Displaying 20 results from an estimated 800 matches similar to: "Trim the RDNIS"
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
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
2005 May 20
1
RDNIS (DNID) Call Routing
I haven't been able to find much support for the RDNIS or DNID variables
online.
I am trying to prove a concept of call routing before we move towards
development of a production system. I need to have calls routed coming into
a call center based on DNIS. What type of syntax is needed in the
extensions.conf file and how can I test it with a softphone (ie: can I
emulate the DNIS from xlite)?
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
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
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
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
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 =>
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
2011 May 20
1
SIP Diversion RDNIS - how to get reason parameter?
Hi out there
To play the correct announcement in app_voicemail I whould be able to read the
SIP Diversion Reason which ist sent by another PBX:
Invite contains:
Diversion: <sip:+41315995003 at 157.161.10.190>;reason=no-
answer;privacy=off;counter=1
Asterisk Logs:
RDNIS for this call is is +41315995003 (reason no-answer;privacy=off;counter=1)
From what I see in the source of chan_sip
2010 Jan 10
2
app_swift 1.6.2 DTMF issue
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you
enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF}
is [un]set in an odd way.
for example consider:
999,1,Swift(some long message that you dont want to wait for|5000|5)
999,n,NoOp(DTMF: ${SWIFT_DTMF})
if while I am listening to the playback, i interrupt and dial:
- "12345", SWIFT_DTMF is set to
2010 Nov 13
2
asterisk 1.8 fax woes
I upgraded from a perfectly working 1.6.2 asterisk installation
(including fax via app_fax_digium) to 1.8.0 this evening.
All my custom modules (including swift <thanks darren!>) are working
fine except for fax.
When a caller connects, asterisk switches to the fax context and hangs
up the call.
i've captured with:
core set verbose 10
core set debug 10
fax set debug on
sip
2012 Jan 23
1
ConfBridge details
running Asterisk 1.8.9.0-rc2, what are the ways to interface with
ConfBridge ?
I see the CLI command 'confbridge' documented for asterisk 10, but i
dont see how to interface with confbridge on 1.8
What I'm trying to do is keep track of conferences that are used.
I tried something like the below, but not only does Confbridge not
return, but i'd need something that erases the
2012 Sep 20
6
accept email and make phone call?
Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone
call to a specific number and make an announcement?
I imagine the first part is the big question.
joe a.
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
2008 Dec 16
4
RDNIS and asterisk
I have a couple of numbers that are diverted to a number that is
conected to an isdn30 card, running asterisk 1.4.
eg.
123456 => 22334455
654321 => 22334455
What I would like to know is the number of the orginal number dialled
(123456 or 654321). I thought that RDNIS was the answer, but it is
always coming up blank.
When I did a debug on the pri span, I saw the following message
2011 Apr 15
3
sip error logging
I recently noticed that asterisk is not logging unknown sip connections.
I'm not sure if I've broken something or if asterisk itself has been
broken.
the last entry I have is:
/var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c:
Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' -
No matching peer found
my logger.conf
2012 Jul 18
4
asterisk 1.8 on Solaris/sparc
I've got the latest asterisk 1.8 running on a Netra X1 with Solaris 10 u10.
The system itself is happy and phone calls (between two parties) seem fine.
Unfortunately, when a caller listens to a Playback recording, there
seems to be moments of stutter - perhaps 1 second of stutter for every
10 seconds of Playback. The stutter is not consistent at the same point
of the playback file.
To
2010 Aug 29
1
evil disconnect of call with cisco 1760
I have asterisk 1.6.2.11 talking to a cisco 1760 running
ipvoicek9-mz.124-25b.
whenever a call goes through the 1760's FXO or FXS (in or out) there is
a 915 second maximum call time due to asterisk hanging up the call
because of a "critical packet" being missed.
I read doc/sip-retransmit.txt and I don't see anything there that is
helpful to my situation - the asterisk box is
2010 Nov 15
2
SIP calls destroyed after 1:20
After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP
calls are being destroyed after 1 minute and 20 seconds (80 seconds).
Asterisk is sending a BYE message - I have no idea why.
http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug.
can anyone suggest how i can further deal with this?
--
Jeremy Kister
http://jeremy.kister.net./