similar to: outdial to phone for new VM notification

Using Asterisk 1.6.1.14 and dahdi 2.2.0.2+2.2.0. We're placing outbound calls over an analog line. Some of these calls are going to cell phones that play music rather than providing a standard ring. As a result, the Dial command sometimes returns a DIALSTATUS of CHANUNAVAIL and sometimes it returns BUSY. The problem is that this is happening on calls that are being answered. Has anyone else

I would like to use * as a standalone voicemail system. As such I need it to be able to outdial a certain extension for MWI-ON and another extension for MWI-OFF Is there anyway to get * to automatically dial an extension when a voicemail is left and another extension when the mailbox is cleared? Thanks -------------- next part -------------- An HTML attachment was

Hello, I have set up Comedian Mail on my Asterisk system. I am using Voicemail not Voicemail2 in my extensions.conf file. The system works great except for 1 thing...It is not possible to create custom unavailable or greeting messages for 3/4 voicemail boxes. For some odd reason 3/4 users are unable to modify the default voicemail prompt with their own custom greeting. The greeting gets

Hi list, Is this supposed to be a joke? It doesn't sound very professional. comedian n 1: a professional performer who tells jokes and performs comical acts Moby Thesaurus words for "comedian": banana, buffoon, burlesquer, card, caricaturist, choreographer, clown, comedienne, comic, cutup, dramatist, dramatizer, dramaturge, droll, epigrammatist,

One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp

Has anyone been able to successfully integrate the Avaya AUDIX voicemail system with Asterisk? I have been diligently investigating converting our small (Ontario, Canada) office to Asterisk, and ditching our Avaya PBX. However, our head office (New Jersey, USA) maintains our AUDIX system, and a) have no intentions of leaving it and b) some users rely upon AUDIX's ability to transfer

A user has their unavailable message played and once that message is over the Comedian message is played right after. Is there any way to prevent the Comedian message being played if the user's unavailable/busy message is being played. Thanks, Kurt

Can Comedian Mail handle more than just an away and busy message? I've got a client that would like even more of them..... I can write an app to replace messages externally, but I was wondering of comedian could handle it internally. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 2967 bytes Desc: not

Hello: We offer several options for accessing Comedian mail via Asterisk and IP phones. We'd like to add one more. We would like to be able to save a message on the Asterisk server in such a way that it can be retrieved using the telephone set and Comedian mail as it is today and/or via an IMAP client. If accessed via am IMAP client we would like to use an "externotify script" to

Hello, On a Debian Buster instance, I compiled Asterisk 17.3.0 from source. I enables all 3 File, IMAP and ODBC voicemail modules but I'm still using classical File module (in modules;conf and voicemail.conf): cd asterisk-17.3.0 ... make menuselect.makeopts menuselect/menuselect --enable app_voicemail_imap menuselect.makeopts; done menuselect/menuselect --enable app_voicemail_odbc

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

Hi all, This is my first post to the list. I have searched the net far and wide but can't find an answer to this problem. When I have call forward working or use the voicemail from a SIP phone, the first part of the message is always cut off. So instead of hearing "call forward cancelled" I hear "l forward cancelled". Or in voicemail I hear "edian mail"

I have a user who goes into Comedian Mail for the first time and goes thru the initial setup, changes password, records name, etc. Problem is that every time he calls in, it thinks that it's his first time and keeps reprompting him. His password change is reflected in voicemail.conf. Others do not have this problem. Where does Asterisk maintain the "first time" flag? Any ideas

Two questions: 1. Does anyone have a good way to transfer a call from inside comedian mail to the current extension? The problem is: let's say the phone goes to voicemail after 4 rings, yet I don't hear it until the 3rd ring. I come running into my office but miss it by a split second. Is there a way I can barge in on the person leaving a message for my mailbox while they're

What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the

Hello, I am working on an outdial project and the Asterisk box is connected behind a PBX via SIP. When a call from the Asterisk box is routed out over the PRI attached to the PBX I am not getting proper call progress. The PBX is indicating that the call is answered while it is still ringing at the far end. Does anyone have any suggestions on how I should go about waiting for a variable number

I have successfully built and made asterisk talk SIP extension to SIP extension, read all the docs, and about 1000 emails from the archive. The trunk side of Asterisk, from the docs perspective, is a smidgin TDM-centric, Analogue, T1, zaptel.conf etc..... Asterisk cares not about the externally presented digits as the telco KNOWS which time-slot or analogue line the call came from I live in an

Brad's recent list of enhancements look good, but I haven't looked at the code yet. If the code looks good, I hope it will be committed to the project CVS. Here's a partial list of enhancements that I would like to see in Comedian Mail. I am probably interested in helping to fund the enhancement of Asterisk voicemail. Is anyone else interested? -Address message to multiple

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();