similar to: Voicemail and Cisco phones: Dialplan example

; 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

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

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'll do that, but I'm sure that it's the Polycom :) Caller ID shows up fine on my 7960... W/ the name on top and the number below that. -- Executing NoOp("SIP/614-3ede", "Caller*ID is Matthew Marlowe <6092521155>") in new stack When the phone rings, only 'Matthew Marlowe' would display. When I answer, both the Name & Number will show.

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

I've managed to use GhoustScript (gs) to take a postscript file and convert it to tiffg3, but I CANNOT seem to make it merge multiple files. Here is the output from tiffinfo on the file that SG generates: fteTYGeh2v.tif: TIFF Directory at offset 0x8 Subfile Type: multi-page document (2 = 0x2) Image Width: 1728 Image Length: 1056 Resolution: 204, 96 pixels/inch Bits/Sample: 1

On Fri, 2004-05-07 at 16:30, anthm@lists.digium.com wrote: > Update of /usr/cvsroot/asterisk/apps > In directory mongoose.digium.com:/tmp/cvs-serv17955/apps > added D() parameter to app_dial to allow post connect dtmf stream to be sent using above call > +" 'D([digits])' -- Send DTMF digit string *after* called party has answered\n" > +"

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 =>

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

These are the three cheap SIP phones that I've used. Grandstream BT10x $65/street Number only LCD Zultys ZIP 2 $100/retail No LCD Uniden UIP 200 $120/retail PoE, built-in switch -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 "In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss."

I'm getting the following when sending to a specific fax machine. Any ideas? File name is '/var/spool/asterisk/email2fax/7F2SOeYJiU.tif' Changed from phase 0 to 2 Slow carrier up Slow carrier down Slow carrier up <<< NSF: 20 00 00 11 80 00 8a 49 10 53 54 49 52 4c 49 4e 47 20 43 4f 56 49 4e 47 54 00 67 00 80 80 80 0c 01 02 NSF without final frame tag The remote is made by

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)?

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

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

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

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

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