similar to: Using manager originate and Dial() once inside dialplan

Hello, I'd like to use the AMI interface to originate a call to a context in a dialplan, and handoff the dial control to the context. Whenever I execute the below action, the recipient does ring, but when I answer it dials the recipient again. I believe this is because once answered the system is going to execute the Context/Exten/Prio in the Originate action? Action: Originate Channel:

Does anyone know of a way to make an originate request coming over the management interface (e.g. AstTapi click-to-dial) include the relevant Alert-Info SIP headers to enable the originating phone to auto-answer? I've tried setting up a custom context (see below), but the dial plan is only entered AFTER the originating call is answered, so the SIP header is added to the terminating call leg,

Hi, The ami manager call out with an originate through dadhi to a local number (A). If this call is answered, then the ami manager redirect this call to a dial command. This dial command calls through dadhi to another local number (B). Number B answers this call and number A en B are connected. If number B and number A hangs up, there is will be no CDR be written If the dial command is commented

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 would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: "123 <123>" Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1 at

Hey list, Im in this HUGE crisis. Im trying to get a Polycom 601 with two expansion modules to work. I need the XML config files I guess. Does anyone have these I can have? Im trying to get this phone up and running, and haveing MUCHO problems, can someone help me out!! Im sure if I see the configs I can see how it works, just need those XML files!! The ones from the 501 that I have dont seem to

Hi everyone, Ive just installed a bunch of Snom 360s, and now having a NIGHTMARE of problems! Ive got a receponist phone with a extra sidecar on it. And when she gets 2+ calls and puts them on hold, when she goes to transfer them out the calls on hold get merged together. Somehow the calls on hold get merged and not to the extension needed!! Any help on this would be great guys, that would be

Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the

Has anyone used the G option with the Dial app? I'm looking for a way to control the called party leg. Specifically, I'd like to pass a few variables to the called side for some call control. Here's a synopsis of what I'm doing: Make outbound call w/ AMI Originate action. Called party answers ("Customer") Customer identifies himself, and now I use Dial w/ the G

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

Hi All, I have been running a environment with asterisk 1.4.20.1 for some time now with no issue but have recently added some extra functionality (enabled call recording via MixMonitor) and ran into some deadlock issues which seem to be well documented with earlier 1.4.x releases so have decided to take the plunge and upgrade. I decided to start testing with 1.6.2 but have run into a couple

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

I'm trying to keep the DND status of my Snom phones and the astdb in line but I'm stuck on integrating my gui DND button which talks to * using the manager interface (actually it uses Astmanproxy as the gui host is on a different network to asterisk and can't see the Snom's across the network). All's working fine in my Dialplan; when someone dials the code for DND-on or

Hi, I've recently upgraded from Asterisk 1.4.22 to 1.4.29, and I'm now having a problem with Originate and chan_local. I'm using the following Manager API action to originate a call: Action: originate Priority: 1 Context: trunk Callerid: 100 Channel: Local/100 at callback/n Exten: 123456789 Variable: USERFIELD=127.0.0.1|USEREXT=123456789 WaitTime: 30 This is intended to first call

Hi people! Sorry for bringing up some annoying issue.. yes, it's AMD again... But I was searching the last days for a solution for my problem and didn't really find anything. Now I'm hoping that someone of you has maybe an idea for me. :) My setup: --------- I use the Asterik Manager API to generate outgoing calls (by using "Originate" messages). These outgoing calls

Hi List, I cant seem to find the setting that changes this! You put a person on hold, they are on hold like normal, but after a few seconds the phone will then start having dialtone coming from the speakerphone, really weird!! Anyone know how to fix this? I see where it could be nice, but in this case, we just want them on hold is all, no dialtone! Any help would be great! Thanks! Ron

Dear List, My Dial command: exten => _X.,n,Dial(SIP/PBX2/1234,60,G(connect-jack^${EXTEN}^1)) exten => h,1,.... [connect-jack] exten => _X.,1,NoOp(${CHANNEL}) ; Leg A exten => _X.,2,NoOp(${CHANNEL}) ; Leg B The problem is: after answering, [connect-jack] both priorities are executed, and right after executing them call drops. Log: -- Executing [123456 at NPDB2:76]

Hello, If I connect an Asterisk 1.6 to a PBX via Q.SIG and A (on the PBX) calls B (a SIP phone on Asterisk). B answers and puts A on hold. Then B calls C (on the PABX) and does an attended transfer. Is it possible to configure Asterisk so that it does the Q.SIG ?Path Replacement Feature? ? The Q.SIG "Path Replacement Feature" requires the following: After both legs of the

Hello, all. I see there is an "o" option for the Dial() command which reverts to the previous behavior of using the original callerid throughout the call - I suppose more specifically, using the callerid from leg 1 for leg 2 in B2BUA if I understand it correctly. That seems to be highly desirable behavior; I know we are seeing some problems with call history and call forwarding because

Hi. (Asterisk 16.2.1) I'm using AMI Originate to initiate calls, and I'm passing some additional data in to the dialplan context using the Variable: parameter. Works fine. https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerAction_Originate Now I need to do the same thing but from another context in my dialplan, so I was expecting to use the Originate() dialplan command,