similar to: DEVICE STATE "In use"

Hello list. We just upgraded to 1.6.1.11. We are using real time information stored on mysql databases. That is all running fine. Now, since we upgraded, some member don't get calls from queues. In CLI: "queue show" shows something like: 611 (Local/611 at agents) with penalty 20 (realtime) (*In use*) has taken no calls yet We use the extension 611 in different computers, in the

Hello list members, after programing of dialplan I have some messy Custom:hints which I can see in 'devstate list'. I didn't find any possibility how to remove this hints from Asterisk and I want remove them.? Can you help me with that, please? I tried search about that something in documentation or on Google, but I didn't find anything.? asterisk*CLI> devstate list ?

Have an interesting problem, Using asterisk 1.6.0.1 Phone A receives voicemail, dials into VoiceMailMain, Phone B's BLF for A lights up. Phone A deletes the voicemail but still in VoiceMailMain, Phone B's BLF for A goes off. Phone A hang's up, Phone B's BLF for A goes on. From this point forward the Phone B's BLF for A seems to always show the opposite of what it

Hello Have a setup of asterisk with realtime SIP devices. Trying to organise monitoring of my SIP devices. Once device registered, its state becomes NOT_INUSE (result of DEVICE_STATE(SIP/device) function). Simulating of device breakage - powerdown it. Waiting for a while (minute or two), retrieving DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI: sip qualify peer

Hello, A few days ago I've asked about the ability to play a stuttered "ringing" tone when the called party is already on the phone. I've found a partial solution for it. To describe again the problem: When a user is on a call and someone else calls him, the caller does not know that the called party is on the phone (while the called party wants to know that someone else is

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 =>

One of my sites asked for a way to identify if the person they are calling on another extension is already on another call. To that end, I wrote a bit of code in the dialplan for my extensions that checks to see if the extension they are dialing has a device status that is anything other than NOT_INUSE. If the device is NOT_INUSE, then it dials the call normally. If it has a different status,

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! Have a weired problem with Avaya 9650 phones: extensions.conf exten => 0317998975,hint,SIP/0317998975 exten => 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1) exten => 0317998975,2,Hangup() exten => 0317998975-INUSE,1,VoiceMail(0317998975 at inputinterior.se,bs) exten => 0317998975-INUSE,2,Hangup() exten => 0317998975-NOANSWER,1,VoiceMail(0317998975 at

Wait a second... Action: DBGet\r\nFamily: DS\r\nKey: 0733025975\r\n\r\n In the dialplan: exten =3D> 0106024975,1,Set(DB(DS/0733025975)=3DINUSE) exten =3D> 0106024975,n,Hangup() exten =3D> 0106024976,1,Set(DB(DS/0733025975)=3DUNAVAILABLE) exten =3D> 0106024976,n,Hangup() Just a short call to my cell phone, to se if i get anything back, my = cell phone doesn=E2=80=99t even ring. Wait

I have 2 asterisk servers - serverA and serverC - connected via IAX2. On serverA, I have a "telemarketer hold" extension which, if I transfer a caller into it, loops around playing music & "please wait" messages, until they give up & hang up the phone. Also on serverA, I have a custom devstate, which lights a lamp on a phone connected to serverA, which tells me if

-----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

Hi. Im having trouble setting variables in channel dialplan and re-using them in Extension dialplan... Im using the following call file: Channel: Local/210332450 at ZonNew-Outbound CallerID: ZonNew-Outbound:49:210332450: MaxRetries: 5 RetryTime: 10 WaitTime: 60 Account: Outbound210332450 Context: agents Extension: 888210332450 Set: __PARTNER=ZonNew-Outbound Set: NUMBER=210332450 - In

Hi We have the following test .call file and test dialplan: I can't set a custom CDR var to a value on one channel leg, and another value on the connected channel leg? Is there a way I can woraround this issue? ## test call file Channel: Local/queue at TiagoGeada CallerID: teste-geada:0:210332450: MaxRetries: 0 RetryTime: 1 WaitTime: 8640 Account: teste-geada Context: TiagoGeada

Hello all, I am using asterisk 1.4.17 together with misdn, once in a while: -when a call was put on hold -the operator tries to call a internal party for transfering the call -the internal party doesn't answer the phone -the operator wants to get the external line backup again by putting the call "off hold" And then the external line is disconnected. an exact log of events is

In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF updates to SIP peers that have subscribed to a hint looking at a Custom device if that Custom device state is RINGING or RING_INUSE. All other states seem to be working correctly. The hint section of the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show

Hello, Raw stats: Version:1.8.3.2 OS:Centos 5.6 Special setup: postgre database I am having a few queue issues with Asterisk specifically relating to breaking out from queues while on hold. The intent is that while someone is on hold they can press a key (lets say *) to break from the queue and go elsewhere (in this case to leave a message). However In all of my testing I am unable to get

Hi list! We currently have a PRI gateway composed by a box with two Digium quad-span PRI cards (a TE420 and a ). One of the cards is filled with TELCO1, while the other has first two slots filled with TELCO2, and 3rd slot with TELCO3. I am currently having (timer ?) issues on TELCO3 (span 7) D-Chan (202 as determined by dahdi_genconf ) is constantly failing causing on-going calls to terminate.

I'd recommend using DEVICE_STATE On your extension 101, Check the DEVICE_STATE of peer SIP/101, If it's not 'NOT_INUSE' then dial it, Otherwise dial SIP/102 exten => 101,1,ExecIf($["${DEVICE_STATE(SIP/101)}"="NOT_INUSE"]?Dial(SIP/101,40)) same => n,Dial(SIP/102,40,t) same => n,Hangup() On Wed, Feb 25, 2015 at 2:08 PM, ricky gutierrez