<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 12 (filtered medium)"><title>Inter-astersik dialling encounteres no audio</title><style><!--
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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoPlainText>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.<o:p></o:p></p><p class=MsoPlainText><o:p> </o:p></p><p class=MsoPlainText><o:p> </o:p></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] <b>On Behalf Of </b>Lee, John (Sydney)<br><b>Sent:</b> Friday, September 16, 2011 3:51 AM<br><b>To:</b> asterisk-users@lists.digium.com<br><b>Subject:</b> [asterisk-users] Inter-astersik dialling encounteres no audio<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>I have been deploying Asterisk (open source PABX) in the company which I work.</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>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.</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is allowed to go through but not RTP (UDP 16384-32767).</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'> </span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>Case A</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>======</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>This is a simplified diagram of how I am testing the dialling between 2 subnets.</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>In this case, phone A is registered in Asterisk A and phone</span><span lang=EN-AU> </span><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>B is registered in Asterisk B.</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>Phone A <--> Asterisk A <--> Router A <<==>> WAN <<==>> Router B <--> Asterisk B <--> Phone B </span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'> </span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>Case B</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>======</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>However, before I have tested successfully using this kind of connection.</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>In this case, phone B1 and B2 are registered in Asterisk B although they are on different subnets.</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>Both phone B1 and B2 can ring and audio is allowed to pass through.</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>Phone B1 <--> Router A <<==>> WAN <<==>> Router B <--> Asterisk B <--> Phone B2</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'> </span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>I am</span><span lang=EN-AU> </span><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>mystified why audio is allowed go through in case B but not case A.</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'> </span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>Can someone be kind enough to help me to understand why I have this problem?</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>If the router is blocking RTP traffic, then why is that I have no audio problem in case B?</span><o:p></o:p></p><p><span lang=EN-AU style='font-size:10.0pt;font-family:"Courier New"'>Thanks in advance.</span><o:p></o:p></p></div></body></html>