1. Brekeke Product Name and Version: SIP Server 2. Java version:1.7 3. OS type and the version:Windows 2008 R2 4. UA (phone), gateway or other hardware/software involved: 5. Your problem: Call gets connected..but after 30 seconds I see this "242 Destination unreachable (port unreachable) error ICMP ...

Finally I have it working. thanks for the replies

That resulted in no audio end to end. So back to my original pattern.

Thanks. I will give it a try.

I don't have access to the router settings.

No. For internal communication it should be internal IPs. But from the SIP phone in the internet to internal devices, it has to be external as it in the public internet. But I got it working with following Deploy Patterns. I don't know how it is working. Let me know if there is a better way ...

setup is in the following link.
http://postimg.org/image/s5dbmx16h/

See the setup here.
http://postimg.org/image/s5dbmx16h/

Thanks for the reply. [Deploy Pattern] $nat = true $continue = true &net.rtp.ifdst.audio = 2.2.2.2 The above pattern works when SIP phone in public internet makes a call to the SIP phone registered to the brekeke SIP server (which is inside the NAT). But now I have a Lync+mediation server behind ...

It is already set.
Even with this settings:RTP Relay on I don't know how will the local IP of Berkeke go as remote IP to the person out in the public internet?

I think I will have to manipulate the dial patterns. Not sure where I have to change the local IP with the remote IP.

Tried with no luck. because I don't know how will my local IP go as remote IP to the person out in the public internet?

Because I have two scenarios. Setup:- I have a SIP device and SIP proxy in public internet. SIP phone is registered with this SIP proxy. There is NO NAT with the device and the proxy. Then in Amazon Cloud over a NAT I have the following:- (1-4 shows the call flow routing order) 1. RADVision SIP ...

1. Brekeke Product Name and Version: SIP Server 3.3.4.4 2. Java version: 3. OS type and the version: Windows 2008 R2 4. UA (phone), gateway or other hardware/software involved: 5. Your problem: Can I intercept SIP session progress, OK based on message type, contact and TO? then can I change ifdst ...

1. Brekeke Product Name and Version:SIP Server 3.3.4.4 2. Java version: 3. OS type and the version: Windows 2008 R2 64 bit 4. UA (phone), gateway or other hardware/software involved: (similar to PATTERN 8) Xlite SIP Phone (public IP) ----->(internet)-->Amazon NAT router-----> RADVision Proxy ...