Advolex - Kein schöner Leben

Ein privates Blog, völlig ohne Allgemeininteresse.

Name: Advolex
Location: Lidingö, Sweden

Thursday, January 29, 2009

asterisk behind NAT: externip

SIP and One-Way Audio

Är detta anledningen? - Även jag börjar bli nojjig p g a DynDNS.

-Mikael

clipped from forum.voxilla.com
I currently have 6 on-prem extensions, 6 off-prem extensions, 9 two-way SIP service connections, one inbound only SIP service connection and one IAX2 two-way service connection. Each one of these connections came with its own portion of pain.

The reason my brain centers on the dyndns is this: up until about a month ago I was running my Asterisk on a Linksys WRT54GS router, which passed traffic wonderfully because the Asterisk was actually on the public side of the router and not in a DMZ or behind forwarded ports. The WRT doesn't have enough horsepower to do switching and voice processing, so I couldn't set up either an IVR function or a Voice Mail function. That led me to shift the Asterisk to a dedicated Intel box.

I installed Asterisk@Home (because I couldn't get Asterisk CVS to install well over Fedora Core 3 on a Pentium 200 MMX), put the Asterisk server in my router's DMZ (the same WRT with Linksys firmware) and migrated my .conf files to the new box. The box roared to life, but would not pass audio to or from FWD or any of my off-prem extensions, although every other connection worked.

It wasn't until I set my externip to my dyndns FQDN that I could get audio on the troubled connections, although the audio would quit every once in a while. That's when I learned to reload my sip.conf after suffering a PPPoE IP address change. Everything has been smooth ever since, which is why I have been fixated on that part of the configuration.

Your question about ping response is simple. Unless you have forwarded the ping port (whose number escapes me) to your Asterisk box, it is your router that is responding to the ping. An easy way to confirm successful forwarding through your router is to forward port 80 (http) or 22 (ssh) through your router to the Asterisk box and see if you can reach AMP from a web browser or open a remote shell or SFTP session via an SSH client, since both of these server daemons are native to Asterisk@Home.

I wouldn't be so quick to change out your Asterisk version because of this problem. First of all, there's a directory /etc/asterisk/default that has clean copies of all of the .conf files should you feel you are irretrievably corrupt somewhere. Secondly, if you have other services and clients working you probably haven't screwed anything up.

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