similar to: Problem with AMI Originate

Hello all, What are the possible values returned by AMI Originate when it's called with Async set to 0? Is there any way to find out whether the dialed channel was busy, invalid, etc. without requiring Async to be 1? Thanks in advance, Leo

Hello there, I'm new to Asterisk and I'm trying to figure out a way to make the Asterisk Manager Interface (AMI) accessible to multiple users at the same time. Would anyone recommend an AMI proxy that could be accessed from PHP code? Thanks in advance, Leo

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

We have several customers we need to place outbound calls for (in a single system). May have to place calls for thousands of different caller ids. Customer signs a contract guaranteeing the caller id they provide is owned by them. I have everything setup for AMI Originate and can place the calls. However, I'm encountering a problem with the caller id. The system I'm dialing through

Hi Guys, I am trying to make a web form where a person is allowed to put in $phoneNumber, $dialNumber, and $spoofNumber to make a call with spoof caller ID. There are a few problems that I am facing with Asterisk AMI Originate command. The reason why I want to use the darn AMI Originate is because I am sending calls to mobile phones and I want to have some accountability and to know if a call was

Lets say I have to make 40 phone calls quickly. If I use the AMI interface to originate a call, close the connection, open another connection etc... This works. but is slow... If I open the AMI interface and originate a call - DONT close the interface , get the response, originate another call, some of the calls are missed. even though I get OK response. (All calls are Async mode). If I open

Hi, Using FOP, I've met a situation which makes me ask this simple question : Are both A and B commands bellow equivalent ? A. CLI: originate SIP/9122 application dial Local/9123 at local B. AMI/FOP: 192.168.64.5 -> Action: Originate 192.168.64.5 -> Channel: SIP/9122 192.168.64.5 -> Async: True 192.168.64.5 -> Callerid: 9122 Guest2 <9122> 192.168.64.5 -> Exten: 9123

Working with asterisk 1.4; using the AMI Originate command, it is possible to do something like: Variable: CDR(accountcode)123456 Or must the variable names be "var[n]" where n is a number? I'd like to set the accountcode for a Local channel that originates a call. Thanks. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE

When calling Originate Action, it rings my phone. If I do not answer, I receive a Channel Event: Hangup, followed by receiving an OriginateResponse event with a Failure Response, Reason 3. My phone continues to ring. If I answer the phone at this point, it attempts to answer, but does not succeed. Looking at the asterisk debug, it appears to destroy the SIP dialog for the call. It also

Hi guys I seem to be having a problem, I don't know if it's a bug or whether I'm just doing it incorrectly. I want to set about 3 channel variables when I originate a call via AMI. All the documentation I have found says to do it like this: Variable: variable1=value|variable2=value|variable3=value However when I do this it runs them all together and I end up with:

While doing a security audit on a system I maintain, I stumbled upon an unvalidated use of a variable to compose an Originate request to the local Asterisk instance via AMI. Taking as an example an earlier exploit for FreePBX, I realized that this, combined with Application: System as an injected value, could allow arbitrary code execution. I am in the process of fixing all instances of this bug

I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133 at 1002

Hello all, I wonder if somebody could provide me with some advice on how to track what looks like a bug to me: I've got a PHP AGI script that is called whenever I dial into the system and also whenever I issue a specific Originate() request via AMI. The script works fine when I dial in. However, when I run it via Originate(), it sometimes does not play anything, sometimes plays part

Hi, I'm getting an issue while executing AMI Originate. I'm getting "extension does not exists" on Originate's Response, and on the other hand Asterisk CLI say "fwrite() returned error: Broken pipe" Please suggest me what is wrong. Muhammad Faheem ### my originate code block ...

Hello, I am using a perl script to pull call info from a DB and place calls via telnet and AMI, all on local machine of course. My problem is that I need to capture any response from the carier, such as this taht appears in the CLI: [Mar 1 12:55:50] == Using SIP RTP CoS mark 5 [Mar 1 12:55:50] -- Got SIP response 503 "No Circuit Available" back from xxx.xxx.xxx.xxx:5060 [Mar

Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: > Action: originate > Channel: SIP/zoiper > Exten: yziquel > Priority: 1 > Timeout: 30 > Context: internal > > Response: Error > Message: Originate failed > > Event:

Hi, I'm getting errors while originating a call through AMI. [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe Asterisk version 11.0.1

I installed version 1.6.0.3-rc1 and my AMI application stopped working. I reinstalled 1.6.0.1 and it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel:

Hi All, I am in the process of migrating from 1.4.20 to 1.6.2.x and have stumbled across a number of "differences" between the 2 versions that are forcing me to use local channels in my dialplan (mainly centered around the different behavior of CDR fields in the 2 versions) . Previously, I would place a call via an AMI Originate action similar to: action:.Originate..

An additional follow-up question, if I need to set the P-Asserted-Identity on the create (originate), is there a way to do this with ARI? From: asterisk-users <asterisk-users-bounces at lists.digium.com> On Behalf Of Dan Cropp Sent: Friday, August 7, 2020 11:51 AM To: 'asterisk-users at lists.digium.com' <asterisk-users at lists.digium.com> Subject: [asterisk-users] With ARI,