similar to: A way to run extenrnotify when IMAP events take place...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is doing some IVR work prior to forwarding calls to the PBX and it also acts as the voice mail server for the PBX, with Asterisk configured for IMAP storage. When a call comes in and the caller leaves a voice mail, the VoiceMail application calls the program configured in

I'm trying to use Asterisk for its voicemail capabilities while interfacing with a legacy Toshiba PBX. Is there a way to have Asterisk send a DTMF code to an extension to turn on the message waiting indicator light? When a user leaves a voicemail, I want Asterisk to pick up one of the lines attached to it, and then dial #63<ext>, which is what sets the message waiting indicator light

I have "externnotify = /var/lib/asterisk/agi-bin/vm_notify.pl" so that when someone is left a voicemail it will call the person's mobile phone and prompt them with the new message. The perl script simply originates a call to a persons mobile phone and connects it to their voicemail using VoiceMailMain. Problem is when user hangs up from checking their messages, it runs the

I'm trying to get the BLF to work correctly on my Polycom phones. I have the buddy watch working correctly, but can't get the BLF to change based on the state... Example: When an extension is ringing, I get the same 'red light' that I get when the extension is actually in use... I was wondering if anyone had any experience with getting the Polycom phone to

Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks

Hi Jeff - > I have IMAP voicemail working with Exchange 2003 using a single username and > password for multiple mailboxes. Sorry to hijack this thread (at least I changed the Subject), but this really caught my eye. I was under the impression that Exchange's IMAP doesn't have the master user feature and therefore can't do single username authentication for multiple mailboxes.

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

I just completed building a feature rich asterisk voicemail system using perl, php, and mysql. My only concern is that the system i built will not be able to handle the call volume needed. Let me start by explaining my setup. Incoming call -> route.agi (perl -> mysql lookup) -> AGI -> voicemailbox (using mysql odbc) or terminate with wrong number message if a message is left in a

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040331/c2abf19f/attachment.htm -------------- next part -------------- Hello, Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone tell how to get started implementing it for Asterisk. Regards, Tony

When a user has voicemail, I would like * to call the user at a pre-determined number (internal or external) and play a message that the user has voicemail, and then give the user the option to login to voicemail and pick up the message. I know about the externnotify feature, but I don't see a way to use it to accomplish what I want. I've checked the archives, etc., but I don't see

Hi, In exim4, I can see lines such as : mainlog.9:2008-03-12 08:53:28 1JZLmC-0000E7-0A <= root at foo.com U=root P=local S=43802 id=Asterisk-0-123413860-4174-2662 at ipbx-bs-60200 In my voicemail.conf, I see : ; If you need to have an external program, i.e. /usr/bin/myapp called when a ;externnotify=/usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp called when a

Hi Please help! I have one X101P and TDM400P in my asterisk Box When i make a call from * to PSTN, everything goes Ok, When the PSTN hangups or * hangups, the busy tone is detected and * disconnects the channel without problems. The problem occurs when the call comes from PSTN. When * hangups, the other end (at pstn) does not hangup, it only presents silence. Please tell me how to solve this

Hi, I'm a brand newbie to asterisk trying to set it up for the first time and I can't get a softphone to connect, the connection times out. I had a trixbox pro install working, but I need more control and would like to learn to do it with asterisk. In /var/log/asterisk/messages I see: WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI

A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 905 bytes Desc: not available Url : https://stat.ethz.ch/pipermail/r-help/attachments/20010420/61ada248/attachment.bin

G'Day, I'm a little confused why the c function has the followng effect on classes - is this a feature ? My workround [ class(cc) <- c("POSIXt", "POSIXct") ] seems to do the job. Many thanks Bernie McConnell "R version 1.3.1, 2001-08-31" on NT > aa <- as.POSIXct("2001-09-23") > bb <- as.POSIXct("2001-09-24") > cc

G'Day, I want to access in a function called from tapply the current factor. In my example below, all I want to do is to write the current factor on each histogram. Needless to say my example does not work. I would be grateful for pointers in the right direction. Many thanks Bernie McConnell Sea Mammal Reserach Unit cc <- 1:10 ff <- rep(c("a","b"),5) pp<-

Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below. The first warning is about music on hold because it

Greetings, The Asterisk.org development team has released Asterisk 1.6.0-beta5. As of this beta of 1.6.0, 1.6.0 is now feature frozen. In addition to a number of bug fixes, the following new features have been added since beta4: * The SMDI interface in Asterisk has been reworked to fix a number of issues as well as add some new features. SMDI message information is now accessed in the

Greetings, The Asterisk.org development team has released Asterisk 1.6.0-beta5. As of this beta of 1.6.0, 1.6.0 is now feature frozen. In addition to a number of bug fixes, the following new features have been added since beta4: * The SMDI interface in Asterisk has been reworked to fix a number of issues as well as add some new features. SMDI message information is now accessed in the

Greetings, Scenario: Centos box with eth1 (10.0.0.0/24) and eth0 (192.168.0.0/24) segment on eth0 has access to full bandwidth of uplink Both are on 100mbps switches Requirements: bandwith on segment on eth1 needs to be throttled to different speeds - say 32, 64, 128kbps and the such. Required for application performance testing purposes. As usual my girlfriend google is very verbose.