<span style="font-family: monospace;">Reorder tone can be used for many things, is there anything I've missed?<br><br></span><pre><br> 7.4.2 401 Unauthorized .................................... 78<br> 7.4.4 403 Forbidden ....................................... 78
<br> 7.4.5 404 Not Found ....................................... 78<br> 7.4.6 405 Method Not Allowed .............................. 78<br> 7.4.7 406 Not Acceptable .................................. 79
<br> 7.4.11 410 Gone ............................................ 79<br> 7.4.16 420 Bad Extension ................................... 80<br> 7.4.17 480 Temporarily Unavailable ......................... 80
<br> 7.4.18 481 Call Leg/Transaction Does Not Exist ............. 81<br> 7.4.19 482 Loop Detected ................................... 81<br> 7.4.20 483 Too Many Hops ................................... 81
<br> 7.4.21 484 Address Incomplete .............................. 81<br> 7.4.22 485 Ambiguous ....................................... 81<br> 7.4.23 486 Busy Here ....................................... 82
<br> 7.5 Server Failure 5xx .................................. 82<br> 7.5.1 500 Server Internal Error ........................... 82<br> 7.5.2 501 Not Implemented ................................. 82
<br> 7.5.3 502 Bad Gateway ..................................... 82<br> 7.5.4 503 Service Unavailable ............................. 83<br> 7.5.5 504 Gateway Time-out ................................ 83
<br> 7.5.6 505 Version Not Supported ........................... 83<br> 7.6 Global Failures 6xx ................................. 83<br> 7.6.1 600 Busy Everywhere ................................. 83
<br> 7.6.2 603 Decline ......................................... 84<br> 7.6.3 604 Does Not Exist Anywhere ......................... 84<br> 7.6.4 606 Not Acceptable .................................. 84
</pre>All of these are defined by RFC2543. 183 is not defined until 2 years later.<br><br>Do you have any examples where ringing is indicated and it should not be? I would really like to know, I am not trying to say you are wrong, I've must have never encountered such a situation, if a recorded message is played from the far switch, the audio should be passed, if it tone is played that is legacy pstn if its over the network or the near end such as a PBX generating the tone, anything that is digitally interconnected to a proper ss7 network, be it an ISDN line, PRI or SIP provider, should pass proper progress out of band. If you are using analog lines then get rid of progressinband configurations and do as Mr. Wieling suggests.
<br><br><div><span class="gmail_quote">On 12/11/06, <b class="gmail_sendername">Douglas Garstang</b> <<a href="mailto:dgarstang@oneeighty.com">dgarstang@oneeighty.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div>
<div><span><font color="#0000ff" face="Arial" size="2">Andrew,</font></span></div>
<div><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div><span><font color="#0000ff" face="Arial" size="2">I
don't think it's a Polycom issue. We took Asterisk out of the picture and had
our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike
Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session
Progress, and the polycom's play the correct tones in this
case.</font></span></div>
<div><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div><span><font color="#0000ff" face="Arial" size="2">We
WANT Asterisk to send progress tones in band. In our case it IS needed.
What's the SIP response for a reorder then if we don't need in band progress
tones? There is none. In a situation where the PSTN end sends back a reorder, or
some other unusual tone, all the UA ends up hearing is the closest SIP
approximation, which is ringing, which is not correct.</font></span></div>
<div><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div><span><font color="#0000ff" face="Arial" size="2">I have
tried to explain my issues in detail in this list in the past, and I have
invariably met with responses like 'I don't understand' or 'why would you want
to do that?'. I get much better understanding of my issues, and therefore better
replies, when I break the problem down and only explain the relevant
portions.</font></span></div>
<div><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div><span><font color="#0000ff" face="Arial" size="2">I
really don't appreciate your tone.</font></span></div><span class="sg">
<div><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div><span><font color="#0000ff" face="Arial" size="2">Douglas.</font></span></div></span><div><span class="e" id="q_10f73fd2df688e45_2">
<div><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div><font face="Tahoma"><font size="2"><span><font color="#0000ff" face="Arial"> </font></span></font></font></div>
<div><font face="Tahoma"><font size="2"><span> </span>-----Original Message-----<br><b>From:</b>
Andrew Joakimsen [mailto:<a href="mailto:joakimsen@gmail.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">joakimsen@gmail.com</a>]<br><b>Sent:</b> Monday, December
11, 2006 4:40 PM<br><b>To:</b> Asterisk Users Mailing List - Non-Commercial
Discussion<br><b>Subject:</b> Re: [asterisk-users] Asterisk Sends 180-RINGING to
UA even withprogressinband=yes<br><br></font></font></div>
<blockquote style="border-left: 2px solid rgb(0, 0, 255); padding-left: 5px; margin-left: 5px;">When
we send 183, that means 'inband progress' is available. That does _not_
necessarily mean that it is ringing, it could be any sort of progress tone, or
even audio from an IVR. If your ATA does not stop its own ringing generator
and start forwarding the audio, it is broken.<br><br>It is my understanding
that Polycom's SIP implemenation does not currectly handle these responses.
See: <a href="http://bugs.digium.com/view.php?id=3129" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">http://bugs.digium.com/view.php?id=3129</a><br><br>In
the future it would help that instead of nitpicking some little low level
technical detail you describe what your actual problem is, you would get more
input that way. progessinband=yes means that the call progress WILL BE SEND
INBAND, which in 99% of cases is not needed, and does not make sense. You are
also wasting additinal resources because asterisk must generate progress tones
too. <br><br>
<div><span class="gmail_quote">On 12/11/06, <b class="gmail_sendername">Douglas
Garstang</b> <<a href="mailto:dgarstang@oneeighty.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">dgarstang@oneeighty.com</a>>
wrote:</span>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">I
have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my
polycom phones and then it also sends 183-Session Progress. That doesn't
seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if
progressinband=yes ?
<br><br>Doug.<br>_______________________________________________<br>--Bandwidth
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</div><br></blockquote></span></div></div>
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