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) <[email protected] > 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. > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
