sundance-communications.com/forum Forums VOIP Phones & IP PBX VoIP & Cloud Phones, Hosted PBX, General IP Phones Trixbox CE 2.6.1 Remote Extension (X-lite)

Hey guys,

I can't seem to get a voice path on remote extensions outside of my network. After much reading on various forums and trying a million different configs here's what I'm using:
sip_nat.conf:
nat = yes
externip = 142.xxx.xxx.xxx
externhost = mydomain.name
externrefresh = 120
localnet = 192.168.101.0/255.255.255.0

Extension settings:
secret = pwd
dtmfmode = rfc2833
canreinvite = no
context = from-internal
host = dynamic
type = friend
nat = yes
port = 5060
qualify = yes

I can call to & from other extensions, the voice path just never sets up. I've tried this from a number of different networks & ISP's. One thing that bothers me is when I call the remote extension from say extension 221 the display shows [email protected] calling. Why do I see a loopback address?

Any ideas?

three things to try.

1) Either define externhost with a valid FQDN or put a semicolon in front of it and externrefresh

2) Forward UDP ports 10000-20000 to your trixbox system

3) set the expiration time to 60-seconds by putting in 'defaultexpirey=60' in the top part of the file(make the phones re-register every 60-seconds)

I am also guessing that your Trixbox system is behind a nat, as well as the remote-end phones. Welcome to the fun that is SIP/NAT hell

Let me know if those help out.

Thanks Kumba, the ever knowledgeable VOIP expert! I actually did that very thing (before I had a chance to read your post) in sip_nat.conf:
nat = yes
externip = 142.xxx.xxx.xxx
;externhost = mydomain.name
;externrefresh = 120
localnet = 192.168.101.0/255.255.255.0

and instantly solved the issue. Though I'd tried that already :-(

I had already forwarded the required ports in my router and had even gone as far as putting it in the DMZ.

Thanks again!

You have to remember that after you make and configuration file changes that in Trixbox you have to tell it to reload or restart or something like that. It doesn't automatically load them changes in.

I forget to do that all the time with asterisk. Always frustrating when I burn an hour debugging something because I forgot to tell it to reload.

I've always wondered this; when doing a reload, will it drop the existing calls? Do you need to do it while idle?

If you are doing a reload it will not. If you are doing a restart it will kill everything.

The problem with doing a reload is when you do MAJOR changes to the dialplan. You could end up with a scenario where the current step in the call flow is looking for a place in the dial-plan that no longer exists. If all you are doing is changing simple things like sip user entries, voicemail entries, or some fairly minor parts of the dialplan it wont effect much. If you are doing major changes you would be best served to do a restart or reload with no active calls.

Anywhere I can get after hours Trixbox CE phone support? Having problems configuring a PRI to send or receive calls.

Thanks