search for: ringgroups

...=> 582 XTENS => 255X DTRK => smg01 CONF => 2559 ADALNUMER => 2550 ADALDIAL => SIP/2551&SIP/2552&SIP/2553&SIP/2554 BAKVAKT => 7712555 [general] static=yes writeprotect=yes clearglobalvars=no userscontext=default [dialplan-1] include => conferences include => ringgroups include => internal include => landlines include => gsm include => special include => international include => parkedcalls [dialplan-2] include => conferences include => ringgroups include => internal include => landlines include => parkedcalls [dialplan-3] includ...

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, I have asterisk installed. When a connection comes from the outside one of our phones rings for about 45 seconds. Is it possible to another phone fetch the call while it's ringing on the first phone? I don't want to use ringgroups because the second phone would be ringing also. Thanks Nuno Fernandes -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/piper...

...mailbox=120 at default host=dynamic dtmfmode=inband disallow= dial=SIP/120 context=from-internal canreinvite=no callgroup= callerid=device <120> allow= accountcode= call-limit=50 Maybe someone has an idea how to setup the scenario without using ringgroups... Thanks a lot, Stefan

Hey, I have a few dids with callcentric. They seem to work fine most of the time but at some points I get "handle_request_invite: Failed to authenticate user <sip:PSTNnumber" This happens intermittently. The way I understand it the insecure=port,invite should tell asterisk not to authenticate users coming from that host. But its not working for some reason. This is my sip.conf

...ing *12. - "Custom" device technology support - this means devices that are not configured directly in AMP's admin can still be used (ie: SCCP) - Asterisk Recording Interface (ARI). ARI is a php interface to Voicemail and Monitor recordings. (written by littlejohnconsulting.com) - RingGroups now use strategies: Ring All (default), Hunt, Memory Hunt - DID Routes re-written as Inbound Routing. This allows for DID specific fax emails and call answering options. - Queues can now play a "welcome" message to callers upon joining. - HINT priorities for FIXED devices - Interface...

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...

...(works normal) Executing [6000 at DLPN_WorldcallDial:1] Macro("SIP/8888-0000001e", "stdexten,6000,SIP/6000") in new stack This is dlpn_worldcalldial [DLPN_WorldcallDial] include = default include = CallingRule_worldcall include = parkedcalls include = conferences include = ringgroups include = voicemenus include = queues include = voicemailgroups include = directory include = pagegroups include = page_an_extension Thanks a lot if you can tell me what to check

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

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

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

...functionality ported to new module system -- Added Modules: Conferences, Time Conditions, Asterisk CLI, Online Support -- Inbound routes can set ALERT_INFO variable for SIP devices -- Outbound Routes can now use an Authenticate Password File -- Queue Static Agents can have penalties applied -- Ringgroups can play an announcement to caller before dialing -- Ability to force Emergency Caller ID for devices using an Emergency Outbound Route. -- Much improved form validation for all modules - Requires Asterisk 1.2.x - Using native music on hold support - no more mpg123!! - FOP .24 - ARI 00.08.03 - no...

Hi guys, I've an asterisk 1.2.5 runing as production system. Now it becomes very important to my customer an exact analysis of CDRs for their QoS to their customers. I've been analysing the CDRs, and i notice many entries like this: Calldate |Channel |Source | Clid | Dst | Disposition | Duration

Hi all, Lets say I have my incoming calls transfered to my mobile phone. When a call comes in, Asterisk will answer the call and ask the caller to hold the line while the call is being transfered. I know how to do this, but i dont want the caller to hear me answer the mobile phone. They can hear some music on hold. When I answer Asterisk will read the callerID to me and I can then decide if this

Hi All, For my setup, i am using a macro to dial a certain extension not just a simple Dial(SIP/exten). I would like to setup a ringgroup, for now what i only found is by simply dialing like this Dial(SIP/exten1&SIP/exten&SIP/exten3) but i cant use since i'm using a macro, is there a way i can call multiple macros at the same time like the way Dial calls it? Hope i'm not

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

We need incoming calls to simultaneously ring SIP phones, and a cell phone which is called via a SIP or IAX trunk. When the cell phone answers we'd like a brief prompt played (e.g. "press # to accept call") and if # is pressed connect the incoming call to the cell phone. ZAP trunks have some of this functionality with the call confirmation option, but we must use SIP or IAX trunks.

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 All, > > History ~ > I recently took a position with a call center. At the time they had > about 50 agents in a call queue. The queue was setup to ringall. The > agents use Eyebeam softphones. Everything is local lan, no routers, > everything connected via Cisco 3600 10/100 switches. > > Now we are up to about 150 agents, and I have kept everything pretty

Has anyone attempted to use FreePBX for a business in production mode? Initial take is there are lots of things scripted but a lot of limitations in terms of supporting basic business functions. Inability (or lack of flexibility) is handling multiple incoming pstn lines, dialplan limitations, poor/no documentation, etc, to mention a few. Maybe its just me, but it appears its no where near