similar to: AMI vs. Dialplan Originate

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. > > > >

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. > > >

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") Thank you!

25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: >> I want to change call files, which has caller id in them, to call >> originate from dial plan. >> But I don't see such parameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate >> >> How can I pass callerid

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

HiCould someone please help me how to set Custom generated UUID in Originate action in AMI ? RegardsBabak -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150508/528d5ff1/attachment.html>

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

i have problem to setup application meetme. i'm using asteisk-1.0.3 and sjphone as client. ++++++++++++++++++++++++++++++++++++ monchemin --------------------------------- Do you Yahoo!? Dress up your holiday email, Hollywood style. Learn more. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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. 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 all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

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. https://www.voip-info.org/wiki/view/Asterisk+manager+API says that "There are a finite (but extendable) set of actions available to the client, determined by the modules presently loaded in the Asterisk engine." Can anyone point me at some appropriate documentation for adding custom commands to the AMI to extend the available actions? Thanks, Antony. -- This sentence

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

​I need to launch a remote process at the machine that has the dialer. I could hard-code the IP address in a global variable, but It would be much more elegant if the dialplan would have a "manager" object where I could read "manager-->connected". ​ -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Thursday 31 August 2017 at 18:15:54, Joseph Smith wrote: > I was hoping Asterisk would handle more than 4k simultaneous calls. I know from experience that Asterisk can handle more than 4k simultaneous calls, however it's an extreme case to have all of them playing music on hold. I think that if you tested 4k simultaneous calls with standard media streams on the majority of them, you

Hi, I am experimenting with getting hold of the sip cause and sip response from outgoing call. How could i make a userevent printing the sip cause and/or sip response. I have tried using hangupcause, sip_cause and such , but i am not getting any data. I would at least like to use the q.850 reason codes in the dialplan which i now am unable to do. Any help appreciated. [Beskrivning: Fogwise -

> exten => _13XXXXXXX,1,dial(${OPERATOR},20) Hello "SIP/2.0 401 Unauthorized" Unfortunately the negative. An asterisk indicates a 404 error. On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle <support at drdos.info> wrote: > > On 3/5/19 2:46 AM, Gokan Atmaca wrote: > > Asterisk can send calls, but I don't get a call. What could be the problem? > > >

On Thu, Jun 14, 2012 at 11:44 PM, Jacob Carlborg <doob at me.com> wrote: > On 2012-06-15 07:25, Chandler Carruth wrote: >> Figured it was good to ask this as James is busy making sure the Clang >> doxygen is actually all well formed, and reasonably linked together so >> we have proper docs for the ever growing number of Clang library users... >> >> There are