I am using T1 Signaling and seeing the same problems (I think), so I don't think its just E1. <br><br>
<div><span class="gmail_quote">On 12/29/05, <b class="gmail_sendername">Javier Ergas</b> <<a href="mailto:jergas@gmx.net">jergas@gmx.net</a>> wrote:</span>
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<p><font face="Arial" color="navy" size="2"><span lang="EN-US" style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">I have tried both inband and outofband too unsuccessfully. I think the priindication parameter says how Asterisk reports Busy and Congestion to the PSTN, not the other way around.
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<p><font face="Arial" color="navy" size="2"><span lang="EN-US" style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">In the Asterisk config sirrix.conf (<a onclick="return top.js.OpenExtLink(window,event,this)" href="http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sirrix.conf" target="_blank">
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sirrix.conf</a>) there is a providetones parameter, witch I think handles the way that interface receives the signalization from the PSTN, but I think it won't work for zaptel/Zapata.
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<p><font face="Arial" color="navy" size="2"><span lang="EN-US" style="FONT-SIZE: 10pt; COLOR: navy; FONT-FAMILY: Arial">Today I tried Asterisk 1.2 in another Telco and I experienced the same behavior. I'm starting to think this is a bug in the Asterisk E1 signalization.
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<p><b><font face="Tahoma" size="2"><span style="FONT-WEIGHT: bold; FONT-SIZE: 10pt; FONT-FAMILY: Tahoma">De:</span></font></b><font face="Tahoma" size="2"><span style="FONT-SIZE: 10pt; FONT-FAMILY: Tahoma"> <a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">
asterisk-users-bounces@lists.digium.com</a> [mailto:<a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>
] <b><span style="FONT-WEIGHT: bold">En nombre de </span></b>Joe Pukepail<br><b><span style="FONT-WEIGHT: bold">Enviado el:</span></b> Jueves, 29 de Diciembre de 2005 15:22<span class="q"><br><b><span style="FONT-WEIGHT: bold">
Para:</span></b> Asterisk Users Mailing List - Non-Commercial Discussion<br><b><span style="FONT-WEIGHT: bold">Asunto:</span></b> Re: [Asterisk-Users] PRI: This number has been disconnected</span></span></font></p></div>
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<p><font face="Times New Roman" size="3"><span lang="EN-US" style="FONT-SIZE: 12pt">I have tried both inband and outofband, doesn't seem to make a difference. I added the congension and playtones(congestion) to the dial plan after the dial, but the users just get a busy instead of "Do-De-Dah The number of have reached is not in service <fastbusy>". PRI Debug below.
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<p><font face="Times New Roman" size="3"><span lang="EN-US" style="FONT-SIZE: 12pt"> <span>On 12/29/05, <b><span style="FONT-WEIGHT: bold">Adam Goryachev</span></b> <</span></span><span><a onclick="return top.js.OpenExtLink(window,event,this)" href="mailto:mailinglists@websitemanagers.com.au" target="_blank">
<span lang="EN-US">mailinglists@websitemanagers.com.au</span></a></span></font><span><span lang="EN-US">> wrote:</span></span><span lang="EN-US"> </span></p>
<p><font face="Times New Roman" size="3"><span lang="EN-US" style="FONT-SIZE: 12pt">On Wed, 2005-12-28 at 14:00 -0300, Javier Ergas wrote:<br>> I believe this behavior has nothing to do with the A@H Scripts. I think the
<br>> problem is in the PRI signalization.<br>> I can see the zap hangup messages when trying to call a disconnected number.<br>> .....<br>> -- Executing Dial("SIP/9349-1787", "ZAP/g0/2514990") in new stack
<br>> -- Called g0/2514990<br>> -- Channel 0/2, span 1 got hangup<br>> -- Hungup 'Zap/2-1'<br>> == No one is available to answer at this time<br>> -- Executing Goto("SIP/9349-1787", "s-NOANSWER|1") in new stack
<br>> -- Goto (macro-dialout-trunk,s-NOANSWER,1)<br>> ....<br>> The telco says they are sending inband information with the status of the<br>> call, but Asterisk is hanging up the channel instead of connecting it to let
<br>> hear the audio message.<br>><br>> There is a post with a similar issue here:<br>> </span><a onclick="return top.js.OpenExtLink(window,event,this)" href="http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html" target="_blank">
<span lang="EN-US">http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html </span></a></font><span lang="EN-US"><br>><br>> Is anyone experiencing the same behavior?<br>><br><br>Sounds like the difference between doing inband signalling or out of
<br>band signalling. I think by default, a PRI uses out of band signalling, <br>ie, it just sends a message saying "this number if un reachable" so<br>asterisk just hangs up and plays the local congestion dialplan.
<br><br>What you need to do is use inband signalling, so that asterisk won't <br>hangup, and instead will pass the audio from the telco through.<br><br>See /etc/asterisk/zapata.conf:<br>; PRI Out of band indications.<br>; Enable this to report Busy and Congestion on a PRI using out-of-band
<br>; notification. Inband indication, as used by Asterisk doesn't seem to <br>work<br>; outofband: Signal Busy/Congestion out of band with<br>RELEASE/DISCONNECT<br>; inband: Signal Busy/Congestion using in-band tones
<br>priindication = outofband<br><br><br>Regards,<br>Adam<br><br>_______________________________________________ <br>--Bandwidth and Colocation provided by </span><a onclick="return top.js.OpenExtLink(window,event,this)" href="http://easynews.com/" target="_blank">
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