similar to: ami

I want to be able to pass any number (variable length) to a context and then forward that to another asterisk server for processing by that servers dial plan.? I have the two talking IAX2 so that part is done. I can also dial a number from the sending to the server asterisk. The problem is I don't want to have to create (duplicate) dial plans at originating Asterisk to equal those at 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:

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,

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

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. 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! I have trouble with connection to AMI 1.1 wich enabled on Elastix "*Asterisk Call Manager/1.1* *Action: Login Username: admin Secret: qweasd123* *Response: Error* *Message: Missing action in request*" Elastix versions: "* Kernel* * Linux(x86_64)-2.6.18-348.1.1.el5* * Elastix* * elastix-2.4.0-1* * elastix-portknock-0.0.1-0* * elastix-agenda-2.4.0-1* *

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,

I am trying to make sure my DID and SIP account details are working properly and engaging the extensions.conf and dial plan. I have a successful SIP session registered: Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922) Asterisk*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time sip3.voipvoip.com:5060

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, 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 ...

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

I have a question about dispatching calls... If I try to dispatch a call on line 1 using the AMI and I check in my table to see if line 1 is available and it is.... So I have done my checking now I dispatch my call and at that same time a call comes in on line 1 and now its no longer available for me to make a call, I connect on AMI and my call fails.... How do I prevent this from happening?

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

Hello List. I am puzzled and how asterisk listens to calls or connections from clients. When I do a netstat -nat I don't see asterisk listening on port 5060. Now, I'm testing a server with three network interfaces: two to the internet doing load balancing and the other to our LAN. I would like asterisk to only accept connections coming from our LAN but, can't find where to configure

Hi. I have three servers running corosync and pacemaker, to maintain a floating address between them. This is working fine, and I can, for example, SSH to the floating address and get to whichever server has the address at the time. I am trying to connect to the same server (using the same address) for AMI, and it just isn't working, even though I can connect to the primary address of

Hi guys, I hope you could help me. I am trying to get connected through AMI but something is not working. Both php code and manager.conf were working well in asterisk 1.4 1. Sometimes it gets connected and sometimes it doesn't: == Connect attempt from '192.168.25.241' unable to authenticate == Connect attempt from '192.168.25.241' unable to authenticate == Manager

Hi, We recently had a customer that set up Asterisk with port 5038 open to the world with standard configs for the AMI (by that I mean they copied and pasted configs that they saw online). Digging around a bit it seems the attacker used the AMI action "pjsip show auths" followed by "pjsip show auth <peer name>" which got them the credentials to their account. I know we

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