similar to: Manager Interface "Action: Originate" changed

Try Capitalizing Your Actions And The Parameters For Them: Action: Login Login: admin Secret: mypass Action: Originate Exten: 200 Context: stations Channel: SIP/agent07 Priority: 1 Callerid: James Bond Calling That should work. We use manager actions extensively in our applications and the managerAPI is sometimes finicky for capitalisation. and a capitalised first letter of the action and

I am trying implement two-stage dialing. Scenario is following: 1. * Dials SIP agent 2. SIP agent answer the phone and provide dial tone 3. * Sends DTMF string 4. "Bridge" channel with calling party I thought that something like: exten => _2XX,2,Dial_but_not_connect_(SIP/BYEXTENSION,10) exten => _2XX,3,Wait,1 exten => _2XX,4,SendDTMF($DTMF_DIGITS) Should do it. Thank

Hi to all. I'm begin a use the AMI and i have the need to get the uniqueid from the call i have generate using the Action Originate. Anyone can help me? When I generate these commands: action: Originate channel: SIP/101 application: Dial data: SIP/100,120,Ttr The only response I get when the call is answered, is this: Response: Success Message: Originate successfully queued Thanks a

Hello, I am migrating a PABX system based in Asterisk 1.4 to Asterisk 13, with success. I have an application that sends an action Originate to AMI for calling, it's working well, but when i see to Asterisk's CLI, i see 2 calls for just one originate: pftestes40copiabh*CLI> core show channels verbose Channel Context Extension Prio State Application

Has anyone else had issues with Originate returning the wrong error code? According to the docs, the following errors are supposed to be returned: 0 = no such extension or number 1 = no answer 4 = answered 8 = congested or not available Now in Asterisk 1.4.23 I get some error code 5's but since they're so few I tend not to worry. But what is concerning is the number of Error 0's I

I'm using the Originate Action on the Asterisk Manager to place calls between two extensions in async mode. Is there any way to cancel the Originate Action before I get the OriginateResponse action? I'm unable to perform a Hangup because I can't know the channel name before I get the response... thanks in advance! -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel.

An application commanding asterisk with AMI is going to launch lots of concurrent calls in very few seconds using the Originate AMI command but it's also going to need to be able to cancel very quickly any call of them even before each OriginateResponse event comes in. All the calls will be done by the same trunk (a trunking enabled channel). But there's a problem for canceling any call:

Gang, I'm wondering if anyone has run into this problem and found a solution. When I use the manager interface to generate a call, I don't get very much information in my CDR records when the dial status is BUSY, FAILED, NOANSWER, etc. I am putting the dialed number into the CDR Userfield in my dialplan, but the field doesn't populate the CDR record unless the Originate action is

Hi Asterisk-users, I am working with Aterisk Manager API's. I can login successfuly with the following. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel:

major undertaking. But since you are using an AGI to control the Queue command instead of using it from the dialplan, you have more control over this problem than you realize. For simplicity of illustration, let's say your AGI simply wants to take a call and send it to the next agent in the queue. Your Agents are Agent007, AgentQ and AgentM. Because you did the Polycom transfer from

I use the action Originate?i want the called first ringing?the called answer,callee ringing.it can achieve? Best regards, justhinker -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101011/27846a78/attachment.htm

Hi people! I need a help. I connect with Asterisk Manager and execute an Originate Action that asks for Asterisk to call for a number 09194**** and to transfer to peer 101. This command enters in my dial plan extension that I made exclusively for tests. The problem is: If I do the call through ORIGINATE Action, the Monitor() just makes the record file, but this is empty, if I do

There is a problem when transferring calls using REFER, asterisk does not notify dialplan. I've been told to use AMI as a workaround to notify my dialplan/routing program but that would require a huge change to our software. I was wondering if there is any intention of fixing this problem. Here is issue as stated in chan_sip.c "this is currently broken as we have no way of telling the

Hi, I'm straggling with setting up a call via manager interface. Basic functionality works fine but I try to use this addons: Application: Playback Data: beep when a call is answered by A side, 'beep' is played correctly but no further action is taken - I got hangup !!! Why it's not connecting to B side of connection after playing the 'beep'. When removing

I've discovered that when dialing out using API's Originate action, a no answer is considered a failed attempt, while a busy is considered a successful attempt. The problem I'm having is that when I dial an invalid number, say a disconnected number that gives a fast busy, my CDRs are identical to those generated by a no answer attempt. Is there a way to distinguish between a no

Hi all, do anybody know, whether is it possible to route packets incoming to ethernet interface as broadcasts? ~~~~~|WirelessDevice/WD|-----eth0-|LinuxRouter/RT|-eth1---(10.18.63.0/24) tcpdump: listening on eth0 0:a:e6:ac:e8:7a ff:ff:ff:ff:ff:ff 98: 192.168.7.11 > 10.18.63.249: icmp: echo request (DF) 0:a:e6:ac:e8:7a ff:ff:ff:ff:ff:ff 98: 192.168.7.11 > 10.18.63.249: icmp: echo request

Dear All, I am originating the call directly to the SIP Provider using the maganger interface + originate (ASYNC) command. Here is the PHP-AGI Script. $call = $asm->send_request('Originate', array('Channel'=>"SIP/416XXXXXXX at ABC/n", 'Context'=>'ORIG',

Hello Everyone, I want to send a few numbers (variables) when doing Asterisk AMI Originate command and then have Festival read them back to customer in the context. How should this be done? Following is my not working example and also reference on AMI Originate Command: Command: Originate Channel: SIP/123 Context: TestFestival Priority: 1 Exten: 555 Variable: $numberONE, $numberTWO, $numberTHREE

<<I am getting "Message: Originate failed" even the phone is ringing on the other end of the line.>> Originate will ring your own extension first and when you pick up, call the other number. If you don't pick up your extension, you will receive the message you see. Bill Seddon ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf

Hi I am trying Manager interface for originate a call. This is what I get --------------- Action: Originate Exten: 555 CallerID: test <6656> Context: local Timeout: 600 Channel: SIP/8782 Priority: 1 Response: Error Message: Originate failed ---------------- What do I do wrong? Thank you Serge _________________________________________________________________ MSN Premium with Virus Guard