@dbhost/dbname")
Can you help me with some ideas of how can I handle that?
Thank you.
Hi everyone,
I've got a specific case: when the inv_fr times out, I need to add a Reason
header to the CANCEL generated by kamailio. I've tried to see if I could do
it in the onsend_route, but that one is not triggered for the generated
CANCEL. I also checked event_route[tm:local-request], but that one isn't
triggered either for the generated CANCEL.
Is there any way to do it? Or maybe to have any pointer about where to look
in the code so I may try to trigger event_route[tm:local-request] for these
generated CANCELs?
Regards,
Alfonso
Hello guys,
I have a setup like this:
FS(private)-->(private)Kamailio1(Public)-->Kamailio2(public)-->provider.
I have mhomed=0, enable_double_rr=1, I'm listening on 2 different ports, 1
private, 1 public (with advertise) and on Kamailio1 the DISPATCH ROUTE I
force the socket
$fs = "LISTEN_PRIVATE_PROTO:LISTEN_PRIVATE_IF:5080";
To the private ip when the destination IP is a private IP.
This all works fine.
But When calling OUT (i.e. public) kamailio adds itself only for the
private IP:
*INVITE sip:12345678@mydmain.com <sip%3A12345678(a)mydmain.com>
SIP/2.0Record-Route: <sip:172.31.25.124:5080;lr;did=9da.3a61;nat=yes>Via:
SIP/2.0/UDP
172.31.25.124:5080;branch=z9hG4bKf05e.979fd257ac7f288804ef146d12c1e0fb.0Via:
SIP/2.0/UDP
172.31.6.1:5080;received=172.31.6.1;rport=5080;branch=z9hG4bKQ3H8mgHaUjaBm*
So when INFO messages come from the provider to kamailio2, kamailio 2 tries
to send the INFO straight to "172.21.25.124".
Should i manually add a record-route when calling OUT with
record_route_advertised_address("1.2.3.4:5090");
Or is there a better way of getting this working?
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
Hi there!
Configuring the UAC module to use "remote registers" in a Kamailio 4.4.3.
Everything it's working except for the flags database field to which I
always put a value of 0. But when Kamailio starts to launch the
REGISTRATION that field does not persist in the DB.
In the kamailio.cfg file I have not made any changes, I only use the
remote registers DB feature.
Is it normal behavior? Do I have any parameters not set or incorrectly set?
Regards, JV
Hi there,
Does Kamailio have a function to add URI parameters? I looked around but
couldn't find one. I refuse to believe Kamailio doesn't have one.
There's a *add_uri_param(param)*, but that's strictly for R-URI. I need it
for something else, Contact.
Thanks,
--Sergiu
Hi
I want to configure Kamailio SIP server to act as a SBC.
I had read the article of Kamailio working as SBC to connect MS Team project:
https://skalatan.de/en/blog/kamailio-sbc-teams
But I can not find the kamailio.cfg file for this scenario.
Can anybody give me a configuration template of SBC?
BRs
Albert
Hi,
I am using FreeSwitches behind the Kamailio proxy server and I am trying to
allow multiple registration to my extensions.
So, following configuration is sample of my Kamailio
modparam("registrar", "xavp_cfg", "reg")
......
$xavp(reg=>max_contacts) = 10;
save("location");
....
I saw my phones could register with the same account credentials via
several phones such as Cisco, Zoiper, Yealing etc. When the call is forward
to this extension, all of them are ringing. Very Nice.
But, when I am trying to REGISTER WebRTC supports soft-phones to my system
and with the same account credentials, my extensions are not ringing like
in the previous scenario. WebRTC uses Websocket (WS) technology and
clients register to Kamailio via usrloc module.
When the call is forward to this extension, Kamailio try to replicate
WebRTC'S INVITE packet to other phones (Cisco, Yealing, zoiper etc) and
none of them understand incoming INVITE request because of WebRTC
supported protocols (ICE ,a=candidate) , in a brief, phones could not
recognize/understand incoming WebRTC request.
This is a really tough issue for me, how can I send appropriate INVITEs for
each of them.
