I have an Asterisk box with a public IP address and two SIP clients behind
the same NAT device(I also have SIP clients behind different NATs). I want
to know is it possible for Asterisk to detect if both clients are behind
the same NAT and use direct media between them and use other options for
cli
Hello Everyone.
Our environment is a register free setup, and our phones are set as
host=dynamic.
The problem we are experiencing is for inbound calls:
Name/username HostDyn Forcerport ACL Port
Status Realtime
222/222 (Unspecified)D N A 0
Unmo
Hello Markus,
Thank you so much for your response. Our switch is already generating
the needed P-Asserted header:
P-Asserted-Identity: "John Doe"
; user=phone; nat=yes.
I really did not want to have to rebuild it using `SIPAddHeader`
however, if I have no choice,
can someone please provide an ex
Am 16.02.2014 03:30, schrieb Nick Cameo:
Tried setting `sendrpid = yes` and still same problem. We really don't want to
have to `SipAddHeader` as it is already being formed by our switch.
From http://www.voip-info.org/wiki/view/Asterisk+SIP+trustrpid :
-snip-
P-Asserted-Identity
Asterisk does
Hi All,
I'm on a middle of an asterisk installation/configuration for my company
and I'm testing the TLS configuration.
For this reason, I used the ast_tls_cert script to build the ssl
certificates for my server.
On sip.conf file:
tlsenable=yes
tlsbindaddr=0.0.0.0
tlscertfile=/etc/asterisk/keys/a
