Always Busy (486)

1. Brekeke Product Name and version:
Brekeke SIP Server, Version 2.4.7.3
Brekeke SIP PBX, Version 2.4.7.3

2. Java version:
JRE 1.6

3. OS type and the version:
Windows 2003 Server

4. UA (phone), gateway or other hardware/software involved:
X-Lite Phones

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html :

6. Your problem:
I am using a SIP SERVER (in a different network) with is working as Gateway to any other Systems (is already in use). This SIP Server should forward the (SIP)-Calls to the SIP PBX. There I have two problems:

- The SIP Server from the PBX answers all the time with "407". To solve it I have copied the existing default rule "To PBX auth" and modified it like this
Matching Patterns
$request=^INVITE
$addr=195\.65\.249\.235
Deploay Patterns
$target=127.0.0.1:15060
$transport=udp
$b2bua=false
$auth=false
&net.sip.fixed.addrport.uac=true

- Since now it is possible to call to the SIP PBX but now it answers to me all the time with "486 Busy"
In the ARS I have enabled the rule gw1. The values are set like this:
V1 = ^.+$ (this means everything!?)
V3 = 1073 (this is the extension where to call should go to.)
V4 = ^.+$ (this means everything!?)

Was it correct to change the rule from the Dial plan?
What do I have to set in the PBX that the call is accepted (not busy)?
By the way the outdial works proper.

This SIP Server should forward the (SIP)-Calls to the SIP PBX.

are you using brekeke pbx with bundled sip server or both brekeke pbx and brekeke sip server?
where are brekeke sip server and brekeke pbx installed? on the same pc or different pc?
if on the same pc, donot running both of them. brekeke pbx already has bundled sip server. donot need to install a separate sip server

I'm using a SIP Server and a SIP PBX bundled with SIP Server.
The SIP Server is in a different network in our Hosting Center and has connection to our office over Internet (VPN). At this location we have a gateway to the PSTN.
The SIP PBX bundled with SIP Server stays in our office.

Calls from the PSTN are recevied on the SIP Server and then they will be forwarded to the SIP PBX bundled with SIP Server.

This will mean that the SIP Server is "talking" to the SIP Server from the PBX and both are iin different networks.

As I worte the connection from PBX to the SIP Server (and then to the PSTN) is running fine.

V1 should be your gateway pstn number
try v1 = .+
this will make all calls sent to user 1073 even calls between extensions
it is better to defined more detailed matching patterns to make the ars rule only for calls from gateway/other sip server, like
From: sip:.+@other sip server IP