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<DIV dir=ltr align=left><SPAN class=893172721-03012006><FONT face=Arial
color=#0000ff size=2>We cheated with ours.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=893172721-03012006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=893172721-03012006><FONT face=Arial
color=#0000ff size=2>One our ECS we did a "change dialplan parameters" and added
routing pattern 31 to the "ETA"</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=893172721-03012006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=893172721-03012006><FONT face=Arial
color=#0000ff size=2>Once this was done we programmed routing pattern 31 to send
via the appropriate signaling/trunk group out to Asterisk. The end result
was that if the Avaya didnt know the extension from its own tables, it sent it
to Asterisk to handle.</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=893172721-03012006><FONT face=Arial
color=#0000ff size=2></FONT></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=893172721-03012006></SPAN> </DIV>
<DIV align=left><FONT face=Arial size=2></FONT> </DIV>
<DIV align=left><FONT face=Arial size=2>Jonathan O'Connor</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Senior System Administrator</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Inoveris LLC</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Direct Line (614)
791-3742</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Fax (614) 791-3748</FONT></DIV>
<DIV align=left><FONT face=Arial size=2>Helpdesk <FONT
color=#0000ff>866-456-1566</FONT></FONT></DIV>
<DIV align=left> </DIV>
<DIV align=left><FONT face=Arial size=2></FONT> </DIV>
<DIV> </DIV><BR>
<BLOCKQUOTE
style="PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #0000ff 2px solid; MARGIN-RIGHT: 0px">
<DIV class=OutlookMessageHeader lang=en-us dir=ltr align=left>
<HR tabIndex=-1>
<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Alyed
Tzompa<BR><B>Sent:</B> Tuesday, January 03, 2006 1:53 PM<BR><B>To:</B>
kristian@netatonce.se; asterisk-users@lists.digium.com<BR><B>Subject:</B> Re:
[Asterisk-Users] Asterisk PRI problems.<BR></FONT><BR></DIV>
<DIV></DIV><FONT face=arial size=2>It should be possible, but I think that the
problem lies in the connection between Avaya and Asterisk, let me explain:
your Avaya most likely uses different<BR>connection parametrs for handling the
calls coming from /going to PSTN and the ones </FONT><FONT face=arial
size=2>coming from /going to Asterisk. </FONT><FONT face=arial
size=2><BR><BR>If you need to pass let's say extensions' 20XX traffic, from
Avaya through Asterisk, just set this in the Avaya and let it dial to Asterisk
as if it was another<BR>extension, once in the Asterisk, you can handle it
with the extensions.conf<BR><BR>Alyed </FONT><BR><BR><FONT
face="Tahoma, Arial, Sans-Serif" size=2>
<HR align=center width="100%" SIZE=2>
Return-Path: <kristian@netatonce.se> Tue Jan 03 00:22:49
2006<BR>Received: from mailfront1.citynet.nu [217.10.96.36] by
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08:22:45 +0100 (CET)<BR>Date: Tue, 3 Jan 2006 10:19:45 +0100<BR>From: Kristian
Larsson <kristian@netatonce.se><BR>To:
alyed.tzompa@simitel.com,<BR>Asterisk Users Mailing List - Non-Commercial
Discussion <asterisk-users@lists.digium.com><BR>Subject: Re:
[Asterisk-Users] Asterisk PRI problems.<BR>Message-ID:
<20060103091945.GA21352@juniks.net><BR>References:
<05db140eaa38451c887627d047a03e4d@simitel.com><BR>Mime-Version:
1.0<BR>Content-Type: text/plain; charset=iso-8859-15<BR>Content-Disposition:
inline<BR>In-Reply-To:
<05db140eaa38451c887627d047a03e4d@simitel.com><BR>User-Agent:
Mutt/1.5.8i<BR>X-Virus-Scanned: by amavis at citynet.nu<BR>X-Spam-Status: No,
hits=-4.5 tagged_above=-999.0 required=999.0<BR>tests=ALL_TRUSTED,
BAYES_00<BR>X-Spam-Level: <BR>X-SmarterMail-Spam: SPF_None</FONT><BR><BR>On
Mon, Jan 02, 2006 at 12:09:33PM -0700, Alyed Tzompa wrote:<BR>> You are not
gonna be able to modify this behaviour from the asterisk since in your case
asterisk is only receiving the digits from someone else (an Avaya in your case
but could be PSTN for instance)<BR>> <BR>> Just asked an Avaya support
guy and told me you should take a look at the ARS Digit Analysis Table, and
modify the Min and Max values to suit your needs.<BR>I just found something
called "digit delay time"<BR>and "digit delay count", it just happens so
that<BR>digit delay count is set to four, ie the number of<BR>digits before
the avaya dials out. I can not try<BR>it out right now since the system is
being used -<BR>has to wait for the maintenance window
tomorrown<BR>evening.<BR><BR>But what I find a little strange is that the
Avaya<BR>works when connected to my telco, shouldn't it be<BR>possibly to
simulate a telco fully with Asterisk?<BR><BR>Thanks
Alyed!<BR><BR>Kristian.<BR><BR><BR><BR>> Hope this helps<BR>>
ww6<BR>> <BR>> ----------------------------------------<BR>>
Return-Path: <ASTERISK-USERS-BOUNCES>Mon Jan 02 05:56:31 2006<BR>>
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X-Original-To: asterisk-users@lists.digium.com<BR>> Delivered-To:
asterisk-users@lists.digium.com<BR>> Date: Mon, 2 Jan 2006 15:52:51
+0100<BR>> From: Kristian Larsson <KRISTIAN><BR>> To:
asterisk-users@lists.digium.com<BR>> Message-ID:
<20060102145251.GD9366@juniks.net><BR>> Mime-Version: 1.0<BR>>
Content-Type: text/plain; charset=iso-8859-15<BR>> Content-Disposition:
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<KRISTIAN>[90/4]<BR>> Subject: [Asterisk-Users] Asterisk PRI
problems.<BR>> X-BeenThere: asterisk-users@lists.digium.com<BR>>
X-Mailman-Version: 2.1.5<BR>> Precedence: list<BR>> Reply-To: Asterisk
Users Mailing List - Non-Commercial Discussion<BR>>
<ASTERISK-USERS><BR>> List-Id: Asterisk Users Mailing List - Non-Commercial
Discussion<BR>> <ASTERISK-USERS.LISTS.DIGIUM.COM><BR>> List-Unsubscribe:
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@lists.digium.com?subject="unsubscribe"><BR>> List-Archive: <HTTP:><BR>>
List-Post: <MAILTO:ASTERISK-USERS @lists.digium.com=""><BR>> List-Help:
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@lists.digium.com?subject="subscribe"><BR>> Sender:
asterisk-users-bounces@lists.digium.com<BR>> Errors-To:
asterisk-users-bounces@lists.digium.com<BR>> X-SmarterMail-Spam:
SPF_None<BR>> X-Rcpt-To: <ALYED.TZOMPA @simitel.com=""><BR>> <BR>> I
have an Avaya IP Office PBX connected to an<BR>> Asterisk system via a
Sangoma ISDN PRI card.<BR>> Dialing from the as<BR>> terisk system into
the avaya works just fine but<BR>> when trying to call from a phone
connected to the<BR>> avaya syste<BR>> m something goes wrong. After
punching the first<BR>> four digits the Avaya calls out, shouldn't it
wait<BR>> for all di<BR>> gits and then dial out?<BR>> If I try to
dial a three digit number it waits for<BR>> a while then dials.<BR>>
<BR>> Is this some feature to let the CO know of which<BR>> area code
the calls is going ahead of time?<BR>> Is there some way to circumvent this
using hacks<BR>> on the asterisk side?<BR>> <BR>> -- <BR>>
Kristian Larsson, Net At Once AB<BR>> Email: kristian@netatonce.se<BR>>
Phone: +46 470 592717<BR>> Cell: +46 704 910401<BR>>
_______________________________________________<BR>> --Bandwidth and
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<BR><BR>> _______________________________________________<BR>>
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Asterisk-Users mailing list<BR>> To UNSUBSCRIBE or update options
visit:<BR>>
http://lists.digium.com/mailman/listinfo/asterisk-users<BR><BR><BR>--
<BR>Kristian Larsson, Net At Once AB<BR>Email: kristian@netatonce.se<BR>Phone:
+46 470 592717<BR>Cell: +46 704
910401<BR><BR></BLOCKQUOTE></ALYED.TZOMPA></MAILTO:ASTERISK-USERS-REQUEST></HTTP:></MAILTO:ASTERISK-USERS-REQUEST></MAILTO:ASTERISK-USERS></HTTP:></MAILTO:ASTERISK-USERS-REQUEST></HTTP:></ASTERISK-USERS.LISTS.DIGIUM.COM></ASTERISK-USERS></KRISTIAN></KRISTIAN></ASTERISK-USERS></ASTERISK-USERS></ASTERISK-USERS></ASTERISK-USERS></ASTERISK-USERS-BOUNCES></BODY></HTML>