similar to: Voicemail and Time Conditions

Folks, We have several people using SIP softphones in the office. When they leave for the day, they power down their workstations, causing their registration with Asterix to quickly timeout. Here's the entry for one such extension in extensions.conf: exten => 8102,1,Dial(SIP/someone,20) exten => 8102,2,Voicemail(u8102) exten => 8102,3,Hangup exten => 8102,102,Voicemail(b8102)

Friends in the Asterisk community, One thing I avoided working with for a long time is the Asterisk voicemail code. One module in Asterisk I've constantly been naming as one of the worst parts is voicemail. One part of Asterisk that I've been kind of avoiding during my trainings is voicemail. And there's where I've spent a lot of time recently... Life is strange. Instead

Asterisk isn't playing my voicemail greetings even though they are defined. Below are the relevant configs(from show dialplan) as well as the level 3 verbose messages asterisk is giving. Also a listing of the directory. Asterisk just plays the "The person at extension..." message, not the greetings I have recorded. Thanks -- asterisk*CLI> show dialplan macro-stdexten [

1.6.2.0-rc1. I am having trouble with voice mail intermittently not working correctly on CHANUNAVAIL. (it may happen for other statuses too, haven't checked). Basically here's what happens: -- Executing [1651xxxxxx at mydids:1] Macro("SIP/ipkall-trunk-14838bc8", "phone,1651xxxxxx") in new stack -- Executing [s at macro-phone:1]

In asterisk 1.6 the voicemail prefix u b don't work, I have: exten => 1,3,Voicemail(u11) and it keeps telling me: No entry in Voicemail config file for "u11" exten => 1,3,Voicemail(11) works, Isn't prefix "u" suppose to play: "The person at extension ... 11 ... is unavailable," ? -- Joseph

On 2017-04-20 05:14 AM, J Montoya or A J Stiles wrote: > This is just screaming "configuration mismatch" -- or, possibly, "latent bug > whereby things parsed in separate places should be treated the same, but are > actually getting treated differently". I really don't want to be the "my system isn't working so there must be a bug in Asterisk" guy

Hello everybody, I'm using Asterisk ( 1.6.1.9 ) Voicemail. For example, if i call extension *11 which is not registered, from extension *12, i have no greetings at all, i only have the "please leave a message after the beep". I tried to record the busy, unavailable and temporary greetings for extension *11 using VoiveMailMain and the file are well created on the file system.

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

Hi, I have a load of IAX extensions, which I'm trying to set up a standard macro to dial them, which gives unavailable or busy voicemail if there is no answer or the phone is in use respectively. The macro I have at the moment is: ; std-exten macro, ${ARG1} = Device to call, ${ARG2} = voicemail box [macro-std-exten] ; Call the user for 20 seconds exten => s,1,Dial(${ARG1},20,tr) exten

Hello All, I have my own AGI script running and I am trying to push the call to voice mail when Busy, Unavailable and Not Answered. Everything is working fine but the only problem is voice mail greetings for Busy and Unavailable is not played. By default only "Temp Greetings" voice mail greetings is played. I am passing the correct parameters for Busy => 'b', Unavailable

Hi, I'm trying to set up remote voicemail pickup. I've created the following dialplan, but when I press *, I am not sent to voicemailmain. The unavailable message just continues to play as normal. exten => 2345551111,1,Set(MAILBOXID=1) exten => 2345551111,n,Set(MAILBOXCONTEXT=company3) exten => 2345551111,n,Voicemail(${MAILBOXID}@${MAILBOXCONTEXT},u) exten =>

I have a pretty standard setup with Asterisk acting as a PABX for a bunch of SIP handsets (in this case, SwissVoice IP10S). My users are complaining that when they forward their phones to their cellphones on unavailable (i.e. forward when no-answer), their cellphone only rings once or twice, and then Asterisk sends the call through to Voicemail. I'm using the standard extension Macro

Hi, 1. How do you then, synced then unread message presence with custom device status ? From an external program ? When a user leaves VoiceMailMan application ? Using externnotify ? 2. What is MWI:101 at default expression for (see [2] ? Cheers [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+button+to+Monitor+a+Voicemail+Box 2017-11-21 17:58 GMT+01:00 John Kiniston <johnkiniston at

---------------------------------------------------------------------- Message: 1 Date: Tue, 6 Mar 2007 20:02:07 +0100 From: Olle E Johansson <oej@edvina.net> Subject: [asterisk-users] Building a new voicemail system... Testers needed! To: Asterisk Non-Commercial Discussion Users Mailing List - <asterisk-users@lists.digium.com> Message-ID:

Hello , I want to enable recording for a few extensions. In sip.conf it is defined as record_out=Always record_in=Always under the section of extension.but it doesn't work. Extensions are defined in the extension_additional.conf file like exten => 10,1,Macro(exten-vm,10,10) exten => 10,hint,SIP/10 exten => ${VM_PREFIX}10,1,Macro(vm,10,DIRECTDIAL) I can't be sure

A call comes in from the PSTN, Asterisk answers it, it goes to the directory, and then to the extension the caller designates and the user at that extension answers. The user at the extension then wants to transfer the call to another extension; on the Cisco 7940 they push the ?more? soft key, then the ?Transfer? soft key, then enter the extension number they want to transfer to, and hit the

Hi guys, I've posted my dialplan design in flowchart format and would welcome your comments for improvements. I'd like to know my logic looks ok before I go much farther or start rewriting my extensions.conf. http://www.triadinternetsystems.com/asterisk/ For the Standard Extension macro, I was trying to do the following: 1. execute telezapper if user has turned it on 2. execute

In this dialplan, the SIP user agent is a Sipura two line adapter with line 1 as SIP ID "1000" and line 2 as SIP ID "2000". Basically I have this set up so that 1000 and 2000 are "lines in hunting" on incoming extension "555". I want an incoming call to try to ring ext. 1000, if 1000 is busy, then ring 2000, if 2000 is also busy than ring Voicemail. Here

Does this model give you functioning mwi? > -----Original Message----- > From: Ray Jackson [mailto:ray@jacksonz.net] > Sent: Friday, January 05, 2007 3:17 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [asterisk-users] Voicemail personalised greetings using > DB/IMAPbackend? > > > Hi all, > > I am attempting to build a horizontally

Hi all, I am attempting to build a horizontally scalable Asterisk deployment and am getting very close to achieving that goal. With Asterisk 1.4 I now have an IMAP backend for Voicemail messages which is great as users can check the same messages either through the voice portal or using Webmail. However, I'm not sure the best way of dealing with personalised greetings such as a