similar to: Manipulate extension state in 1.8.x

You can use a custom device state to do it. [dnd] ;DND Toggle exten => *363,1,Answer() same => n,Set(CURRENT_PRESENCE=${DEVICE_STATE(Custom:DND${CHANNEL(peername)})}) same => n,GotoIf($[${CURRENT_PRESENCE}=NOT_INUSE]?*78,1:*79,1) ;DND On exten => *78,1,NoOP(Turning DND On) same => n,Set(DEVICE_STATE(Custom:DND${CHANNEL(peername)})=BUSY) same =>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I must be missing something here but I can't figure out why I can't get DEVICE_STATE() to give me anything other than "NOT_INUSE". I have two extensions: 6666 and 6668. I used 6668 to make a call to yet another phone, so I know that it's busy. I then use 6666 to call 6668 and in the dialplan have a noop to see what

Is there a way skip / ignore the member whose status is busy in the Queue. I have two channel member in queue and i have set the peer limit 2 for these members. I want to skip those member who are currently on the call (answered to calls) and now their status is busy, if Queue see the busy status caller will not enter in the Queue and go to the next priority. [test-queue] strategy = rrmemory

Hi list, I need your help ,I have an incoming call x the ivr and the operator takes the call. ext "101" , If a second call reenters and the operator is talking, I want to send to the extension 102 I use the Variable DIALSTATUS , but not working check IVR [IVRINMA] exten => s,1,Wait(1) exten => s,n,Set(CHANNEL(language)=es) same=> n,Set(TIMEOUT(digit)=4) same=>

On Tue, Nov 25, 2014 at 8:14 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Mon, Nov 24, 2014 at 2:12 PM, James Lamanna <jlamanna at gmail.com> wrote: > > Also, how big does the cache in frame.c grow to? > > I've recompiled with MALLOC_DEBUG on that server: > > > > asterisk -rx "memory show summary" > > > > .... > >

Hello, Some of my users have phones lacking a DND button. I need to provide an extension they can dial that will put them in DND, i.e. tell the server not to send them any calls until they get off the DND. I've researched it for almost 3 days now and tried a range of configurations. I'm hoping somebody here has an answer. Currently, I have this in extensions.conf [app-dnd-on]

Hi and happy new year! My question: - two extensions: 1111 and 2222 - an active call on 1111 - incoming calls to 1111 should be forwarded to 1111 (call advice!) and 2222 I know how can I forward an incoming call to more than an extension, but I have no idea how can I get the information, that 1111 has already an active call... I think, I need something like: exten =>

Hello - I've read Asterisk should be able to activate a do not disturb feature to turn off the ringers on extensions. I checked the wiki and can't find documentation for how to do it. Here's my attempt, added to extensions.conf: [dnd-on] exten => _#78,1,Answer exten => _#78,n,Wait(1) exten => _#78,n,Macro(user-callerid,) exten =>

Hi! Trying to figure out what I am doing wrong... 1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256) 1 Cell phone 00733025975 attached through H323. extensions.conf exten => 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1) exten => 975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 975-INUSE,2,Hangup() exten =>

Hi all! I am trying to figure out how DEVICE_STATE is working, no luck so far. sip.conf [0317998975] type=friend regexten=0317998975 secret=???? username=0317998975 callerid="Magnus Benngard" mailbox=0317998975 at inputinterior.se host=dynamic canreinvite=yes dtmfmode=rfc2833 nat=yes disallow=all allow=alaw extensions.conf exten => 0317998975,hint,SIP/0317998975 exten =>

Also, how big does the cache in frame.c grow to? I've recompiled with MALLOC_DEBUG on that server: asterisk -rx "memory show summary" .... 1780466242 bytes (1780181594 cache) in 2352909 allocations in file frame.c ... Seems like a ridiculous cache. On Mon, Nov 24, 2014 at 9:02 AM, James Lamanna <jlamanna at gmail.com> wrote: > cat /proc/cpuinfo lists 4 cores. >

i have posted this question couple of times and never really got any hits i wasn't able to provide any debug info Connected to Asterisk 1.6.0.21 currently running on phoneserver (pid = 3309) Verbosity is at least 4 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using SIP VRTP TOS bits 136 == Using SIP VRTP CoS mark 6 == Extension Changed 117[ext-local] new

Hi Brian, many thanks to you for your answers in the past! The always gave me the little bit of mising information... My Asterisk box is running fine now so I want to try the "next step"... And now to all of you .... What I want to implement is to use 1 button of my snom-360 phone for following purpose: If I leave my desk I press this button. A light should show up as an

Hi all, How can I implement a full-featured Call-Waiting behavior on the Asterisk level (e.g. I don't want to relay on end-equipment capabilities)? I found it very strange that such a basic feature is not built-in in Asterisk (and I've googled a lot in search for this). Here is what I need: SomeuserX is calling MyUserA. They are on conversation (assumption: voice is via the Asterisk)

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1

>>> But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... And that we don't. It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been answered. I was successful with using CONNECTEDLINE when issuing

Hi All, I am using a Grandstream BT100 and I have been trying to get the PBX features to work for DND, call foward, etc. These functions do work when I use my POTS phones hooked up to my Zap cards. But I cannot get the PBX functions (ie *78, *79) to work using my SIP phones. Is there a feature that has to be enabled to do this? I know these functions are available within the GS phone but all of

That is the general idea. But how do i make it work? is there somewhere ready? On Wed, Jan 10, 2018 at 6:39 PM, John Kiniston <johnkiniston at gmail.com> wrote: > Define your *72 and *73 extensions in your internal context, Have them set > a value in the ASTDB that you then check when dialing your handsets. > > The same can be done for call forwarding, store a number in the

No idea on how to write it in my system. On Thu, Jan 11, 2018 at 12:17 AM, John Kiniston <johnkiniston at gmail.com> wrote: > There's some example code in the Dial-Users context of the basic-pbx > samples that might be of use in implementing it. > > They are checking a DEVICE_STATE to see if a phone is BUSY, You could > change it to be a database call or implement custom

Hello, is there some way to make Asterisk light up a certain light on an IP-phone ? Like MWI, the message waiting indicator can light up if there is voicemail. Could this light, or even other lights (like BLF-buttons) be used to give a visual notification to the user ? For example : if a certain value is set in the Mysql-DB and Asterisk reads out this value, can Asterisk react upon it inside