similar to: Variable using AMI

I have had a search for this, but didn't come up with any results, so maybe I am using the wrong terms, sorry if this is an FAQ. For those who want to forward their incoming voice calls to a mobile, it could be a cheaper option to call a mobile from another mobile on the same network. This probably wouldn't be useful for users in USA, Canada or Hong Kong as costs to call a mobile is

Hello, When I attempt to make a call using AMI interface with originate action I successfully specify all of the needed parameters but when I try to control the flow of the call I am unable to identify each call because asterisk uses some kind of unique identification appended to the channel string. E.g. channel: SIP/1000 results in SIP/1000-*0845ea38*. I also found an auto-generated unique

Hello, I recently updated my asterisk-addons-1.6.2 to the last revision and I have this problem that I don't know how to interpret, bug or not. I connected a Nokia N80 phone to use chan_mobile and everything works great until the phone starts getting disconnected after the call finished and sometimes during the call attempt. Is this a bug or a possible known issue for Nokia phones? # rpm

Hello list, Why PC modems were not used as FXO devices? Why chan_modem was deprecated? it seemed a nicer option instead of buying expensive gateways. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090802/abb21767/attachment.htm

Hello list, I want to know the options that I have as an Asterisk user to make a call and control its progress. I have been using the spool directory to originate my calls using this format: "Channel: " "Callerid: " "MaxRetries:" "RetryTime: " "WaitTime: " "Context: " "Extension: " "Priority: " Is there any other

Hi, I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels (25 channels per PRI). Can someone please help me understand using Asterisk Manager Interface (AMI) available in Asterisk to dial out 200 numbers and run a campaign for 200 numbers concurrently and play a mp3 file ? Please suggest/guide. Regards, Kaushal

Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I

Hi, I'm working on building a cluster on AWS atop CentOS 7. For development, I've been working in the eu-west-1 (Ireland) region, where the AWS MarketPlace provides an official CentOS 7 AMI (ami-e4ff5c93). However, the production deployment is taking place in AWS's GovCloud region for regulatory reasons, and there, I couldn't find an official CentOS 7 AMI. Are there plans to

Hi folks, We are upgrading from AMI 1.0 to AMI 1.3 and looking for any documents or AMI 1.3 Specifications. I found AMI 1.4 Specification in wiki.asterisk.org but not for AMI 1.3. Can someone provide me the link for AMI 1.3 specification ? Thanks in advance Shishir -------------- next part -------------- An HTML attachment was scrubbed... URL:

https://docs.asterisk.org/latest/Configuration/Interfaces/Asterisk-Manager-Interface-AMI/Asterisk-Manager-Interface-AMI-Changes/ On Tue, Jul 11, 2023 at 11:54 AM, TTT <[lists at telium.io](mailto:On Tue, Jul 11, 2023 at 11:54 AM, TTT <<a href=)> wrote: > Is there a web page that lists the AMI versions mapped to Asterisk versions? > > I noticed that the AMI version increased

Hi, I'm seeing weird behavior with AMI where no events are output until some input is detected (can be an empty line), at which time all the buffered output is spewed out at once. I am maintaining multiple Asterisk installations, and with one installation I have run into a weird buffering problem with AMI. The version is 1.6.1.11 in this particular case, which I am running at multiple

Hi, I've been developing some CTI software around asterisk for a while, mainly with the help of AMI and fast AGI. It works quite fine, but I have some trouble sometimes with the un-synchronized property of these 2. Let me explain, we have a dialplan like this one : exten = s,n,UserEvent(useful_input_data) (...) a few actions exten = s,n,AGI(agi://127.0.0.1:3333/fetch,queuename) The idea is

Hello List, I was writing something in PHP that connects to AMI and sends a data stream ( example of it: http://slackware-es.com/ami-input.txt ), but the file (voicemail.conf , in this case) does not get fully written. I tried pasting the stream directly through telnet to AMI, and everything *appears* to be OK, but the file is not being completely written. No errors on CLI No errors on AMI

Hi, We're using Asterisk 1.8.0 to run a call centre. There is a Java process which talks to Asterisk through AMI, which is part of the software stack that presents a user interface to the call centre agents. We're seeing a strange issue with AMI. Most of the time, it doesn't cause problems, because the Java code is written to cope with it, but occasionally it does, and, in rare

Hello all, I am running Asterisk 10.10.0 and I can send Message between SIP's no problem. However, I would like to be able to send send Message to a SIP from AMI Or CLI. I check the ListCommands On the AMI and it don't have MessageSend. Therefore, I try the SendText. AMI: Action: SendText" Channel: SIP/600" Body: This is a test. Message: This is a test. Extension: 600";

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

I'm using a perl daemon i wrote to connect to AMI and perform actions. The daemon connects to asterisk via AMI at start up. Is there anyway to check if the AMI connection is still alive, for example every 2 seconds. if the connection is not alive, re-connect to AMI? Also, does AMI timeout after a certain amount of time of not sending commands? -------------- next part -------------- An HTML

>> There are SOME protocol level things accessible using CHANNEL[1] but that's it. >> [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Function_CHANNEL I am trying to use the CHANNEL function listed above from the AMI. Since it is not an AMI “action”, but rather a dialplan “function”, I’m trying to figure out how to call this from the AMI. Using a telnet session

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

Hi, folks. I'm new to both Ubuntu Linux and Wine. I've been running Lotus Ami Pro for years under Windows 98 and have about 70MB of Ami Pro files. Nowadays, though, printers are not made to support Windows 98 anymore. I had heard about WINE, and rather than buy another Microsoft product, I decided to see if I could run Ami Pro under another OS. My new printer works perfectly under