Not possible to pickup calls from VPN -> outside VPN modu

[jj_boon]

1. Brekeke Product Name and Version:
3.178/348.2 PBX + SIP server
2. Java version:
1.7.0_21
3. OS type and the version:
Win Server 2008 Std R2 (64bit)
4. UA (phone), gateway or other hardware/software involved:
Brekeke server, also hosting as VPN server (with dual IP's set, one local one VPN tunnel IP)
Grandstream phone GXV-3140 on LAN
Commend SIP intercom module (ET 908A) on VPN, behind another separate network.
5. Your problem:
As this is my first time using PBX/SIP servers through VPN, i'm not very sure on the configurations

Current problem is, when i dial from my grandstream phone -> commend on VPN, it works flawlessly, where the phone quality is suberb.

But when I try to reach Grandstream through the Commend SIP module inside the VPN networks, it shows on my grandstream phone that data is coming in (commend is dialing to grandstream), but I can't initiate a call between both of the SIP modules (Can't pickup the call).

I am wondering where did i do wrong. Could anyone enlighten me on this particular matter?

Simple network IP layout

SERVER
vpn 10.8.0.1 (Open VPN) with 192.168.1.101 as it's Private IP, and 192.168.1.1 as gateway (normal modem without much problems)

Grandstream
192.168.1.210 Fix IP, not in VPN, with gateway 192.168.1.1 (Same as Server's Gateway)

Commend SIP module
192.168.2.211 FIX IP, with VPN address 10.8.0.6 (Open VPN), behind an celullar router with it's gateway address of 192.168.2.1

SIP users

Grandstream is holding user Ext 103. pw 103. (SIP Server 192.168.1.101)
commend is holding user Ext 101. pw 103. (SIP server address 10.8.0.1)

Any idea where could it go wrong? Or is there any logs that i could get from the brekeke itself?

Thanks!

[ambrosio]

It seems Commend's ACK didn't reach the SIP Server or Grandstream.

Go to the SIP Server's [Active Sessions] page and check the [Status].
What kind of call status is it when you pick the call at Grandstream?

[jj_boon]

At the grandstream phone, i have configured to auto pickup, hence both will write accepted on the [status], and shortly later become closing and both went disconnected.

(Addon)
It never goes up to the [Talking] stage, where i believe there is some settings that i didn't configure correctly.

but when i use grandstream to dial to the commend, it goes up to talking stage.

Thanks!

[ambrosio]

The "Accepted" means that the SIP Server received "200 OK" but didn't receive the ACK packet from the UAC (Commend) yet.

Try this rule.

Matching Patterns

$request = ^INVITE $addr = ^10.8

Deploy Patterns

$ifsrc = 10.8.0.1 &net.rtp.ifsrc = 10.8.0.1 $continue = ture

[jj_boon]

[ambrosio]

Yes. It should be $continue = true

Can you capture packets at the SIP server's PC?
http://wiki.brekeke.com/wiki/Capture-Packets-Wireshark

And check the 200 OK packet sent from the SIP Server to the UAC (Commend).