similar to: channel ending with /n

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..

I have an app which handles a Mitel's command port to change the MWI lights. It detects dial tone, plays some DTMF digits, listens for dialtone-or-busy, does a manager event on what it finds, and returns. Since the Mitel command port does not give answer supervision (looks like it's ringing), and since I run this app via a AMI "originate" command, I set up an extension in

Asterisk 12.5 I'm using AMI to initiate a "call me now" feature from the web site. The AMI looks like: Action: Originate Channel: Local/s at callmenow Context: dial-to-customer Exten: s Priority: 1 Async: true Variable: CHANNEL_TO_CUSTOMER=SIP/voipms/1112223333 Timeout: 999999 Dial Plan: [callmenow] exten => s,1,NoOp(callmenow: Queue without answer) same

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

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:

Hello Friends, I have successfully being able to initiate a automatic Call with AMI that leads me to a Extension XXX. In my extension.conf I have: exten => XXX,1,ChanSpy(SIP/). The problem that I have is to listen to a Specific channel that's using SIP. I tried out this: exten => XXX,1,Read(SPYNUM,extension) exten => XXX,n,ChanSpy(SIP/${SPYNUM},q) It asks for a specific

Hi All, I would like to know if you can confirm that, if using origination via AMI, as documented here: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate it is not possible to set the max duration of a call. I mean: what you would do with the L (limit) parameter of the command Dial, is not possible when originating. As well as using the absolute timeout, as documented

El 07/04/15 a las 17:38, Alex Villac??s Lasso escribi?: > I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution. > > Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred

Hello User-List, is it possible to play a sound file directly to a caller channel? Like this AMI command Action: Originate Channel: SIP/20-00001d41 Application: Playback Data: /path/to/audio/file I get an Error Message. My intension is to play a sound file to a caller and the other callers don't hear this. Can someone help me ? Thanks a lot Bye Daniel

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 ...

Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension "exten => dial,1,Dial(SIP/902)" 4. 902 rings, then answers 5.

Folks, I am strugging with Asterisk 1.4 Vs 1.6 differences over AMI Originate? Here is the pastebin... http://pastebin.ca/1805594 Not sure why the local channel won't send to context while the remote channel does. Worked fine in 1.4 but 1.6.1 has issues. Any help? Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, (please excuse me for lack of proper jargon usage and the vagueness of description...) i use Asterisk 11.12.1, (well... as included in FreePBX), . . . The softphones are mostly on machines without proper sound hardware (no mics, no speakers/headsets); This is partly because the workforce is quite conservative in what they want to use :) meaning handsets are important; As the handsets have

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

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

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

>> 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

On Sun, Jul 2, 2023 at 4:18 PM TTT <lists at telium.io> wrote: > >> 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

On 12/3/19 3:04 PM, Joshua C. Colp wrote: > >     The AMI command, after the login, looks like this: > > > >     Action: Originate > >     Channel: SIP/outgoing/%%(destination)s > >     Context: LocalSets > >     CallerID: Vybe Consulting Inc Fax Service <5555551212> > >     Exten: sendfax > >   

Hi Ana, I have exactly the same problem as you! Anyone can help us? Thanks in advance, Nuria >Hello all! > >I'm using Samba 2.2.1-a4 on Red Hat 7.1 as PDC. Everything is working >fine but 2 issues: > >1. When accesing shares on Win98 system from Network Neighborhood on a >NT wks, I get an Access Denied message when double-clicking the Win98 >box. >I can map