similar to: AMI Transfer?

Hey Gang, Trying to figure out how I can do the following (have each part working individually but drawing a blank on combining) 1) click on-screen which sends an AMI originate (works fine) 2) the originated call is to an internal extension that looks up the number to be dialed (works) 3) then via Perl, adding in a SIPAddHeader for answer-after=0.. (works separate from the above) What I

Way back in the mists of time, I built my asterisk installation with SNMP support. It's a bit tedious to get the sub-agent for snmpd set up but once you have it you can poll the OID for the asterisk sub-agent and it will tell you how many calls are up at that moment in time. That said, I actually prefer ARA/ARI to flat file configuration of endpoints and dialplans.  Changes are more or less

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

Hey gang, Trying something a little funky with a click to dial app. (Perl AGI) It's working fine but I'd like to change what displays in HUD's screen. Right now it's showing "Unknown". I've tried setting various CDR fields but can't seem to change the display. I've tried setting CID and other AGI variables but nothing seems to affect that display. Any

On Monday 15 June 2020 at 00:41:14, Bruce Ferrell wrote: > Way back in the mists of time, I built my asterisk installation with SNMP > support. Heh... I never even knew that was possible :) > That said, I actually prefer ARA/ARI to flat file configuration of endpoints > and dialplans. Changes are more or less instantaneous and easily shared > between instances. Agreed - ARA is

I've been trying to get an AGI "transfer" to work for several weeks now. It isn't error-ing out, but it isn't working either. I can't use "dial" in this case due to what I'm trying to accomplish. Does an AGI Transfer actually work? -= Info about application 'Transfer' =- [Synopsis] Transfer caller to remote extension [Description]

I've been Googling all morning and searching voip-info.org but not quite finding what I'm looking for. I've read that you can modify the billing/account information on a CDR via AGI but I can't find an example or a "how to". I'd like to then assign specific accounts in the CDRs. Possible? Thanks, PB -------------- next part -------------- An HTML attachment was

Hi, I noticed that asterisk manager interface will only accept the originate commands in sequential order. For example, if I want to ring two extensions through the AMI, and while first extension is ringing, AMI won't execute and ring second extension until first extension has answered the call. Anybody has any ideas as I had the same results even tested with telnet commands to AMI interface.

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

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

hello, First of all i am using Asterisk 1.6.2.9-2 The following problem seem like a bug to me but im not sure. Any help or comment will be great.. We are trying to implement our own billing software with AGI - Php Scripts. When a hangup received, i am calling a script to calculate the bill and we are trying to write the results to cdr database. We added two fileds to cdr table as rate

I'm parsing ` sudo asterisk -rx "core show calls" | grep active | head -c 1 ` as an external call from within the Asterisk dialplan then passing it to agi, but this seems really hacky and ugly. However, I cannot find any ARI/AGI/AMI function (or global variable I can get with agi) which shows me this. Any ideas?!? In case it helps and you're wondering why... I need to ensure

Hi all. I'm trying to write an AMI handler to take care of cases where a user uses the feature code to start recording a call. When a user starts a recording, my AMI listener receives this event: $VAR1 = { 'Server' => 'example.com', 'Event' => 'MonitorStart', 'Uniqueid' => 'server-1374906101.132305',

Hello, I'm pleased to announce the first release of Obelus, a MIT-licensed Python library to interact with Asterisk using the AMI and AGI protocols. Compared to existing libraries, Obelus is framework- and programming-style-agnostic, and compatible with Python 3 as well as Python 2. It also has an integrated test suite. This is version 0.1, and as such some APIs are a bit draftish and not

Hello, There seems to be a problem with asterisk cdrs when calls are generated via AMI Originate using Local channels. Asterisk writes CDR as soon as A party off-hooks. Resulting in very inaccurate billsec and duration values. Expected CDR in case of local channel origination should be 2 records, one can be when A party answers, and the other should be written when B party hangs up the phone.

I am using asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically "core show channels concise" "sometimes" I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF ]' I'm not

Hi We want optimize my extensions file conf on asterisk 11.4.0 : We have a big quantity of extensions, all are same "design": ; Destination: Gambia Type: Fixe exten => _00220X.,1,Set(CDR(CodeCom)=BUS-GMB) exten => _00220X.,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) exten => _00220X.,3,Set(CALLERID(all)=${NUMID}) exten =>

Hi, i got Ubuntu 6.06 installed and theres a problem with asterisk. I've sucessfully installed it with the command: #apt-get install asterisk Then after installing FreePBX i get this error when restarting asterisk: root@hernandezz-laptop:/home/hernandezz# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) After looking at the logs i

Since I passed from version 1.0 to the 1.2.3. I have Pb with the callerid. If somebody call with presentation of the number all is well. If somebody make call in masked number, i couldn't send a callerid to the phone. It is in a call center and i use the callerid to present the name of the number called to the operator. Before that went. To identify the sda, I use the assignment of the

Hi, Can anyone explain the difference between Asterisk Gateway Interface and Asterisk Management Interface ? Is it correct to consider AGI scope to focus on call handling and AMI scope to anything which can be done with Asterisk froma loading new modules to originating calls ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: