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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