I am having a similar problem, I thought it was because the PRI card is in another server that I connect to via IAX from my server, but we are seeing the same problem, ie getting a hangup instead of unavailable when calling a number that is not in service. I'm using T1 and Asterisk
1.21<br><br>
<div><span class="gmail_quote">On 12/28/05, <b class="gmail_sendername">Javier Ergas</b> <<a href="mailto:jergas@gmx.net">jergas@gmx.net</a>> wrote:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">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><a href="http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html">http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html</a><br><br>Is anyone experiencing the same behavior?<br><br><br>-----Mensaje original-----
<br>De: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] En nombre de Francesco
<br>Peeters (Asterisk)<br>Enviado el: Martes, 27 de Diciembre de 2005 20:09<br>Para: Asterisk Users Mailing List - Non-Commercial Discussion<br>Asunto: Re: [Asterisk-Users] PRI: This number has been disconnected<br><br>On Tue, December 27, 2005 23:37, Javier Ergas said:
<br>> Hi,<br>><br>><br>><br>> I'm running Asterisk@home 1.5 with TE110P E1 PRI in Chile.<br>><br>> When calling an invalid number using, I expect to hear:<br>><br>> "We're sorry you have reached a number which has been disconnected ..."
<br>><br>> And that is indeed what I hear when I dial out from [*] using analog FXO,<br>> or<br>> VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI<br>> interface however, I only hear the allison7/all-circuits-busy-now message.
<br>><br>><br>><br>> There was another issue like this in an old post<br>> (<a href="http://lists.digium.com/pipermail/asterisk-users/2004-April/043597.html">http://lists.digium.com/pipermail/asterisk-users/2004-April/043597.html
</a>)<br>> but I think it isn't the same.<br>><br>><br><SNIP><br><br>I believe this has to do with the AMP macro's being used in A@H. I am<br>seeing similar things.<br><br>For instance: One issue I have is that when a route has multiple trunks,
<br>and the first trunk after a while returns with 'NOANSWER', it merrily<br>continues to the next trunk, which is not quite the behavior I'd expect.<br>Especially as the primary trunk (IAX/VoipBuster) is *much* cheaper (ie
<br>free) as compared to the second trunk (Zap/g1), but the switch is made<br>without any message. This could mean that you might be talking to someone<br>on a different trunk, and instead of a free call, be paying normal fees.
<br><br>This could become expensive if you're calling the USA from Europe!...<br><br>I am currently looking in to ways to enhance those macro's to respond more<br>reliably, as well as return more useful information (busy tone on busy and
<br>no-answer, number disconnected info, etc.) when needed.<br><br>If I do get to a satifactory set of macro's, I will put them up on the<br>Wiki and let the list know... (I'm just starting on doing manual<br>configuring, so it will be a tough job to crack, but also a learning
<br>experience...)<br><br>--<br>F Peeters<br>PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch<br>2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0<br> Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
<br>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:
<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br><br>_______________________________________________<br>--Bandwidth and Colocation provided by
<a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list<br>To UNSUBSCRIBE or update options visit:<br> <a href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users
</a><br></blockquote></div><br>