http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
On 2016-11-09 08:13 AM, Ethy H. Brito wrote:
Hi all
I'd like to log the client IP addr and port used for SIP and RTP *during* in a
call.
The IPs must be the real source IPs (internet accessible).
How are these parameters available from dialplan?
For instance, ${SIPURI} holds the internal "IP:port" if the client is behind
NAT.
I need the external IP:port
Regards
Ethy
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
