similar to: MailboxExists application

I replied to a post of mine a few days ago asking of anyone uses mailboxexists(). I haven't received any replies. Perhaps few use it or perhaps the reply was overlooked. I thought I'd post the question one last time before giving up on it for now... Thanks! -Michael

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is

Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping-service-examples-12.txtI can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked

Hi, Im on Manjaro linux fully up to date running dovecot 2.3.1-2 pigeonhole 0.5.1-2 All is running well except I am having problems with fileinto - it refuses to use variables, and mailboxexists is also failing for me. Im just dealing with plus addressing - should be simple but the behaviour Im experiencing isnt right. require ["variables", "envelope",

Dear All, Thanks ofr the great job so far. I have crawled the doc & web, and did not find oh to check for the presence of a folder ina? a mailbox? I know about "fileinto :create" of course, but the purpose is slightly different: I want to automate the move into a folder based on some rules but *only if* an associated folder is present (which name is based on the rule),

Thanks for that. Turns out it was an issue with exim stripping the local part suffix from the envelope before passing over lmtp to dovecot. Fixed in the exim router config! Although if you turn on trace with spamtest it does result in a crash during mail receipt: lmtp(test at test.net)<7988><cDK3ADztDls0HwAANIsFaQ>: Panic: Unsupported 0x30 specifier starting at #38 in 'extracted

Hello Dovecot users, Much time passed since I last released a new Pigeonhole, so quite a few unreleased fixes were pending until now. Development of new features is shifted to the upcoming Pigeonhole v0.4 for Dovecot v2.2. I intend to release v0.4.0 shortly after Dovecot v2.2.0. Changelog v0.3.4: * Changed error handling to be less of a nuisance for administrators. Strictly

Hello Dovecot users, Much time passed since I last released a new Pigeonhole, so quite a few unreleased fixes were pending until now. Development of new features is shifted to the upcoming Pigeonhole v0.4 for Dovecot v2.2. I intend to release v0.4.0 shortly after Dovecot v2.2.0. Changelog v0.3.4: * Changed error handling to be less of a nuisance for administrators. Strictly

I would like to be able to send calls to voicemail() but then catch the case where this is no VM configured on the extension and then go back to another menu. However, VoiceMail() will return a -1 and then hangup. I would like to trap it somehow. Is that possible? -Michael

Hi, After seeing in "pbx/ael/ael-test/ael-test5/extensions.ael" some interesting use case of RealTime to store extension data (forwardto, dnd, ...) i started to play with it. To my surprise the two applications RealTime() / RealtimeUpdate() have been deprecated in favor of the REALTIME() function. While it's not a big deal for the write part since it's only a matter a shifting

Has anyone had any luck with PostgreSQL and Realtime? The realtime instructions on voip-info seem pretty straight forward... just not woking for me. I've included all of my config files below, and my console output. Entire console bootup output: [root@abox asterisk]# /usr/sbin/asterisk -vvvvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing

Ok, first off, Asterisk is the coolest piece of software I have EVER had the pleasure of using in my 8 years of running linux !! and I know I haven't even scratched the surface feature wise. Before I get too excited, I wanted to get all you experts to look at the how I implemented my after hours test. The goal is to prevent the phone from ringing afer certain hours, just go to VM.

Hello List, I'm using a dialstring like the one below. I want to have three different things happening depending on exit cause. Dial(SIP/${phonenumber},20,gL(20000[:5000][:5000])) These 3 things could happen: 1, Caller hangs up 2, Callee hangs up 3, The 20 seconds is up and call is terminated from Asterisk. Is there a way to separate these 3? Thanks, Best regards, Tobias --------------

I have been bugging by a telemarketer who does not take any cue at all. So I look up the Asterisk Handbook and send his call with the respect caller id to my voicemail. Has any one implemented any of this feature with database for more caller ids to be included?? David Kwok

Hi folks, i'm trying to configure my extensions.conf as small as posible and for that reason i'm using macros. The problem is that maybe I have a misunderstood the concept for the directive "mailbox" in sip.conf. Under my knowledge configuring the mailbox directive to the mailbox I want would be enought to leave an retreive messages in that voicemail box. Of course it seems to

Hello ! In order to chase after a problem I implemented the following dialplan to have an answertime of exactly one minute: exten => xxxxxxxxxxx,1,NoOp(Test wait) exten => xxxxxxxxxxx,n,Answer exten => xxxxxxxxxxx,n,NoOp(Current timestamp: ${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) exten => xxxxxxxxxxx,n,Wait(60) exten => xxxxxxxxxxx,n,NoOp(Current timestamp:

Hi, I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this: [outbound] Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier)) Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME: ${DIALEDTIME}

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm