<div dir="ltr"><br><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Jul 10, 2014 at 12:36 PM, Justin Killen <span dir="ltr"><<a href="mailto:jkillen@allamericanasphalt.com" target="_blank">jkillen@allamericanasphalt.com</a>></span> wrote:<br>
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<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy">Okay, that makes sense. So then, how does
the answer()’d status affect that? From my (admittedly limited) experience, it
seems that playing in-band tones only works if the call has been answered?<u></u><u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy">Here’s the setup I’m using, maybe it will
help if I give an example of what I’m trying to figure out:<u></u><u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy">Phone A <-sip-> asterisk site A
<-PRI-> PTSN <-PRI-> asterisk site B<u></u><u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy">If then I have a dialplan at site B for
the dialed DID that looks like this:<u></u><u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy">Playtone(busy)<u></u><u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy">Busy(20)<u></u><u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy"><u></u> <u></u></span></font></p>
<p class="MsoNormal"><font color="navy" face="Arial"><span style="font-size:10pt;font-family:Arial;color:navy">And given that I’m using default in-band
tones, what is the expected resulting sound coming out of Phone A?</span></font></p></div></div></blockquote><div><br></div><div>Inband tones before answering is known as early media. After answering<br>it is just a normal audio path that the tones would go over.<br>
<br></div><div>Asterisk B in the above situation before answering with the shown dialplan<br>snippet, the Playtone() would effectively be a NoOp because nothing is shown<br>that would open a path for early media. (The Progress() application should be<br>
done first to open the path.) Then the Busy() would cause chan_dahdi/sig_pri<br>to send a PROGRESS message toward Asterisk A if needed to open the media<br>path and start its own busy tones. From Asterisk B to Phone A there are<br>
several things that could block the early media. The telco network,<br>Asterisk A configuration, and Phone A itself.<br><br></div><div>You may need to investigate these options on Asterisk A from sip.conf.sample:<br>;prematuremedia=no ; Some ISDN links send empty media frames before <br>
; the call is in ringing or progress state. The SIP <br> ; channel will then send 183 indicating early media<br> ; which will be empty - thus users get no ring signal.<br>
; Setting this to "yes" will stop any media before we have<br> ; call progress (meaning the SIP channel will not send 183 Session<br> ; Progress for early media). Default is "yes". Also make sure that<br>
; the SIP peer is configured with progressinband=never. <br> ;<br> ; In order for "noanswer" applications to work, you need to run<br>
; the progress() application in the priority before the app.<br><br>;progressinband=never ; If we should generate in-band ringing always<br> ; use 'never' to never use in-band signalling, even in cases<br>
; where some buggy devices might not render it<br> ; Valid values: yes, no, never Default: never<br></div><div><br></div><div>Richard<br></div></div><br></div>
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