search for: commandid

Displaying 6 results from an estimated 6 matches for "commandid".

Did you mean: commanded
2018 Mar 22
2
AMI potential memory leak
...: SIP/192.168.40.105-000000c5 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 1234 CallerIDName: <unknown> ConnectedLineNum: <unknown> ConnectedLineName: <unknown> Language: en AccountCode: 11 Context: ABC Exten: 3002 Priority: 8 Uniqueid: 1521724388.197 Linkedid: 1521724388.197 CommandId: 401382226 Command: EXEC Playback iss/eng/THANK-U&iss/eng/BYE [03/22 08:13:33.643] DEBUG[1228] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: SIP/192.168.40.105-000000c5 ChannelState: 6 ChannelStateDesc: Up CallerIDNum: 1234 CallerIDName: <unknown> Connect...
2018 Mar 21
2
AMI potential memory leak
.... If you look at the event response, the Result field is filled with random characters. I'm not sure what to do because that is a completely random result. It makes no sense. We send the following command to asterisk via AMI Action: AGI ActionID: C44415 Channel: SIP/192.168.40.105-00001338 CommandID: C44415 Command: asyncagi break Asterisk processes it and the Asterisk log shows it sends the following for the AsyncAGIEXEC event... This is coming directly from the Asterisk log so it's clearly Asterisk filling the Result with what appears to be random characters. [03/21 15:44:27.793] DEBUG...
2013 May 08
0
Transfer cmd via AsyncAGI
Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-00000002 CommandID: C8 Command: EXEC Transfer SIP/1003 Destination phone starts ringing. If it answers the call, everything works fine. I am notified the agiexec completed successfully and given a TRANSFERSTATUS of SUCCESS. I am also notified when the call is hungup so that I can cleanup information regarding...
2013 Jul 01
3
Asterisk 1.8.20 AGI function SAY DATETIME does not play anything when mode in say.conf is changed to "new"
Hi I am using following say.conf file. Its a default file, which comes with Asterisk installation. When I call SAY DATETIME AGI function, it simply returns without playing date & time. Where as if I use mode=old setting, it works. Is this a bug or mode=new is not implemented for SAY DATETIME AGI function? [general] mode=new ; method for playing numbers and dates ;
2012 Jun 12
1
puppetdb indicated only facts were replaced, no sign of catalog
...18-9 - /commands] [core.JmsTemplate] Executing callback on JMS Session: Cached JMS Session: ActiveMQSession {id=ID:hadoop4-57420-1339541690768-4:8:1,started=false} 2012-06-12 16:36:05,543 DEBUG [418893110@qtp-1933844018-9 - /commands] [core.JmsTemplate] Sending created message: ActiveMQTextMessage {commandId = 0, responseRequired = false, messageId = null, originalDestination = null, originalTransactionId = null, producerId = null, destination = null, transactionId = null, expiration = 0, timestamp = 0, arrival = 0, brokerInTime = 0, brokerOutTime = 0, correlationId = null, replyTo = null, persistent =...
2015 Aug 24
3
PJSIP add
I am trying to set add a SIP Header to a call before adding it to the Queue. The dial plan sends the call to my macro to perform the work. When I use chan_sip, everything works as expected. When I use PSJIP support, it's not adding the SIP header. Looking at the output, I see the macro is called in both cases. In the PJSIP case, the added sip header never is showing up in the asterisk logs