similar to: Adding custom commands to AMI

On Thursday 06 April 2023 at 18:29:43, Jeff LaCoursiere wrote: > If you just want something easy to use out of the box, install the > FreePBX distro. Given that Steve originally said "I've been using Asterisk, including administering and maintaining it, in some aspect since 2003, but this is the first time I have attempted a from-scratch installation and setup on my own." I

Hi. Is there any way to eliminate AMI manager logins from the logging output (without just turning the log level down and thereby losing lots of other stuff as well)? I'm running Asterisk 13.14.1 as a backend service to LVS/IPVS, and using the AMI login as the "service alive" check to see which backend servers are available to take new commands. This results in lots of [Jun

Hi. (Asterisk 16.2.1) I'm using AMI Originate to initiate calls, and I'm passing some additional data in to the dialplan context using the Variable: parameter. Works fine. https://wiki.asterisk.org/wiki/display/AST/Asterisk+16+ManagerAction_Originate Now I need to do the same thing but from another context in my dialplan, so I was expecting to use the Originate() dialplan command,

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

On Tuesday 22 September 2020 at 13:27:27, Joshua C. Colp wrote: > On Tue, Sep 22, 2020 at 7:37 AM Antony Stone wrote: > > Hi. > > > > (Asterisk 16.2.1) > > > > I'm using AMI Originate to initiate calls, and I'm passing some > > additional data in to the dialplan context using the Variable: > > parameter. Works fine. > > > >

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

With softphone I mean linphone csipsimple or whatever. How should a dialplan lokks like? On 31.01.19 11:26, Antony Stone wrote: > On Thursday 31 January 2019 at 10:59:01, basti wrote: > >> Hello I use this dial paln: >> >> [o2-in] >> exten => o2,1,Answer >> exten => o2,n,Playback(hello-world) >> exten => o2,n,Ringing >> exten =>

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

I am communicating with Asterisk 13.18.3 over the AMI and issue the command: ActionID: 11 Action: command Command: core show calls And the response I get is: Response: Follows Privilege: Command ActionID: 11 --END COMMAND- But where is the call data? What is going wrong on this system? I confirmed the AMI connection has full read/write permissions. Why is the call data

If you just want something easy to use out of the box, install the FreePBX distro. Cheers, j On 4/6/23 11:08 AM, Steve Matzura wrote: > Anthony, > > > No, I had no intention of doing any of those things, for I know not > what they are or why I would need or want to be doing them. Maybe I > should have just stuck with the original idea of installing from > Debian distro.

Is it possible to simply send a message to appear as an AMI message/event, from the dialplan? For example exten =>123,1,ami(myEvent, param1, param2) and in the AMI a message appears like: Event: myEvent Privilege: call,all Channel: PJSIP/misspiggy-00000001 Uniqueid: 1368479157.3 ChannelState: 3 ChannelStateDesc: Up CallerIDNum: 657-5309 CallerIDName: Miss Piggy

On Monday 08 October 2018 at 12:44:43, Yves wrote: > I am looking for an easy way to execute any AGI Command directly from the > dialplan without the need to call an external script. The whole point of AGI is that it calls an external script in order to replace commands in the dialplan. Executing an AGI command without an external script doesn't make sense. Antony. -- There's

Hi. I'm trying to get Asterisk 13 to authenticate when it sends an INVITE, and for some reason it's simply not doing it. I've even resorted to reading the source code to try and work out what I'm doing wrong... In channels/chan_sip.c I find: * SIP Dial string syntax: * SIP/devicename * or SIP/username at domain (SIP uri) * or

On Tue, Sep 22, 2020 at 7:37 AM Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > Hi. > > (Asterisk 16.2.1) > > I'm using AMI Originate to initiate calls, and I'm passing some additional > data in to the dialplan context using the Variable: parameter. Works fine. > > >

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

Can it happen that the routes lead the traffic through another interface? Did you try a packet capture with tcpdump? Do the packets really leave the usb adapter? Can asymmetric routing be in effect? Maybe there were some static routes that disappeared when the adapter was unplugged... On Thu, Apr 20, 2017, 12:41 AM Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On

On Tue, 16 Jan 2018 18:18:18 +0200 Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: > On Tue, Jan 16, 2018 at 11:05:01AM +0100, Paul Neuwirth wrote: > > Hello group, > > > > what is the preferred method to connect to asterisk cli over > > network? I need to run asterisk cli commands remotely. > > As others have mentioned: the manager interface is

On 4/19/17 4:23 PM, Antony Stone wrote: > On Wednesday 19 April 2017 at 23:14:46, Carlos Chavez wrote: > >> On 4/19/17 4:09 PM, Antony Stone wrote: >>> On Wednesday 19 April 2017 at 22:54:51, Carlos Chavez wrote: >>>> I have a server that had been operating for a few years now with >>>> >>>> IAX2 trunks to several other servers.

> Please give us a link to a datasheet for that device. Hello https://www.cisco.com/c/en/us/products/collateral/collaboration-endpoints/unified-sip-phone-3905/data_sheet_c78-651588.html On Fri, Apr 12, 2019 at 3:58 PM Antony Stone <Antony.Stone at asterisk.open.source.it> wrote: > > On Friday 12 April 2019 at 14:42:57, Gokan Atmaca wrote: > > > Hello > > > >

Hi, https://community.cisco.com/t5/ip-telephony-and-phones/cp-3905-asterisk/td-p/1995981 The phone does work, you do need to TFTP the configuration files to the phone though. Doesn't look like custom firmware is required. -- Chris. On Fri, Apr 12, 2019 at 3:29 PM Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On Friday 12 April 2019 at 15:24:27, Gokan Atmaca