[asterisk-users] Inter-astersik dialling encounteres no audio

[Lee, John (Sydney)]

I have been deploying Asterisk (open source PABX) in the company which I
work.

So far, all the Asterisk servers do not really talk to each other.
Recently, I am experimenting to dial from one Asterisk server to another
through the WAN and I encountered a no-audio problem although the
callee's phone can ring.
I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is
allowed to go through but not RTP (UDP 16384-32767).
 
Case A
======
This is a simplified diagram of how I am testing the dialling between 2
subnets.
In this case, phone A is registered in Asterisk A and phone B is
registered in Asterisk B.

Phone A <--> Asterisk A <--> Router A <<==>> WAN <<==>> Router B <-->
Asterisk B <--> Phone B   
 
Case B
======
However, before I have tested successfully using this kind of
connection.
In this case, phone B1 and B2 are registered in Asterisk B although they
are on different subnets.
Both phone B1 and B2 can ring and audio is allowed to pass through.

Phone B1 <--> Router A <<==>> WAN <<==>> Router B <--> Asterisk B <-->
Phone B2
 
I am mystified why audio is allowed go through in case B but not case A.
 
Can someone be kind enough to help me to understand why I have this
problem?
If the router is blocking RTP traffic, then why is that I have no audio
problem in case B?

Thanks in advance.


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