Things are always harder then they’re supposed to be. Plugging in a SIP client to Asterisk should be easy, right? Well, it is, if everything is just perfect. I took a few notes and a few screenshots to help others if they find this fairly typical setup is more difficult then it should be.
Here are the relevant parts (IP addresses and mailbox number changed to protect the innocent) of my sip.conf:
[general]
port => 5060
bindaddr => aaa.bbb.ccc.ddd
disallow => all
allow => ulaw
allow => alaw
context => sip-users
canreinvite => no
nat => no[gabe]
username => gabe
secret => ****
mailbox => 1337
callerid => Gabe Gunderson <1337>
type => friend
host => dynamic
To to avoid NAT issues (on my side, not the server side), this communication is going over an OpenVPN connection. Make sure your tunnel is up before you start Ekiga and go to the preferences page. Set “Listen On” to tun0 and not eth0 (which is the default). The settings don’t seem to “take” until you restart Ekiga.

I like to keep things simple in the codec department. The VSP we’re using offers only G.711 (U-LAW or PCMU and A-LAW or PCMA). We’re using U-LAW since it’s the standard here in the U.S. and it costs the CPU very little to go from U-LAW < => U-LAW. It’s a little fatter in the tubes, but that’s not a big deal on this particular install.

Here’s a typical Account. I’ve chosen to use names for sip users instead of numbers. It makes the configs a little easier to read and you can always map numbers to names in the dial-plan.

If you have problems, there are a few things you can do to help figure out what’s going on.
Try the following:
- Run Ekiga from the command line with the debug switch:
‘ekiga -d 2′ - Or get specific debugging from Asterisk, like so:
server*CLI> sip debug peer gabe
SIP Debugging Enabled for IP: aaa.bbb.ccc.ddd:5060





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