similar to: RDNIS with tel: vs. sip: header

Displaying 20 results from an estimated 1000 matches similar to: "RDNIS with tel: vs. sip: header"

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
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
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 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 =>
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)?
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
2010 Oct 26
2
Trim the RDNIS
What I am needing to do is to trim the 1 from beginning of the RDNIS and I have tried using the CUT function but cannot seem to make it work for me. What we have is a phone number like this, 18881232342 and want to make it like this 8881232342. I appreciate any help that you guys can give. Thanks! -- *Chris Ramirez* TELE-ONE COMMUNICATIONS, INC. cramirez at tele-onecom.com 903-531-0777
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
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
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
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
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
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 ${CALLERID(rdnis). But as you can see I'm getting the information. My question is, Does
2004 Jun 04
1
Voicemail and Cisco phones: Dialplan example
Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>
2006 Dec 15
1
What's up with DATETIME and TIMESTAMP in Asterisk 1.4beta3 ?
Hello, In Asterisk 1.4 beta 3, the UPGRADE.txt file says: Variables: * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM}, ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE}, and ${LANGUAGE} have all been deprecated in favor of their related dialplan functions. You are encouraged to move towards the associated dialplan function, as these
2006 Mar 06
1
cdr records on transfer
Hello! i'm trying to set up transfer without using the respective asterisk-function but with the built-in phone functions. my goal is to have the first callleg billed to the caller and the second callleg to the callee, who is responsible for the forward(and i can't bill a unknown caller anyways) so far it's working without problems, but my cdr's are messed. with the help of the
2003 Nov 18
1
Asterisk with External Voicemail
If anyone could help me with this, I'd appreciate it! I've got an Asterisk deployment where I'd like to use an existing external Octel voicemail system. I've been trying to define an extension that if the call isn't answered in a few rings, to dial our external voicemail number. That voicemail system works by seeing the CALLED number and routing the call to the
2003 Nov 26
3
AGI - CallerID ??
I have a client who needs an application for there field techs to call in when they arrive on site and when they leave. The logic behind it seems pretty simple. I am going to write something in AGI to capture some DTMF tones and update this data into MySQL to run some reports from. But here's my initial problem. I have started to create a simple AGI script to capture the CallerID, but I
2003 Sep 12
1
asterisk and defunct perl procs
Trying to figure out why I'm having all of my test (and demo) perl script in a defunct status. Each run creates a problem: ps output root 26253 1356 0 16:39 pts/1 00:00:00 asterisk -vvvc root 26270 26253 0 16:40 pts/1 00:00:00 [pj.pl <defunct>] root 26271 26253 0 16:40 pts/1 00:00:00 [pj.pl <defunct>] root 26273 26253 0 16:40 pts/1 00:00:00 [pj.pl