similar to: Dispatching calls question

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

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

I'm writing some code that needs to access AMI in PHP. (I'll probably be doing AGI later as well.) I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and appears to be a bit behind current Asterisk -- No event handler for event 'fullybooted'. What PHP framework/library are you using -- and why? -- Thanks in advance,

I was playing with the AMI today. Trying to get a setup that only responds to connections on the SAME box. like 192.168.1.10 So .10 is my server and I only want responses originating from .10 to answer the AMI. I set bindaddr to 192.168.1.10 I left it as 0.0.0.0 I set permit to be 127.0.0.1/255.255.255.0 I tried 192.168.1.10/255.255.255.0 I tried 192.168.1.10/255.255.255.255 No matter what I

On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of

this is the point, and the strange thing: DTMF is set to rfc2833, but is working both on incoming and outgoing calls, it is not working only on calls generated with the Originate AMI command, or with the queue member that point to Local dialplan, as you suggested 2016-06-30 22:53 GMT+02:00 John Kiniston <johnkiniston at gmail.com>: > Looking at your logs it looks like you may need to

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:

Wow! I've been *-ing for about 6 years and had literally no idea about that! I can see a way I could put it to a different use, but it seems to be a bit of a sledgehammer to crack the walnut of "how many current callers" compared to one line of (albeit hacky) dialplan. That's making me sound ungrateful. I don't mean to be! On Sun, 14 Jun 2020, 22:39 Steve Edwards,

Anyone know how to suppress the -- Remote UNIX connection / disconnected messages. I have a monitoring application that calls asterisk from the command line to verify some uptime stats. I would like to not have the console log the connections.. Any ideas are appreciated. Thanks Bryant -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV

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

Can someone point me to a page about writing a text file to call an external number and play a TTS with cepstral? I know it includes the creation of a .call file but beyond that im a bit lost.

I'm trying to make an AMI call. I want to call a number, play an announcement when the call is answered, then call a second number and connect the two when the second call is answered. I an able to make a simple call to two numbers and connect them using the manager API but playing the announcement has me beat. Suggestions anyone? Bruce Ferrell

On May 16, 2016 22:15, "Telium Technical Support" <support at telium.ca> wrote: > > You don't mention a configuration generator (like Elastix/FreePBX) so I'll > assume you are using a plain old vanilla Asterisk installation. In which > case all user/endpoint information is kept in config (ini) files, and no > user/endpoint manipulation is done through the

Hi guys, I'm using Asterisk 1.4 and i need a way to hangup all channels. I want to use the teleyapper system for broadcasting call for security reason but i need that all channels are free when a security call is ready to start! I already search in the old post without success. Can anyone help me? Thanks and sorry for my newbie english -------------- next part -------------- An HTML

Hi all, I recently encountered some unexpected behavior with S4 generics dispatching on `...`, which I described in http://stackoverflow.com/questions/43499203/use-callnextmethod-with-dotsmethods TL;DR: `callNextMethod()` doesn't work in methods dispatching on `...`, and arguments of such methods are resolved differently than the arguments of methods dispatching on formal arguments. Could

Hi. I'm trying to get a list of the channels currently in use on an Asterisk server (1.8.32.1 if it matters) over AMI. I send the command "sip show channels", and I get back a response along the lines of (* used to protect the innocent...): Peer User/ANR Call ID Format Hold Last Message Expiry Peer *8.22.*0.34 02035644444

hi all, can anyone give me a guide on any asterisk admin solution / interface for config management, and monitoring? No database use is intended and I prefer open source. Thanks for support. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160516/98f6e448/attachment.html>

All, I am toying with an idea of using an AGI to be able to 'call' my phone, or phones, in case of severe weather warnings. I have been tinkering with a script that reads from weather underground for the forecast, based off a PHP version of a weather AGI I found on the net. It seems rather trivial to have the AGI as a script, that does nothing unless a condition is met, and

Great, thanks Michael for you quick response! I started off with a question on SO because I was not sure whether this was an actual bug or I was just missing something obvious. I'm looking forward to the patch. Cheers, Andrzej On Thu, Apr 20, 2017 at 10:28 PM, Michael Lawrence < lawrence.michael at gene.com> wrote: > Thanks for pointing out these issues. I have a fix that I will