<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
Hi!<br>
<br>
Thank you all for the info!<br>
<br>
But I think I haven't explained my scenario well enough. <br>
<br>
I am not relaying the calls to SIP.<br>
<br>
What happens is the following (the scenario is: a call started from an
ISDN E1 PBX, through the asterisk, to the PSTN, and was answered by the
remote party):<br>
<br>
1. A CONNECT comes from the PSTN containing a Display IE (which has
info sent by the telco that is used for billing purposes by the PBX)
into span 1 of the asterisk. (The telco is emulating an old billing
procedure based on an impulse count. This was commonly used in analog
lines.)<br>
<br>
2. The asterisk relays the call to span 2 (which is connected to the
PBX).<br>
<br>
3. The CONNECT that is sent from span 2 to the PBX does not have the
Display IE. The asterisk strips this IE from the CONNECT message.<br>
<br>
<br>
This is my problem. Is there a way that i can force the asterisk not to
strip the Display IE?<br>
<br>
Thanks and best regards to all,<br>
<br>
Óscar Patrício<br>
<br>
<br>
Anthony Francis wrote:
<blockquote cite="mid46A77282.30402@rockynet.com" type="cite">
<pre wrap="">Damon Estep wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Try putting a 1 second wait as step 1 in the dialplan, the SIP invite is probably being send before the display IE arrives. The display IE is used for CNAM delivery, and should not exceed 15 characters.
It is very common to put a message in the display IE that indicates that the CNAM info will be sent in a subsequent Facility IE, and for that you must wait 1 second.
If the ISDN setup actually contains information in the display IE, and that is not being captured as the CNAM (callerid(name)) you might need to capture he ISDN messaging to debug it, the telco can usually provide such a trace.
I would bet that the display IE contains a information following message, and what you really want is in the facility IE that follows. Very common, as is the Wait(1) workaround.
-----Original Message-----
From: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a> [<a class="moz-txt-link-freetext" href="mailto:asterisk-users-bounces@lists.digium.com">mailto:asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Oscar Patricio
Sent: Wednesday, July 25, 2007 6:55 AM
To: <a class="moz-txt-link-abbreviated" href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>
Subject: [asterisk-users] Display IE
Hi!
I have an Asterisk Box that has 2 E1 connections: one to the PSTN and
one to a PBX. It is acting as a telephony gateway. I have a problem: the
PSTN sends information in the Display IE (in setup, information ,
etc.messages) that the PBX needs por internal processing.
The asterisk does not relay the ISDN frames coming from the PSTN. It
regenerates them, and when it does, it ommits the Display IE.
Is there a way that i can force the asterisk, not to ignore this IE in
the ISDN messages?
If anyone can shed some light on this issue, i would be very grateful.
Thanks in advance,
Best regards,
Óscar Patrício
_______________________________________________
--Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a>--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
_______________________________________________
--Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a>--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<pre wrap=""><!---->Or trace it yourself with:
pri intense debug span 1
Make sure you change the 1 to whatever span these calls are coming across.
Anthony
_______________________________________________
--Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a>--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
<br>
</body>
</html>