Displaying 20 results from an estimated 2000 matches similar to: "Invalid DivertingLegInformation2 component received 0x38"
2010 Sep 24
1
RDNIS not passed from one box to another with BRI access
Hi,
I've configured a new Asterisk 1.6.1 box to replace an old Bristuffed 1.2
Asterisk.
Since then, it happens that forwarded calls are not presented the way they
used to be.
It seems that now, some endpoints are displaying the original caller id
(that's what I'm trying to achive), while some are displaying the
redirecting number :
so if A calls B, B forwards to C
depending on where
2010 Jun 19
0
OT - Explain RDNIS
Hi,
1. Can you explain what RDNIS is when it should used with Asterisk ?
I've read this http://www.voip-info.org/wiki/view/RDNIS but I'm still
wondering if I should use it.
My understanding is:
when dialing out through an ISDN line, Asterisk is sending two numbers
(using signalling channel) :
- one is CALLERID(num) which is supposed to be whatever the Asterisk admin
whishes to be,
- one
2010 May 15
1
q931.c modifications for CLID Presentation
Hi Guys,
We have a problem with Caller ID not being displayed. I want to test
everything to see where the problem is with the incoming Caller ID and why
it's not displaying.
I am tracking this down to "Presentation prohibited of network provided
number" even though the Caller doesn't use *67 and even though they haven't
asked their provider to block their CLID for outbound.
2006 Apr 28
2
Asterisk DNID/RDNIS with Dial iax2
Dear Asterisk-Users:
Question:
========
How do I get asterisk to pass DNID/RDNIS information between
asterisk machines using iax2, in a Dial(IAX2...) command ?
Setup:
=====
I have two asterisk boxes, MASTER and SLAVE. MASTER is running
1.2.0 and SLAVE is running 1.2.1. The main box handles incoming calls
on a multiple lines (both via hardware connection to our internal PBX
and calls
2010 Aug 11
1
Youmail RDNIS
Does anyone know the mechanism by which companies like YouMail (and MNO's
using their own voicemail system) are able to redirect ALL calls from a ALL
subscribers to *just one* voicemail DID, yet determine WHICH subscriber did
the redirection?
I had always assumed this was simply done using RDNIS. In other words, the
original calling party's CallerID is passed with the redirected
2008 Sep 23
0
PRI incoming call forward / call redirect
Good morning,
I have a Bell Canada PRI here (switchtype=national) and I am trying to perform
a call-forward-unconditional on one of the DIDs.
The idea is that when DID 5551234 receives a call, Asterisk redirects it back
out the same PRI to some external number.
This is simple enough to do with something along these lines:
[PRI]
exten => 5551234,1,Set(CALLERID(RDNIS)=${EXTEN})
exten =>
2006 Dec 30
0
Theory behind RDNIS and does it work or not?
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
<title></title>
</head>
<body bgcolor="#ffffff" text="#000000">
Hello everybody,<br>
<br>
currently I'm implementing redirection
2014 Aug 28
1
RDNIS with tel: vs. sip: header
Has anyone had success patching chan_sip.c so that Asterisk will recognize
the tel: header for RDNIS information?
exten = get_in_brackets(tmp);
if (!strncasecmp(exten, "sip:", 4)) {
exten += 4;
} else if (!strncasecmp(exten, "sips:", 5)) {
exten += 5;
} else {
ast_log(LOG_WARNING, "Huh? Not an
2004 Jul 15
2
Cisco phones and Messages and Forward ToVM keys
; Below assumes you are using the same number for Voicemail boxes as
extensions
; if ${RDNIS} is blank then GotoIf will go to extension 2, otherwise it
will go to extension 102
exten => 8500,1,GoToIf($[X${RDNIS} = X]?2:102)
exten => 8500,2,VoiceMailMain(s${CALLERIDNUM})
exten => 8500,3,Hangup
exten => 8500,102,VoiceMail(u${RDNIS})
exten => 8500,103,Hangup
; you should now be able
2009 Dec 02
2
Variable Name needed
Other than having stripping out IPs this is what I am receiving for my
voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works
fine calls that come in on my PRI. BUT at least from this VOIP source
the To field which is my RDNIS information for these calls, doesn't
actually fill into ${CALLERID(rdnis). But as you can see I'm getting the
information.
My question is, Does
2009 Dec 02
0
FW: Variable Name needed
It might be worth mentioning the voip call is coming from a number we
have thru bandwidth.com in case anyone uses them.
James Shigley
From: James A. Shigley
Sent: Wednesday, December 02, 2009 3:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Variable Name needed
That wasn't it either. I tried a few other likely fields from
2004 Oct 26
2
RDNIS
I'm trying to use RDNIS with asterisk, and I don't appear to be
receiving any information (the value is blank). The upstream who
provides the PRI says they are passing all the info through, I don't see
this value coming across. I've tried it with a Verizon call forward, as
well as a Nextel with the same results for both. I'm trying to use this
for Voicemail. I'm using
2009 Mar 24
0
Issue with RDNIS
Hello,
Does anyone know why I am unable to retrieve the "Redirecting Number"?
I've done a "pri debug span 1/1" and can see the number being passed
correctly to Asterisk:
< Redirecting Number (len=15) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
< Ext: 0 Presentation: Presentation allowed
of
2007 Feb 09
0
Asterisk 1.2.14 - Chanspy, sound issues.
I upgraded my Asterisk system to version 1.2.14 to check if the sound
quality issues I was having with Chanspy in 1.2.7 remained. I'm still
getting them, and I'm honestly out of ideas except from RTFS.
The called party sounds normally fine, but it's impossible to hear the
caller. Sometimes, when the called party is talking, the caller can also
be heard. The conversation sounds broken,
2006 Dec 09
2
RDNIS question
Perhaps I've got the whole concept wrong, but here goes:
Using 1.4, when someone from the outside dials my direct line (123456),
I want it to call my extension at work (SIP/456), my extension in my
home office (vpn connection to corporate lan, SIP/678) and my mobile
(654321). So my dialplan is thus:
exten => 123456,1,Dial(SIP/456&SIP/678&Zap/G3c/07803654321,30)
exten =>
2006 Jan 14
4
Ugly echo cancel, with Bristuff/Zaphfc
I'm using bristuffed Asterisk with ISDN/ZAPHFC
I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in
zapata.conf, but without echocancel I have bad (incoming) echo
Through PSTN/FXO sound is ok with or without echocancel.
I tried other echo cancellers (in zconfig.h) two times:
ECHO_CAN_KB1 (this was default)
ECHO_CAN_MARK2
ECHO_CAN_MG2
after any change I compiled (make
2010 Apr 11
1
Asterisk in Debian/Lenny without Junghanns.net support?
Hi!
Asterisk in Debian/Lenny claims to be bristuffed, not? At least the
the Debian patch tracking system shows the bristuff-patches:
[1] http://bit.ly/bRRHe7
We have a QuadBRI-Card and recently needed support from Junghanns.net
but they refused telling us there is no bristuff installed because of
the show version output:
*CLI> show version
Asterisk 1.4.21.2~dfsg-3+lenny1 built by pbuilder @
2004 Jun 04
1
Voicemail and Cisco phones: Dialplan example
Assume you have the messages button on your Cisco phone set to dial
3009. Here's an sample dialplan entry that will make the "DND" and
"ToVM" and "Messages" button work as expected. This should work for
both -stable and -head.
exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4)
exten => 3009,2,VoicemailMain()
exten => 3009,3,Hangup
exten =>
2006 Jan 25
1
ISDN D-channel disconnects for a minute every 5 minutes
I have a problem with Asterisk-bristuffed using a zaphfc card.
I am located in the Netherlands, so I have an ISDN line from KPN. When I
start Asterisk, and plug in the ISDN line, everything works perfectly for
about 5 minutes. And then the ISDN line is down for 1 minute, and after that
minute, the line comes back up and works for another 5 minutes. Every time
the line goes down I get the error
2005 Jan 28
1
1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping
I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a
call with '*8' - the call will drop after about 20 or so seconds. Is
this a general problem with Asterisk 1.0.2?
As this is the latest release that it appears Klaus-Peter Junghanns has
for public consumption - is there anything I can patch for just this
problem - or has Klaus-Peter Junghanns (or anyone else) been