similar to: actionID on manager events

Is it well known that setting the ActionID when connecting to AMI has absolutely no effect? Is this fixed in Asterisk 1.4? If you add an ActionID to your Originate command for example, it looks like the only events that come back with an ActionID associated are the initial response, OriginateSuccess and OriginateFailure. That's it. No other events have an ActionID associated. This pretty much

hi, afaik, the action-id provided with the OriginateAction should only show up in the OriginateSuccess or OriginateFailure event. Intermediate events that are generated when the channels are create will NOT carry the action-id of the originate. The async flag tells asterisk to process originates in parallel, i.e. if you have two users originating calls and NO async flag set, the second originate

Skipped content of type multipart/alternative-------------- next part -------------- Reliably Transmitting (no NAT) to 111.111.111.50:16666: INVITE sip:111.111.111.50:16666 SIP/2.0 Via: SIP/2.0/UDP 111.111.111.8:5060;branch=z9hG4bK360502ae;rport From: "asterisk" <sip:asterisk@111.111.111.8>;tag=as348de10b To: <sip:111.111.111.50:16666> Contact:

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

Hi, I'm currently trying to originate a call with 2 variables set. I tried doing it via manager API and call File and both failed, because the vars were not separated. I'm using Asterisk 1.2_beta1 on this machine Can anyone here verify wether this is a bug or just a stupid error on my part? This is the callfile I tried to use, after the manager way failed: Channel:

Not sure why, but Vitelity changed the settings to IP based authentication on me. Here's the new sip.conf settings they sent me. type=friend dtmfmode=auto host=64.2.142.93 allow=all nat=yes canreinvite=no trustrpid=yes sendrpid=yes When I use these settings to originate calls using the sip.conf they sent me, everything works. Action: Originate ActionID: S8 Channel:

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

Hi all, When the 'Originate' command is issued with 'Async' open set to 'yes', I got the response right away with the correct 'ActionID'. What follows is the 'Newchannel' event with a 'Channel' ID, but their is no 'ActionID' to tie it back to the command. How do you guys deal with this? -------------- next part -------------- An HTML

Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will

Hi. I'm making calls via the Manager OriginateAction. My action is set to be async and therefore I receive originiate events. Within the originate event that I receive there is a reason code. In the event of failure I need to dermine why the call failed (no pickup, rejected, no such number, circuit busy, ect) and inform the user with a meaningful message. I assume that one is suppose to

I installed version 1.6.0.3-rc1 and my AMI application stopped working. I reinstalled 1.6.0.1 and it worked again. I reinstalled 1.6.0.3-rc1 and it stopped. Looks like a problem in the software to me. Following the same steps using the same code for the AMI and conf files for * I get bad behavior in 1.6.0.3-rc1 and good behavior in 1.6.0.1. I have this action: Action: Originate Channel:

Apparently there are subtle diferences between meaning of MeetmeJoin event depending on channel type. Problem is: after originating a call from channel to MeetMe room i.e.: [meetme] exten => 1,1,Answer exten => 1,2,Meetme(kolejka|dqM) than: Context: meetme Exten: 1 Priority: 1 ActionID: 1077925740-00000004 Timeout: 5000 Action: Originate Async: true Channel: somechannel I get eventually

I am using AMI Originate to perform a new outbound call. The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header. For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being passed through the PJSIP INVITE header Action: Originate ActionID: S598 Channel: PJSIP/133 at 1002

Hello All, I'am a new Asterisk user, and i have the following question. The following is the Status of all open channels from my Asterisk system, which was received through the Asterisk Manager Interface ((AMI)). ==================================================================== action: Status actionid: 65066874_3# Response: Success ActionID: 65066874_3# Message: Channel status will

Hello list, Can someone please point me out why would a stream like the following only write ONE line (the first) on the given file? Action: login Username: test Secret: 123456 Action: UpdateConfig SrcFilename: voicemail2.conf DstFilename: voicemail2.conf Action-000000: Append Cat-000000: default Var-000000: 127 Value-000000: >5555, Jason Bourne97, jason25 at noCia.gov.do ActionID:

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

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

My application fires several calls thru AMI ORIGINATE command. For example if I have 3 operators I do 3 ORIGINATEs. My trouble is when one operator quit for some reason, I should kill the corresponding ORIGINATE. Of course, I could let the call ring and hangup after the customer pick-up. But this is not the case, I do have to kill the corresponding ORIGINATE. I could execute a soft hangup,

Dear all, I am from Saudi Arabiya and I am using digium hardware with asterisk 1.6. When I am executing following AMI originate API. Orginate start to execute extenstion without knowing of PSTN(FXO) channel is ringing. Any one can help me to resolve this issue ? Action: Originate Channel: Dahdi/g0/2923878 Context: outbound-ivr Exten: 1234 Priority: 1 ActionID: ABC45678901234567890

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at