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
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
Hello Markus,
Thank you so much for your response. Our switch is already generating
the needed P-Asserted header:
P-Asserted-Identity: John Doe
sip:14167493...@toronto.location.com; user=phone; nat=yes.
I really did not want to have to rebuild it using `SIPAddHeader`
however, if I have no
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
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
