similar to: Include and Realtime

Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I'd appreciate it. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/076ff188/attachment.htm

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten =>

Hello, I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I

Hi, Does anyone know where this is suddenly coming from? -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected Thanks Dan p.s. sorry about the last post. hit the mouse by mistake and it sent the email. -------------- next part

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

Hi I tried to install the * in my Radhat 7.3 machine. The "gmake" and "gmake install" went fine, but "gmake sample" gave me errors of "configs/*.adsi", "sounds/demo-*" and "sounds/*.mp3" errors. I created a dummy.adsi in the configs folder and commented out the "sounds/demo-*" and "sounds/*.mp3" in the Makefile to

Is there a guy somewhere on how much bandwidth each codec uses, along with the advantages and disadvantages of each one? Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/3bee2776/attachment.htm

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more

Does anyone know how to rewind the music on hold? Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/ab26b18f/attachment.htm

Ah no, I'm asking what code I put inside of parkedcalls? This example works? [ramais] include => parkedcalls [parkedcalls] exten => 700,1,ParkedCall(701) exten => 702,1,ParkedCall(702) exten => 703,1,ParkedCall(703) exten => 704,1,ParkedCall(704) This exten works? 2016-02-03 17:27 GMT-02:00, Doug Lytle <support at drdos.info>: >>>> On Feb 3, 2016, at

Humm, thanks for your reply But whats is the code in parkedcalls context. Please, can you give an example? Thanks very much. 2016-02-03 17:15 GMT-02:00, Richard Mudgett <rmudgett at digium.com>: > On Wed, Feb 3, 2016 at 1:05 PM, Vitor Mazuco <vitor.mazuco at gmail.com> > wrote: > >> Hi! >> >> I tried to use Parking Calls >> >> I use Asterisk

I am having trouble getting park to work with SIP, I have these config files: /etc/asterisk/parking.conf [general] parkext => 8540 parkpos => 8541-8555 context => parkedcalls parkingtime => 45 /etc/asterisk/extensions.conf include => parkedcalls include => default [default] exten => 3874,1,Dial(SIP/3874|20|tT) Do I need something else somewhere? Is anyone using park and

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! I tried to use Parking Calls I use Asterisk 13, but I can't park any calls and it returns me [Feb 3 16:56:11] WARNING[1693]: pbx.c:12543 ast_context_verify_includes: Context 'ramais' tries to include nonexistent context 'parkedcalls' What is the correct code for put in extensions.conf? Can be this example below? [parkedcalls] exten => 700,1,ParkedCall(701) exten

Hi, Im looking at using MixMonitor to record calls and I know that I need to set the filename first. However, with the number of calls coming in, hard coding the filename isnt an option. So I need to do something like this:- MixMonitor(RANDOMNUMBER.wav) But can't find a way to generate a random number. I thought that maybe I could use a unique variable that already exists for the current

Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan --------------

Hello, I'm trying to figure out how to limit the number of concurrent calls a client can make. I have a client that has 6 SIP accounts. One for each SIP phone. I want to limit it so that they can only make 2 outgoing calls at a time so that I can bill them "per channel" rather than "per extension". A separate (but not so important) issue is that I want them to be able to

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I am trying to set the CDR(userfield) to a certain vaule using the application map of features.conf but I am not able to do it. When I receive a call I would like to tag it with a client code (3 digit numeric) so I can referenci it later from the CDR. I have edited features.conf with something like: code => #111,self,SET(CDR(userfield(111)) or