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<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>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.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>In the Asterisk config
sirrix.conf (http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sirrix.conf)
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.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span lang=EN-US
style='font-size:10.0pt;font-family:Arial;color:navy'>Today I tried Asterisk <st1:metricconverter
ProductID="1.2 in" w:st="on">1.2 in</st1:metricconverter> another Telco and I
experienced the same behavior. I’m starting to think this is a bug in the
Asterisk E1 signalization. <o:p></o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>De:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <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<br>
<b><span style='font-weight:bold'>Para:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Asunto:</span></b> Re: [Asterisk-Users] PRI:
This number has been disconnected</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>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. <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'> …<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>…<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'> <span class=gmailquote>On 12/29/05, <b><span
style='font-weight:bold'>Adam Goryachev</span></b> <</span></span><span
class=gmailquote><a href="mailto:mailinglists@websitemanagers.com.au"><span
lang=EN-US>mailinglists@websitemanagers.com.au</span></a></span></font><span
class=gmailquote><span lang=EN-US>> wrote:</span></span><span lang=EN-US> <o:p></o:p></span></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span lang=EN-US
style='font-size:12.0pt'>On Wed, 2005-12-28 at 14:00 -0300, <st1:PersonName
ProductID="Javier Ergas" w:st="on">Javier Ergas</st1:PersonName> 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
href="http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html"><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>
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