similar to: Asterisk DNID/RDNIS with Dial iax2

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)?

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

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

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

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

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

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 =>

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

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

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

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

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

Hi I've just started to play around with AGI scripts and have run into problems. When I run Asterisk in console mode everything works just fine. If I run Asterisk in 'regular' mode (not console) it crashes if I hang up on the script. I have used Python scripts to test this and also the "agi-test.agi" script. (the Asterisk code was compiled from CVS code just a few days

Hi, I have a problem with R, after an update: this piece of code: cat("creating resolver data frame\n"); dfG<-cbind(dfG,2^(RGN$G)) dfR<-cbind(dfR,2^(RGN$R)) suddenly, creates values to inf. Is the syntax changed for the ^ symbol in the latest R downloadable version? Kind regards, wilfred -- Dr. Ir. Wilfred Van IJcken Labmanager Genomics Erasmus Center for Biomics,

Hello! I have been doing some tests with call transfers and I have been looking at the CDRs that Asterisk generates. Scenario 1: A calls B B answers and does a blind transfer to C (using # key) C answers and talks with A Scenario 2: A calls B B answers and does an attended transfer do C (using the phone's transfer key) C answers, B hangs up, and C talks with A For scenario 1, the CDR shows

Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. The clearest demonstration of the problem is that if I dial extension 125 configured as... exten => 125,1,Ringing exten => 125,2,Wait(3) exten => 125,3,Answer exten => 125,4,Wait(2) exten =>

<!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

Hi all, In order to use the "open url" function of zoiper (it opens an url based on the asterisk $callerid(dnid)), I need rewriting of the dnid. In my dialplan I have: exten => 1000,3,Set(CALLERID(dnid)=newdnid) exten => 1000,4,Noop(${CALLERID(dnid)}) exten => 1000,5,Queue(test-queue) but the callerid(dnid) shows the extension called (the member of the test-queue) and not

Hi i have an asterisk pbx with E1 port connected to another PBX. Im trying to send the DNID/DNIS to the PBX here's my dialplan exten => 8881111111,1,Dial(ZAP/g2) exten => 8881111111,n,Hangup() The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or ZAP/g1 the PBX get the number 1. What should i add to send the extension number as DNID/DNIS? Thanks! --------------

Hi, I'm using videocalling on asterisk 1.4.10. When I setup the videocall with exten = n,1,h324m_gw(s at video2webanswer), I loose the variable DNID (${CALLERID(dnid)}) Before the videocall is set up, this variable is filled and after this videocall this variable is empty. Also all local variables are empty. If al look at the A-number (${CALLERID(num)} this variable is not empty