[RESOLVED] SIP Configuration Question


#1

Hi:

No, I don’t have a whole list of problems and 2 miles of console output. :smile: I set-up an Asterisk PBX for use as an IP-only inter-office phone system using SIP phones. That’s my plan, anyway. Everything’s working as it should, except one item. I created a couple of extensions and gave them voicemail, although I hit some snags (where’s VoiceMail2?). But that’s besides the point. It works. I can call, I can leave messages in voicemail, I can listen to 'em.

My issue is this: I seem to be misunderstanding how the host= option in the sip.conf file works. If I set it to a specific IP, and I make a call to another extension associated with that IP, it rings a few times and turns me over to voicemail. Excellent, that’s exactly the way I’d expect it to work. However, if I set the extensions to host=dynamic, the connection loops back and rings on another line (X-Lite can handle three simultaneous calls).

This doesn’t make any sense to me. My SIP phone is registered (as far as I know) as a different extension. Why would the PBX loop the call back to a host that isn’t currently associated with that extension name? Shouldn’t it instead find the target extension unreachable, and turn me over to VM?

What am I missing here? I tried the defaultip= option and it didn’t do anything useful. Static IPs for everyone isn’t a realistic possibility, especially if I want their extension to follow them around. Any hints for me? Thanks in advance!

p.s. I googled this problem, and couldn’t find anyone else who noticed the same thing. Maybe the term isn’t “looping back”. I’m also covering all the bases here, and reading all the stuff I can find besides posting here.

p.p.s. I’m using X-Lite on GNU/Linux.

EDIT: Once the other extension is registered once, it works as expected, even when the remote phone is disconnected.