Basic question about "sip show peers"


#1

When executing “sip show peers”, there is a column labeled “Port”. Is that the port that the phone is listening for SIP messages on? Or is that the port that the server is listening on?


#2

If I am not mistaken, that should be the port that asterisk will be sending its sip signalling to, which may not be the same as the port that the phone is listening on if you have NAT=yes for that port as it may be going through address and port translation. (For example, I have remote extensions, they show up as high port numbers because of NAT, but the far end phone is listening on 5060).

p


#3

Thanks for the reply, but now I’m even more confused. :confused:

Can you please describe to me how you have your NATed network setup? I’m having some issues with NAT and I’m not sure where the problem lays. Cisco phone work fine, but there are issues with the Aastra handsets.


#4

you may want to try describing your problem and your setup if you are having issues.

In general, nat=yes and qualify=yes for the remote extension, and then make sure that externhost and externrefresh, or externip are set as well as localhost so Asterisk knows what is local and what is not.

I haven’t used the Aastra’s to know if they are particular about anything. I know polycoms often need their registration set down to about 60sec (meaning registering about every 30 sec) to keep some NAT holes open.

p


#5

I’ve been using the same registration trick to keep the Aastra pinholes open as well. However, I’ve ran into a situation where one gateway (a Netgear Prosafe firewall) kills the pinhole immediately. I’m also concerned about the excessive and unneccesary registrations the more phones I rollout.

On the other hand, my Cisco 7960 works behind a NAT without a problem without having to use keepalive. I have it register using port 5077, created a matching NAT pinhole in the firewall, and set it reregister only once an hour.

This is how I’d like the Aastra phones to work. But, I’ve been going around and around with their technical support about why it doesn’t work with their product. (To their credit, they’ve honestly been trying to help though).


#6

I wouldn’t worry about the load from registration, it would probably take over 1500 phones to equal one VoIP call at least as far as the number of packets being sent are concerned, if you were registering every 30 seconds.

p