similar to: Newbie Voicemail: Just use one [context] invoicemail.conf?!

I'm trying to set up my extensions.conf file using some of the existing macros like stdexten, etc. while at the same time having two logically separate virtual PBX's (with no "default" context) and two trunks coming into separate contexts, i.e. one for residence and one for my at-home business. I noticed, however, that macro-stdexten depends on the "default" context:

Hello, We recently upgraded our Asterisk box from 1.4 to 1.6.1. In both versions of voicemail you can press 3 for advanced options, 5 to leave a message and enter an extension to leave a voicemail. This feature worked fine under 1.4. Now under 1.6.1 all the prompts are the same but when you enter the extension it reads back the extension (or says the recorded name if present) then goes straight

Hello, am attempting again to resolve an issue with multi-tenancy and the forwarding to VMs between mailboxes. If in a multi-tenancy environment one uses custom contexts ie. [a1-ext1](a1) mailbox=101 at a1 and the associated voicemail.conf entry: [a1] 101 => 1234,My User 1,addr1 at email.com,,tz=eastern|imapuser=addr1 at email.com|imapfolder=Inbox 102 => 1234,My User 2,addr2 at

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have

Hi, For a few weeks now, our asterisk server has been experiencing something very odd. From time to time, voicemail.conf would go blank. We finally tracked it down to happening when someone attempts to change their password. It seems the file is touched, but not written to, and we're left with a blank voicemail file. Permissions seem to be fine: -rw-rw-r-- 1 asterisk asterisk 12707

Hi, I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan [macro-stdexten] exten => s,1,Dial(${ARG2}) exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If

Experiencing a problem when users attempt to forward a voicemail from within VoiceMailMain(Option 8) I see the console message: Couldn't not find mailbox XXX in context default As why are running in a multi-tenant environment voicemail.conf has been separated into individual contexts. The users retrieve their email by dialing an extension which calls VoiceMailMail(XXX at VMContext) so how

hi all, I can not get voicemail working in realtime with asterisk-1.2.6. extconfig.conf is correct voicemail => odbc,asterisk,voicemail_users i am getting the fallowing error Executing Answer("SIP/xx.xx.xx.xxx-0a02e1c0", "") in new stack -- Executing Set("SIP/xx.xx.xxx-0a02e1c0", "foo=102") in new stack -- Executing

Hi Guys, i spent some time to figure this out (since i love how dialplan is written) but i decided to ask for your help guys. i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in pbx2 extensions.conf: i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr) in pbx1, i

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 [

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension

with the below defined in logger.conf on 11.6 cert 6 I am not getting any log message other than notice and warning in any files when doing module reload logger - queue log is the only one that says it restarts *CLI> module reload logger == Parsing '/etc/asterisk/logger.conf': Found Asterisk Queue Logger restarted built fresh box with make samples - added 2 stations, dialing from

Paste your extensions.conf section that is relevant. -Matthew ----- Original Message ----- From: "Greg - Cirelle Enterprises" <gcirino@cirelle.com> To: <asterisk-dev@lists.digium.com> Sent: Monday, December 27, 2004 4:32 PM Subject: [Asterisk-Dev] realtime voicemail > Let me clarify my last message. > > If I put in the wrong password I get polled > again for

Hi, I just setup my * with digium. I started testing voicemail first between atas, and i am not sure why it is not prompting me any when the call is not answered or if busy. i only get continuous ringback and the following message: asterisk*CLI> -- Executing Dial("SIP/6882332-1697", "SIP/5104112978|20|tr") in new stack -- Called 5104112978 --

Is it possible to create extensions in the voicemail.conf remotely by using the manager interface. I cannot seem to find any documents or examples describing that capability. Jonathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081015/82b359c2/attachment.htm

Assume you have the messages button on your Cisco phone set to dial 3009. Here's an sample dialplan entry that will make the "DND" and "ToVM" and "Messages" button work as expected. This should work for both -stable and -head. exten => 3009,1,GoToIf($[X${RDNIS} != X]3009,4) exten => 3009,2,VoicemailMain() exten => 3009,3,Hangup exten =>

I am running CentOS 5 with Asterisk 1.4.14. I am trying to setup storage of voicemail messages into MySQL. It is my understanding that I can only do this via ODBC. I installed per http://www.voip-info.org/wiki/view/CentOS+5+and+Asterisk+1.4.x+installation unixODBC unixODBC-devel libtool-ltdl libtool-ltdl-devel and mysql-connector-odbc. I reconfigured and built Asterisk, using menuconfig to

We would like to setup in house SIP phones with numbered extensions for demonstration purposes. What is the syntax to associate a extension with SIP phone? Does the Dial application have a SIP specific entry for example: Dial,SIP/SIPphone/s|15 When I call from one extension to another I get "User is on the phone". We also have Cisco7960s to test. Currently Have X-Lite setup.

hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>

Hello all i have an asterisk setup integrated with mysql via odbc driver myproblem is: when i reading my voicemails, in advanced options the following functions not working with realtime asterisk but working with flat files. 1. Reply to the message(option no:1) 2.Leave a message(option no:5) i have following settings in my general section _ searchcontexts=yes _sendvoicemail=yes [test1] 1001