Hello Vik,
I have similar setup, try define you server configuration in tls.cfg
[server:ip address of alias interface:port of alias interface]
method = TLSv1
verify_certificate = no
require_certificate = no
private_key =
certificate =
ca_list =
crl =
Slava
From: "Vik Killa"
To: "s
I made successful audio calls from browser to browser using Asterisk
13.1 and SIPML5 browser phone.
Asterisk can't manage WebRTC video calls due to lack of codec
negotiation module, but I also faced RTP ports NAT traversal issue. To
my understanding Kamailio is capable to resolve this.
Can anybo
Hello
I need a small guidance on creating a light weight proxy which only forwards
the msgs to my sip server and also does supports symmetrical nated clients.
The way I have created the configuration is a slight modification of :
https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg
Hi Team,
I am using Jitsi client with Kamailio. I was able to IM between
the clients. How can I control the Message routing in Kamailio script. Please
help me with any link.
Regards,
Surya.
___
S
They should be unrelated, RTPPROXY called on INVITES and MSILO on
REGISTER|MESSAGE, why should INVITE's flow change using that msilo route ?
On Thu, Apr 30, 2015 at 4:13 AM, sscc wrote:
> # - msilo params -
> #!ifdef WITH_MSILO
>
> modparam("msilo", "db_url", "mysql://kamailio:abc@local
Hi.
Asterisk users are fine.
Register subscribers Kamailio is my problem.
#!ifdef WITH_ASTERISK
if (!auth_check("$fd", "sipusers", "1")) {
# OK Asterisk Users are no problem
#!else
