On Thu, 10 Nov 2016 00:35:54 +0100 Max Grobecker <max.grobec...@ml.grobecker.info> wrote:
> Hi Ethy, Hi Max and All. > > > Am 09.11.2016 um 17:13 schrieb Ethy H. Brito: > > > 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 > > > You can get the peer's signalling IP address from ${CHANNEL(recvip)} and the > RTP address with ${CHANNEL(rtpsource)} resp. ${CHANNEL(rtpdest)}. If you need > more information (like the codecs used) you can find other channel variables > on https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CHANNEL Hmmmm. ${CHANNEL(rtpsource)} is always returning something like "0.0.0.0:ppppp" where p=[0-9] and ${CHANNEL(rtpdest)} returns the internal (not accessible) IP addr if the caller is behind NAT, therefore, not what I need. Wouldn't these two variables have correct values only after the callee answers the call?? > > Please note that, if you have not disabled re-invites, the RTP address may > change while the call is running. Interesting observation. Thanx 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