search for: 100&sip

...rs, but I'm not sure how to use this with WaitExten. so I have exten => 4349701010,1,Answer() exten => 4349701010,2,ringing exten => 4349701010,3,wait(8) exten => 4349701010,4,Background(asterisk-recording) exten => 4349701010,5,WaitExten(9,m) exten => 4349701010,6,Dial(SIP/100&SIP/123&SIP/132&SIP/134&SIP/149,20) exten => 4349701010,7,VoiceMail(100 at default,u) exten => 4349701010,8,Playback(vm-goodbye) exten => 4349701010,9,Hangup() Where could I check for the extra # keystroke? Thanks for your help. eddie

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/ast...

...rded, I need that voicemail to played on all phones on that system... E.g.: 1) Administrator --> Dial special number 2) Record emergency message (e.g. Snow day don't come in) 3) Hang up 4) System dials all extensions and plays emergency message. Please re-read before you fire off Dial(SIP/100&SIP/101&SIP/102) There are about 500 extensions so I guess either a System(/path/to/perhaps/perlscript.pl) or something? Scenario, school in a mountainous region with constant horrible weather needs their admins to have a number they can call and record a message. That message is to be di...

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

Dear, I'm currently using a Dial command with multiple destinations and channels eg: Dial(SIP/100&SIP/101) I simply would like to know, in real time during the call (from dial plan or AGI), who has picked up the call. Can I find this information in a variable somewhere ? Thank you for your help Patrick

...all Manger). I have some 7960's that have multiple lines on them. The second line specifically is a "helpdesk" line that is shared among multiple phones. Here is how I am making that line ring on multiple phones, maybe you have other suggestions: exten => 135,1,Dial(SIP/135@100&SIP/135@101,20,rt) So this rings the second line on the phones that have the first line as 100 and 101. This works great. When someone leaves a voicemail, the messagelight will only light on the phone that was booted up last. Is there a way to make the light come on all of the helpdesk p...

I am using the AMI interface to start calls. At one point I have a 10 second delay "Wait(10)" in the dialplan... During this time it "seems" that if I then connect with the manager interface and place a call that nothing happens till the 10 seconds is done... I am requesting Async yes... manager_str Action: Originate[CR ][LF ]Async: Yes[CR ][LF ]Channel: SIP/430[CR ][LF]

...ng List - Non-Commercial Discussion Subject: Re: [asterisk-users] Length of dial string Paddy, Why not use local extensions? You can do something like this. Exten => s,1,Dial(Local/set1 at call_all&Local/set2 at call_all&Local/set3 at call_all) [call_all] Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 Exten => set1,1,Dial(SIP/112&SIP/113&SIP/114&SIP/1015&SIP/116&SIP/117 On Fri, May 1, 2020 at 3:22 AM Paddy Grice <paddy...

Hi Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. Thanks in advance. P

Hi all as per the new release notice for 13.33.0 received today - can anyone advise me the max limit of the string to the Dial Command - see * [ASTERISK-27946 <BLOCKED::https://issues.asterisk.org/jira/browse/ASTERISK-27946> ] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION when it shouldn't I have been fighting with this issue for months trying to find a solution I

Paddy, Why not use local extensions? You can do something like this. Exten => s,1,Dial(Local/set1 at call_all&Local/set2 at call_all &Local/set3 at call_all) [call_all] Exten => set1,1,Dial(SIP/100&SIP/101&SIP/102&SIP/103&SIP/104&SIP/105 Exten => set1,1,Dial(SIP/106&SIP/107&SIP/108&SIP/109&SIP/110&SIP/111 Exten => set1,1,Dial(SIP/112&SIP/113&SIP/114&SIP/1015&SIP/116&SIP/117 On Fri, May 1, 2020 at 3:22 AM Paddy Grice <paddy...

Hi all, at time i am trying to get a better idea of callgroups and pickupgroups (especially within the SIP Channel) A Pickupgroup is relative clear - everyone in the same pickupgroup may pickup a call And a callgroup does what ? - The same ? I thought that a callgroup would act like the ZAP groups - so that you then can dial SIP/g1 - and every SIP Client which is in the callgroup 1 does then

...What I want is for the TRANSFER_CONTEXT for all technologies to be the same as the initial context defined in the configuration of the device initiating the transfer. This is not as simple as it seems (unless I am missing something). For example: A call arrives on IAX/1234, and executes DIAL(SIP/100&SIP/101&SIP/102) in the dialplan. Let us then assume that the contexts are configured in the config files as: IAX/1234: context=external SIP/100: context=default SIP/101: context=superuser SIP/102: context=local So depending on who receives the call, and who REFERs it, the...

Hello, Is it possible to use the asterisk manager interface to originate multiple channels? like Action: Originate Channel: SIP/101&SIP/102 So that both extensions 101 and 102 rings simultaneously. I am using asterisk manager interface over http. Thanks

When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE 603. I am dialing **212 with the following config. Anyone have a suggestion? EXTENSIONS.CONF -snip- [BLF_Group_Pickup] ; Defines how the extension to pick up a ringing phone in your BLF group exten => _**XXX,1,Pickup(${EXTEN:2}) exten => _**XXX,n,Hangup() [BLF] ; Defines a BLF Hint for phones exten =>

...ompleted, returning 0 -- Executing [s at incoming-dial:6] Answer("Zap/4-1", "") in new stack -- Executing [s at incoming-dial:7] PlayTones("Zap/4-1", "ring") in new stack -- Executing [s at incoming-dial:8] Dial("Zap/4-1", "SIP/100&SIP/101&SIP/102&SIP/107&SIP/111,20,tr") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using UDPTL TOS bits 184 == Using UDPTL CoS mark 5 -- Called 100 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 == Using UDPTL...

Hello ! I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18 The things work up to 80%, I can transfer the call by BLF button and I can see the green (free) status and red (busy) status. What I cannot do is to accept the call when someone rings a remote extension. The BLF button starts to blink in red telling me that the call is ringing on remote extenson, but if I press it,

Hello group! I?m new to Asterisk but got one running finally :) Now I?m trying to solve following problem. I have company Automated Attendant and each employee have SIP phone at home, SIP phone in office, cell phone. I want all those 3 phones to be ?one?. So, if someone calls our company number and dials my extension - I?d like 3 phones to ring at the same time. What is this feature and where