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<TITLE>Inter-astersik dialling encounteres no audio</TITLE>
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<P DIR=LTR><SPAN LANG="en-au"></SPAN><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">I have been deploying Asterisk (open source PABX) in the company which I work.</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">So</FONT><FONT SIZE=2 FACE="Courier New"> 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.</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">I understand that the no-a</FONT><FONT SIZE=2 FACE="Courier New">udio means that SIP traffic (TCP/UDP 5060) is allowed to go through but not RTP (UDP 16384-32767).</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New"> </FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">Case A</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">======</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">This is a simplified diagram of how I am testing the dialling between 2 subnets.</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">In this case, phone A is registered in Asterisk A and phone</FONT><FONT SIZE=2 FACE="Courier New"></FONT></SPAN><SPAN LANG="en-au"></SPAN><SPAN LANG="en-au"> <FONT SIZE=2 FACE="Courier New">B</FONT></SPAN><SPAN LANG="en-au"></SPAN><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New"> is registered in Asterisk B.</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">Phone A <--> Asterisk A <--> Router A <<==>> WAN <<==>> Router B <--> Asterisk B <--> Phone B </FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New"> </FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">Case B</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">======</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">However, before I have tested successfully using this kind of connection.</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">In this case, phone B1 and B2 are registered in Asterisk B although they are on different subnets.</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">Both phone B1 and B2 can ring and audio is allowed to pass through.</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">Phone B1 <--> Router A <<==>> WAN <<==>> Router B <--> Asterisk B <--> Phone B2</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New"> </FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">I am</FONT> <FONT SIZE=2 FACE="Courier New">mystified why audio is allowed go through in case B but not case A.</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New"> </FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">Can someone be kind enough to help me to understand why I have this problem?</FONT></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">If the router is blocking RTP traffic, then why is that I have no audio problem in case B?</FONT></SPAN><SPAN LANG="en-au"></SPAN><SPAN LANG="en-au"></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"><FONT SIZE=2 FACE="Courier New">Thanks in advance.</FONT></SPAN><SPAN LANG="en-au"></SPAN><SPAN LANG="en-au"></SPAN></P>
<P DIR=LTR><SPAN LANG="en-au"></SPAN><SPAN LANG="en-au"></SPAN></P>
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