| 00:00.14 | brif8 | yes, I have all ports forwarded to .12 so as to get RTP to work |
| 00:00.34 | dant | that'll be why then :) |
| 00:00.52 | *** join/#asterisk M0c (~mochouina@modemcable105.167-203-24.mc.videotron.ca) |
| 00:01.25 | brif8 | ok so if that proves all ports are forwarded then why does the IP phone not connect to the RTP |
| 00:02.20 | dant | do you have to enable NAT on the phone? |
| 00:02.57 | crysflame | can you tcpdump on the router? |
| 00:03.00 | *** join/#asterisk mvand (~mvand@CPE-24-27-138-147.neb.rr.com) |
| 00:03.31 | brif8 | dant: think so will check, |
| 00:03.40 | brif8 | crysflame: dan't tcpdump the router no |
| 00:03.41 | dant | brif8, is your setup something like this? IP Phone <-> NAT <-> Internet <-> NAT <-> * |
| 00:03.55 | brif8 | dant: yes, exactly like that |
| 00:04.28 | brif8 | IP Phone <-> Linksys Router (NAT) <-> Internet <-> Linux Gateway <-> * |
| 00:04.43 | M0c | ish good luck hehe |
| 00:05.15 | slePP | can someone call in via SIP and dial 895? :> |
| 00:05.23 | dant | didn't realise... that could make things a tad difficult |
| 00:05.30 | brif8 | IP Phones NAT set to automatic |
| 00:05.34 | *** join/#asterisk BoRiS2 (~boris@24.77.165.150) |
| 00:05.34 | M0c | IAX nope, SIP yes |
| 00:05.40 | crysflame | is there an OS X SIP client recommendation? |
| 00:05.44 | M0c | IAX is NAT friendly |