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Lee, John (Sydney) wrote:
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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
face="Courier New" size="2">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 face="Courier New" size="2">So</font><font
face="Courier New" size="2"> 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 face="Courier New" size="2">I
understand that the no-a</font><font face="Courier New"
size="2">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 face="Courier New" size="2"> </font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">Case
A</font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">======</font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">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 face="Courier New" size="2">In
this case, phone A is registered in Asterisk A and phone</font></span><span
lang="en-au"></span><span lang="en-au"> <font face="Courier
New" size="2">B</font></span><span lang="en-au"></span><span
lang="en-au"><font face="Courier New" size="2"> is registered
in Asterisk B.</font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">Phone
A <--> Asterisk A <--> Router A
<<==>> WAN <<==>> Router B
<--> Asterisk B <--> Phone B </font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2"> </font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">Case
B</font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">======</font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">However,
before I have tested successfully using this kind of
connection.</font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">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 face="Courier New" size="2">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 face="Courier New" size="2">Phone
B1 <--> Router A <<==>> WAN
<<==>> Router B <--> Asterisk B <-->
Phone B2</font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2"> </font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">I
am</font> <font face="Courier New" size="2">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 face="Courier New" size="2"> </font></span></p>
<p dir="LTR"><span lang="en-au"><font face="Courier New" size="2">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 face="Courier New" size="2">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 face="Courier New" size="2">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>
<br>
</blockquote>
Why not use IAX????<br>
<br>
John Novack<br>
<br>
<pre class="moz-signature" cols="10000">--
Dog is my Co-pilot</pre>
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