If you’re still considering a SIP/RTP proxy for NAT transversal, it’s not going to be an easy task but it certainly is possible. If you’re still up for it, there is documentation on how to configure OpenSIPS (fork of OpenSER and Kamailio’s counterpart) to work with Asterisk. You can view the documentation here: http://www.opensips.org/Resources/DocsTutorials#toc6
Note that while this kind of implementation may be complex, it is a viable solution for your problem. There are commercial services that offer the same service for residential VoIP service: http://www.star2billing.com/products/sip-proxy/
I am just using DMZ till I get it working, then I will limit to required ports only.
Yes, I test it with several routers and same problem.
Below is my sip settings:
Global Settings:
UDP SIP Port: 5060
UDP Bindaddress: 0.0.0.0
TCP SIP Port: Disabled
TLS SIP Port: Disabled
Videosupport: Yes
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
SIP domain support: No
Realm. auth: No
Our auth realm asterisk
Call to non-local dom.: Yes
URI user is phone no: No
Always auth rejects: Yes
Direct RTP setup: No
User Agent: FPBX-2.8.1(1.6.2.16.1)
SDP Session Name: Asterisk PBX 1.6.2.16.1
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Caller ID: Unknown
From: Domain: pbx.mydom.com
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Disabled
Qualify Freq : 60000 ms
Network QoS Settings:
IP ToS SIP: CS3
IP ToS RTP audio: EF
IP ToS RTP video: AF41
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Jitterbuffer forced: No
Jitterbuffer max size: -1
Jitterbuffer resync: -1
Jitterbuffer impl:
Jitterbuffer log: No
You did not describe the network well. The Asterisk system is behind NAT, can you forward ports? If you setup Asterisk NAT options, forward UDP 5060 and whatever range is defined in /etc/asterisk/rtp.conf you should be able to make this work.