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

[John Novack]

Lee, John (Sydney) wrote:
>
> I have been deploying Asterisk (open source PABX) in the company which I work.
>
> Sofar, 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 phoneBis 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.
>
>
Why not use IAX????

John Novack

-- 

Dog is my Co-pilot

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