similar to: macro on ring group

New installation from AsteriskNow 3.0.0 with asterisk 11 and freepbx, running Digium D40 and D70 phones. Direct-dialed extensions work fine, but extensions in RingGroup won't ring - dialparties.agi apparently removes them from the dialstring pre-emptively: dialparties.agi: EXTENSION_STATE: 4 (UNKNOWN) dialparties.agi: Extension 2010 has ExtensionState: 4 What can I do? Any extension

Hi, I'm having problem with call parking. When I park call, either via transfer to xten or park digit sequence from features.conf, I hear the parking lot number read to me and the user gets transferred. However, MOH stops for the caller the moment user is transferred. The user can be retrieved by dialing the parked extension and voice resumes. If the parked user hangs up, the channel state

Hi list, For months everything worked super here in our setup. This week I implemented some new idea in our webbased calendar system. I thought it would be nice to have an option that tells asterisk you are not available for calls during an appointment. For this to work I could no longer use the ringgroup setup: Dial(SIP/10&SIP/11&SIP/12,40,tr) So I thought, why not use the Local channel

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

I've got 4 SIP phone lines with a call-limit of 2 for each. I've written a handy macro to allow my users to dial a phone number and the macro will figure out the next available line to use by first checking if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a backup, and if it can't use the line for either reason it goes to the next line. The problem is that there

Take a look at freepbx 2.2 beta. We have made both ringgroups and follow-me have a call confirmation option. When used, the ringgroup/follow-me extensions that are outside lines (like your cell phone) must confirm they want the call (press 1 to accept, 2 to decline). All the while the caller hears ringing (or MoH if chosen). If no answer, they are sent on to where ever else you want them to go

Looking for a mentor... Having some issues with Asterisk 1.4.22 install. I am new to both Linux and Asterisk, however have 20+ years programming experience. First off I hate asking questions I could answer myself. I have and am reading The Asterisk manual, 2nd edition. I have successfully installed CentOS 5.2 and used yum to get a C compiler, current speed bump is with ./configure bash:

On Thu, Mar 25, 1999 at 09:16:32AM -0600, Andy Bakun wrote: > Sorry that this is an older message, but I had a few questions. NP. > Nicolas Williams wrote: > > > On Thu, Mar 18, 1999 at 07:36:52AM +1100, Andy Bakun wrote: > > > Every day samba gets closer to having more complete NT acl and group > > > support, and it would be nice to be able to do things on the

Hi, I set up a ring group. I would like for people who select a certain voice menu option to ring a list of extensions (I have just one extension in there at the moment) and if it doesn't answer to go to an extension's voice mail. I am using a version of asterisk from CVS, last updated a couple of weeks ago. This line in extensions_addtional.conf sends the call to ringgroup 3 if

Hi, I'm using heartbeat as a failover for my asterisk server. on the active server 1 i have 10.10.10.1 eth0 10.10.10.3 secondary eth0 asterisk listens to the secondary ip, so that if server 1 fails, server 2 will then get that IP. so if server 1 fails, server 2 will have the IP 10.10.10.2 eth0 10.10.10.3 secondary eth0 problem is i have to bind asterisk to the secondary IP if dont, i

Hi, i'm currently using Originate command on AMI, i can call a certain channel like a SIP user SIP/1000 then once 1000 is answered it dials out to amobile or landline. Would just like to know if i can use AMI to dialout to a mobile or landline first (instead of SIP user) and once answered, dial another mobile or landline again. If not is it possible to call a macro from the AMI? i think

I use the following macro for my extensions. It only works with Zap channels and assumes that any Caller*ID number that is 4 digits is an internal call and all other calls are external calls. Use like this: exten => 1234,1,Macro(std-exten,Zap/4,20) [macro-std-exten] ; ; Caller*ID is 4 digits (internal call) ; exten => s/_XXXX,1,Dial(${ARG1}r2,${ARG2}) exten =>

Clean up of patch for letting kernel run other than ring 0: a. Add some comments about the SEGMENT_IS_*_CODE() macros. b. Add a USER_RPL macro. (Code was comparing a value to a mask in some places and to the magic number 3 in other places.) c. Add macros for table indicator field and use them. d. Change the entry.S tests for LDT stack segment to use the macros. Signed-off-by: Chuck Ebbert

Clean up of patch for letting kernel run other than ring 0: a. Add some comments about the SEGMENT_IS_*_CODE() macros. b. Add a USER_RPL macro. (Code was comparing a value to a mask in some places and to the magic number 3 in other places.) c. Add macros for table indicator field and use them. d. Change the entry.S tests for LDT stack segment to use the macros. Signed-off-by: Chuck Ebbert

sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows SIP/XYZ at 119.68.0.90:5060 SIP/XYZ at 202.16.34.10:5678 so dial command with unique-id i want to use will be Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT) and not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)

Hello Nhadie, I had a very similar situation. My solution, even tough might not look very wise, solved my problem the way I needed. I repeated the GotoIftime command in the next line in my extensions.conf . Like this: GotoIfTime(22:00-23:59|*|30|jul?test,s,1) GotoIfTime(00:00-02:00|*|31|jul?test,s,1) Rgs, Marco Cordeiro -----Mensagem original----- De: asterisk-users-bounces at

Hi all, Is it just me and not reading the docs right, or has anybody else had problems with the AbsoluteTimeout application and the 'T' extension when used inside a macro? [macro-attended] ; ARG1 is the device to dial out on, SIP or Zap, or whatever ; ARG2 is the extension to dial using 'attended' dialing exten => s,1,AbsoluteTimeout(30) exten =>

Asterisk is an excellent PBX system, and makes a very good endpoint in the SIP chain for all sorts of things -- IVR systems, voicemail applications, automated messages, etc. It has an extremely well-written CDR engine, so many people mesh it with billing applications to produce accurate accounting information. It also is fully aware of the media stream, which means it's capable of cutting

Hi, Is it possible to use realtime for Music On Hold? Is it also possible to store the music/audio files on the database, same way a voicemail can be stored on the database? Thank You Regards, Nhadie

X.Org Security Advisory: April 14, 2015 Buffer overflow in MakeBigReq macro in libX11 prior to 1.6 [CVE-2013-7439] ========================================================================== Description: ============ It's been brought to X.Org's attention that this commit: http://cgit.freedesktop.org/xorg/lib/libX11/commit/?id=39547d600a13713e15429f49768e54c3173c828d which was included