What are you guys doing when you have sip phones behind nat on a remote end and then nat and asterisk on another?
Sip hardphone - nat - internet - nat -asterisk What setup are you using? Examples sip.conf and hardphone configs? Firewall settings? This is driving me crazy! I have qualify and nat?yes on sip.conf and when the hardphone calls the call comes thru but neither the caller or myself can heard each other... My nat side is forward port 5060 and 10000-20000 to asterisk... Any ideas? |-----Original Message----- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Matt Schulte |Sent: Lunes, 02 de Mayo de 2005 02:21 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] Debuging SIP | |sip debug |sip debug peer username |sip debug peer ip_address | |-----Original Message----- |From: Anton Krall [mailto:[EMAIL PROTECTED] |Sent: Monday, May 02, 2005 1:44 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: [Asterisk-Users] Debuging SIP | | |Guys. | |Im having NAT problems. Any good tips on how to debug remote SIPS, how |to see which ports are been sent and received, etc? | | |_______________________________________________ |Asterisk-Users mailing list |[email protected] |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |_______________________________________________ |Asterisk-Users mailing list |[email protected] |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | _______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
