Nice undocumented feature discovered!

I spent a fair amount of time recently trying to get PBXes to send to a SIP URI on a particular server that for this document we will call 192.168.1.1. That server does not allow open SIP calling to its extensions.

I had a trunk registered to the same server 192.168.1.1 , but I found that PBXes was sending the call unauthenticated when sent as a SIP URI. I had tried as a classic extension and found that the caller ID was not forwarded from the calling party, although the call was sent with authentication.

I took a short nap, and it came to me. PBXes is Asterisk Based and Asterisk Lets you formulate a dial command such as DNIS@trunkname, in addition to DNIS@domain.ext.

So the name of the trunk we will call ABC. Instead of sending the call as 1234567890@192.168.1.1, I changed the dial command to 1234567890@ABC. The call then arrived to the destination server as authenticated! It was now sending via the registered trunk, not to a random URI, so I suspect the difference was it is now sent with username and password.

Too bad however I am still battling getting the original caller ID passed from PBXes to the external server on the inbound call. Also, apparently also I can not ring two URIs on the same external server via a PBX ring group. It appears only one of the calls is sent..

Nonetheless, it may well be a valuable find.

This post has been edited 3 time(s), it was last edited by sup on 09.04.2007 at 07:07.