Asterisk will send the two SIP endpoints 'reinvite' messages, so that they talk RTP directly with each other. Depending on your version of Asterisk, setting the 'canreinvite' or 'directmedia' option may make a difference, since that will keep the traffic flowing through the servers, and the phones will not need to reach each other directly.
From: [email protected] [mailto:[email protected]] On Behalf Of Lee, John (Sydney) Sent: Friday, September 16, 2011 3:51 AM To: [email protected] Subject: [asterisk-users] Inter-astersik dialling encounteres no audio 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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