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

[Sam Govind]

This obviously is pointing to NAT issue. see if you've configured both
asterisk servers with externip= PUBLICIPOFAsterisks.

Studying SIP traces on each console and specially looking at the SDPs in
INVITE will help you find out exact problem. I expect that one of the
asterisk box is sending the audio to its LAN/Private IP whereas it should be
sending RTPs to Public IP of other Asterisk.


On Fri, Sep 16, 2011 at 12:50 PM, Lee, John (Sydney) <John.Lee at compuware.com
> wrote:

> **
>
> 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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