similar to: Accessing Voicemail by dialing own number

Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own ring back tone as soon as it detect a ring back tone coming from the carrier side... Is there a way

I have a large dial plan here with over 3000 lines, and several dozen macros. As it grew, it became apparent that there was some problems. 1. When you pass arguments to a macro in the form of $ARG1, $ARG2 etc, if that macro calls another macro, and passes arguments like this as well, you lose the original values. 2. When the macro's 'return' some value, it has to set a channel

I am following the example at http://www.voip-info.org/wiki/view/Asterisk+tips+findme but I find that no matter what, the call is connected. Can anyone confirm that config is working for them? Any suggestions appreciated. I need to transfer calls to a list of cell phones, ring all of them, allow them to screen the call, connect the call to the first number that accepts the call, and allow

I am trying to setup an extension of *7XXX that will allow me to dial *7 and then any extension and use the Pickup application to pickup a ringing phone. Ideally it will also check if the phone is ringing somehow and then either dial the extension or pick it up if it is ringing. But I can't get that far. If I use *7268 specially it works fine, but as soon as I introduce any wild

does asterisk support imaps for voicemail storage? Ide really like to use gmail as my imap server... -- Moshe Brevda, CTO ipconnect, ltd. 26 Strauss St., Jerusalem, Israel W. 1.800.800.456 (+9722.569.5295) M. +97254.666.1367 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080421/00ed120c/attachment.htm

When phone A registers, I want phone B to ring, when picked up, it should call phone A and connect the phones. Translated: When GF in Mexico powers up laptop where soft iax-phone registers automatically, I want to talk to her asap :-) How to? Leif

Hi All; Which is better (to have more stable or release versions) of zaptel, libpri and asterisk: to use cvs or svn? In case of using cvs, why I need to type: export CVSROOT=:pserver:anoncvs:anoncvs at cvs.digium.com:/usr/cvsroot In other words: what is the use of pserver, anoncvs, ... with cvs checkout? Note: How can I know all the variables needed for cvs checkout so I might need to do

Running on branch/1.4 I have been watching some the queries from Asterisk and I think I have a place where some efficiency can come, but I am at a lost as to what is calling it... It appears that every 10-15 seconds, every mailbox INBOX and Old gets queried for the number of voice mail files. I have exposed SIP to verify that it wasn't the phones requesting. It is not much of a problem in

I'm sure this has been discussed many times, but I have a question about hoteling. My understanding would be this: A phone sitting on a desk. A user hits 9000 and it asks what extension you'd like to become. You type "1001" and then it asks for your password. You type 1234, and it says you're "logged in". You now are accepting calls at your phone and you're

Hi, this is to inform everybody that the translation of my new book (unstable version) is online at http://www.the-asterisk-book.com The book is a GNU FDL project. So everybody who wants to participate is welcome to do so. Also, everybody who needs material for his own work, feel free to take it as long as the new material will become GNU FDL too. I am glad that Stephen Bosch (who you

What Asterisk command I can use to send a SIP INFO command? Thanks for pointers. Yuan Liu

Can anyone help with BLF for Snom 300s ? (Asterisk 1.4.10.1) I've setup hints for a couple of Snom 300's but Asterisk doesn't send Extension Changed messages to subscribed phones unless the second 'line' button is used (I've tried Snom's version 6 and 7 and two difference 300s). On the Asterisk Console I don't see any message when picking up a Snom 300 and dialing

I have four Polycom 330 phones connected to an asterisk system. There are other VoIP phones connected too. All of the extensions are four digits beginning with 11. From any of the phones, except the Polycom, picking up the handset to call extension 1103 for example works fine. With the Polycom 330, as I press the second 1 of 1103 it stops taking input and gives me an error. I tried

Hi, is anyone on the list using the Siemens Openstage phones together with asterisk? If yes, is it possible to use the programmable keys of these phones together with Asterisk? Thanks for any hints, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93

Hi List; Can someone advise me why in the below context, it does not run the Background step? Once I dial 1000, then it hangup and give congestion signal? If I comment the ResponseTimeOut, then it run the Background but it does not wait till caller enter the digits, once the sound file finish, then it hangup (congestion signal), also in all the situation, it does not go for the t extension, why?

Is the "Whats New at Digium the Asterisk Company" message I got from digium at en25.com really from Digium? If so I suggest to send it from digium.com and not to use those shady Eloqua redirect URLs. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk?

Could someone with inside knowledge comment on that? If the source code says "deprecated" but the CLI help does not mention that - whom do I trust? -------- Original message -------- Subject: Re: [asterisk-users] Ex-Girlfriend syntax and RealTime Extensions From: Philipp Kempgen <philipp.kempgen@amooma.de> Thomas Kenyon wrote: > Philipp Kempgen wrote: >> You might use

Has anyone set it up so that an inside call and an outside call get different unavailable messages? j

Hi everyone, We have a particular user on our Asterisk 1.4.x system who always listens to his voicemail messages via email. - Is there some way to send the voicemail ONLY to email and not retain them on the phone? - Alternatively, can the voicemail system only keep, say, just the last 10 messages (as backup in case of email delivery failure or a message getting deleted in email accidentally

HI, I dont understand the syntax of the dial application when used like this: Dial(Local/${EXTEN}@longdistance) i want to know what is this "Local" doing instead of Tech like SIP, IAX, H323? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: