similar to: Call forwarding and voicemail

Hi All I am trying to simplify a dialplan for a few thousand users. Would what I have below work? If someone dials exten 710001 would it go through answer and then to the macro to try dialing the SIP phone thats registered on 710001 and then onto voicemail if no answer or not signed on? exten => 71XXXX,1,Answer() exten => 71XXXX,2,Macro(71macro,${EXTEN}) exten => 71XXXX,3,Hangup()

Yo, Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate call divert-feature. This one validates if the extension a call-forward is to be set to is actually valid for the current context and additionally saves this context into the DB and always uses it to originate the divert from, as you can't expect the

I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael

Has anyone had any problems with the voicemail app in the last weeks cvs? I was using a checkout from 10-31 and everything worked fine, but the last few days I get a hangup right after it calls the voicemail app. No sound, just an immediate disconnect. Nothing has changed in my extensions.conf, etc. Here is what I have in extensions.conf: [macro-oneline] exten => s,1,Dial(${ARG1},20,tr)

Does anyone know what application I should place this function in? For example with the DB function I currently do something like this to add an entry to the asterisk database: exten => s,n,Set(DB(AGENT/${MACRO_EXTEN:1})=${CALLERID(num)}) To delete the entries I do something like this: exten => s,n,DBDel(AGENT/${MACRO_EXTEN:1}) DBDel is marked as deprecated in favor of the DB_DELETE

I have added the following to a macro that is used for all extensions so a user can access voicemailmain by pressing * during the voicemail prompt ; check voicemail exten => a,1,voicemailmain(${macro_exten}) exten => a,2,hangup The behavior is a little weird, the * key is not recognized during the portion of the greeting where the extension number is being played back, after it is

I have looked on the wiki and in the archives for as much info as I can find about call forwarding. I have it working based on the sample from Tips&Tricks except for handling voice mail. When I detect a forwarded extension, I call Dial(Local/${fwNum}@internal/n,20) to dial the number as thought it were dialed directly from our extensions. If I have the extension forwarded to an internal

Hi all, I am trying to understand how I can get a simple IVR scenario to work properly (having already removed most of my hair...). The basic requirement is as follows: * Caller arrives at our main number * Caller is greeted and then told they can enter an extension number, if known, or wait and their call will be connected to an available rep. * The IVR then dials a group of extensions (if

I use the following macro for my extensions. It only works with Zap channels and assumes that any Caller*ID number that is 4 digits is an internal call and all other calls are external calls. Use like this: exten => 1234,1,Macro(std-exten,Zap/4,20) [macro-std-exten] ; ; Caller*ID is 4 digits (internal call) ; exten => s/_XXXX,1,Dial(${ARG1}r2,${ARG2}) exten =>

I've tried to simplified the dial plan and use "n" instead of numbers but I've noticed it is not executing my voicemail if I substitute number with "n" In the example below when the call is not answered, it does not go to voicemail; call just hangup. exten => 1,1,Playback(transfer) exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw) exten =>

Hi, I have following one-line macro extension: ------------------------ [macro-oneline] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Device(s) to ring ; #exten => s,1,AGI(misterhouse.agi,"CallerID") exten => s,1,NoOp exten => s,2,DBget(temp=CFIM/${MACRO_EXTEN}) ; Get CFIM key, if not existing, goto 103 exten => s,3,Dial(Local/${temp}@default/n) ;

Dear All, I was trying to enable call forwarding, following the steps of the link on voip.org regarding this issue it doesn't work and the phone I am trying to implement on is still ringing. below is my conf in extensions.conf and the CLI output during the process. My configuration is : exten => _*5X.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:2}) exten => _*5X.,2,Hangup exten =>

hello, I have a test setup with 2 asterisk servers, each having a one snom 100 via sip using it. I`m experimenting on how trunking between them would work. I have them setup for RSA authentication which I plan to use in the future. So I`ve setup the keys and servers seem authenticate to each other. One is named phila and other hurricane. Here is what I see on phila: -- Registered

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091111/2b828eff/attachment.htm -------------- next part -------------- Hello. I am trying to execute an fax reception script and i am getting the following: [Nov 11 08:40:52] WARNING[12800]: app_system.c:88 system_exec_helper: Unable to execute '/var/lib/asterisk/scripts/mailfax ""

I don?t know if the last message was with content. So, I sent again. I have installed a Digium card TE210P and unicall for use MFC/R2. I think that it?s all right but I can?t make and receive calls. I?m using asterisk 2.1 with the patch made by Jos? P. Leit?o and the follow libs: libsupertone-0.0.2 libunicall-0.0.3 libmfcr2-0.0.3 zaptel 2.1 My number is 34318300. The Telco send me only 8300.

Hello all, Can this be done ? Would setting the variable FAXRESOLUTION to a appropriate value affect this change ? > http://www.asteriskguru.com/tutorials/rxfax.html Variables connected with the application LOCALSTATIONID - used by to application to identify itself to the remote end LOCALHEADERINFO - used to generate a header line on each page REMOTESTATIONID - set by the application, the

Hi All, I was made aware today that some of my calls coming in are not going to voicemail... Below are some logs, and the macro that should run on the incoming_pstn context for that extension. I can see that theres a non-zero exit before it gets to voicemail, but I've no idea why. In this case theres 2 SIP clients to sim-call. On other occasions it works fine. In the CDR logs, I can see

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

I have Spandsp working fine. Asterisk sees a fax on the zap port and redirects the call to the fax-in area. This works if I have a simple dialing rules that goes answers first and waits 10 secs then goes to the next item. If it hears a fax it goes to the right place. Here is a sample that works. [incoming] exten => 2019,1,Goto(test,s,1) [test] exten => s,1,answer exten => s,2,wait(5)