similar to: Direct entry to your own voice mailbox

I would like to access VoiceMailMain2 skipping extension and password prompting if calling from a resource that has a mailbox defined. What variables can I use to retrieve the calling channel & calling extension (if it exists)? Here is what I'm trying to accomplish (of course ${CallingResourse.MailBox} is not a real way to retrieve this info)... exten =>

Hi, I've been trying to use the set_callerid function in the AGI. It sets the CallerIDname properly but I can't figure out how to set the CallerIDnumber. Is it at at possible ? Cheers. SL

Can someone help me with this AGI script to send an email. It just isn't working. The file is being called in the dialplan and is saved as em.agi but it isn't sending the email. #!/usr/bin/php4 -q <?php ob_implicit_flush(true); set_time_limit(6); $in = fopen("php://stdin","r"); $stdlog = fopen("/var/log/asterisk/my_agi.log", "w"); //

i've just spent the pass 2 days trying to get AGI to work with PHP; i made a lot of silly mistakes along the way which could have been avoided if only there were some kinda howto or samples. at the risk of looking stupid, i decided to shared my experience in hopes that it might help some newbie get going with PHP. 1. first order of business is to be aware of your php environment; i m NOT

I lost my "connection lost" error.. but now I''ve got a "Mysql::Error: MySQL server has gone away" error.. It comes from: /usr/lib/ruby/gems/1.8/gems/activerecord-1.15.6/lib/active_record/connection_adapters/abstract_adapter.rb:128:in ''log''. I''m not sure if Rails / BackgroundRb uses it .. but i''ve got the mysql (2.7) gem

This morning we got a fault management message from one of our production servers stating that a fault in one of our pools had been detected and fixed. Looking into the error using fmdump gives: fmdump -v -u 90ea244e-1ea9-4bd6-d2be-e4e7a021f006 TIME UUID SUNW-MSG-ID Oct 22 09:29:05.3448 90ea244e-1ea9-4bd6-d2be-e4e7a021f006 FMD-8000-4M Repaired

Hola, estoy teniendo problemas para conseguir que mi función haga lo que quiero. Necesito que coja los valores de la variable que le indico, le quite la letra que precede a dichos valores y los convierta en números enteros. Dejo un ejemplo de los datos que estoy tratando y de varias opciones de función con las que intento que funcione sin resultado. #Ejemplo >

List, I finally decided to break down & start playing with AGI scripts, but for the life of me, I can't figure out what I am doing wrong. Below is a super simple script to run a query in mysql to see how many call records there are for the extension calling in, then print the total in the CLI. This is all I get on the CLI: -- Executing AGI("SIP/216-0baa",

I noticed this issue in stringr::str_replace, but it also affects sub() in base R. If the pattern in a call to one of these needs to be a regular expression, then backslashes in the replacement text are treated specially. For example, gsub("a|b", "\\", "abcdef") gives "def", not "\\\\def" as I wanted. To get the latter, I need to escape

Hello, I recently started playing with voicemail2. I'm having two minor problems that I can't seem to find discussed in the archives. 1) New message 0 in mailbox 7606. New voice mail message count seems to start with 0 for the first new message instead of 1. Any tricks to fix this? 2) When listening to messages with VoicemailMain2, the time stamp is in GMT and not corrected for the

Hi guys, I'm running Asterisk-0.5.0 and accidentally stumbled on this problem while in the VoicemailMain2 application: If you login to it, or even if you call it w/ 's<extension>' to skip the login and press an '8' near the beginning (and possibly at any point, I'm not sure), the channel seems to lockup, even if the handset is hungup, the channel remains frozen

On 4/13/16 11:57 AM, A J Stiles wrote: > You could try > *CLI> dialplan show Between my older backup and dialplan show, I guess that's my best shot. Thanks :D

Voicemail.conf in MySQL is not functioning where I get the following error from Asterisk messages log file: CLI debug output is as follows: Executing VoiceMailMain2("SIP/2205-3df0", "") in new stack -- Playing 'vm-login' -- Playing 'vm-password' -- Incorrect password '1234' for user '0' (context = <any>) -- Playing

I saw a request for this feature raised several years ago. Has this been implemented yet? If not, how are users monitoring traffic on port 8000, and even better can you split it up by mountpoint? I want to know how much traffic our icecast clients are using each month. Thank you, Shannon

I have a question, I am unclear on how to park a call. I know that you are supposed to be able to press "#" and then transfer the call to extension 700. However, * doesn't seem to be graping the dtmf. I am using dtmfmode=inband. Asterisk is in the media path as well. Thanks in advance Justin

Asterisk Users: I've been working hard on app_voicemail2 which is an enhanced scalability version of app_voicemail. Specifically, its features are: * Highly improved internal architecture (maybe someone else can actually code on it) * Foot print for getting mailboxes from DB (for Vonage) * Segmentable mailboxes, allowing you to truly multihost voicemail for multiple companies

Dear Eliza, Try this: Lines1<-readLines(textConnection("1911.01.01?????? 7.87 1911.01.02?????? 9.26 1911.01.03?????? 8.06 1911.01.04?????? 8.13 1911.01.05????? 12.90 1911.02.06?????? 5.45 1911.02.07?????? 3.26 1911.03.08?????? 5.70 1911.03.09?????? 9.24 1911.04.10?????? 7.60 1911.05.11????? 14.82 1911.05.12????? 14.10 1911.06.13?????? 7.87 1911.06.14?????? 9.26

Has anyone browsed through the source code and made a list of menu option for VoiceMailMain2? Or know of some user documentation hiding in Internet land some place? If not there well be soon. Ho hum.

Dear Group, I have been successful in setting up the Agents, queues and getting agents to log in. Is there a way that I could configure the system so that the agent is called back. i.e. the agent logs into the system, a call is destined for them and their phone rings. If some one has this setup I would be very interested in hearing from them. Warm Regards and Thanks --------------- Shad

Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 <-> PSTN gw)? - Asterisk ip: 192.168.1.10 - h323<->PSTN gw: 192.168.1.20 I've tried: exten => _9XXXXXXXX,1,Dial(OH323/192.1.1.20) or exten => _9XXXXXXXX,1,Dial(OH323/BYEXTENSION@192.1.1.20) but it does not work at all. If my h323 client