search for: varset

Hello gurus, I have some variables, and i am creating combinations for analysis in the end i need these variables to be displayed like "LEPTIN+SAA+PTH". currently i am using loop to perform this. I would appreciate any pointers to do it without the loop. > mols=c("LEPTIN","SAA","PTH","sEGFR") > samples=mols[1:3] > samples [1]

...event: Event: MeetmeLeave Privilege: call,all Channel: SIP/1001-000000b3 Uniqueid: 1384275118.914 Meetme: 65001 Usernum: 1 CallerIDNum: 110 CallerIDName: device ConnectedLineNum: 110 ConnectedLineName: <unknown> Duration: 64 [Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/1001-000000b3 Variable: MEETMESECS Value: 64 Uniqueid: 1384275118.914 [Nov 12 16:53:02] DEBUG[23332] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: SIP/1001-000000b3 Variable: RTPAUDIOQOS Value: ssrc=797521620;themssrc=278781419;lp...

...st<-as.data.frame(cbind(round(runif(50,0,5)),round(runif(50,0,3)),roun d(runif(50,0,4)))) sapply(test, table)->vardist sapply(test, function(x) round(table(x)/sum(table(x))*100,1) )->vardist1 par(mfrow=c(1,3)) sapply(vardist1, function(x) barplot(x, ylim=c(0,100),main="Varset1",xlab=names(x))) par(mfrow=c(1,1)) Names don't show up although names(vardist) works. Also I would like to put a single Title on this plot instead of repeating "Varset" three times. Any hints appreciated. Thanx Herry

...5] DEBUG[18506] chan_sip.c: SIP attended transfer: Transferer channel SIP/Lync-00000003, transferee channel DAHDI/i1/500-2 [Mar 4 08:21:05] DEBUG[18506] chan_sip.c: Got SIP transfer, applying to bridged peer 'DAHDI/i1/500-2' [Mar 4 08:21:05] DEBUG[18502] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: DAHDI/i1/500-2 Variable: SIPREFERRINGCONTEXT Value: from-internal Uniqueid: 1299244801.4 [Mar 4 08:21:05] DEBUG[18502] manager.c: Examining event: Event: VarSet Privilege: dialplan,all Channel: DAHDI/i1/500-2 Variable: SIPREFERREDBYHDR Value: Uniqueid: 1299244801...

...terisk 13.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: Improvements made in this release: ----------------------------------- * ASTERISK-25256 - [patch]Post AMI VarSet to empty string events when Asterisk deletes a dialplan variable. (Reported by Richard Mudgett) * ASTERISK-25067 - Sorcery Caching: Implement a new caching module (Reported by Matt Jordan) * ASTERISK-25040 - pbx: Improve performance of reloads by making hint destruction mo...

...is sent to the gateway with a string as payload to enable > half duplex > > communication. So, we were planning to run an AGI script with meetme > (AGI_BACKGROUND) to > > receive the MESSAGE (using AGI command 'RECEIVE TEXT') from both ends and > to generate a > > VarSet AMI event. > > > > Operator (wants to talk) -> SIP:MESSAGE ->MeetMe(asterisk)-> SIP:MESSAGE > -> radio gateway > > And vise versa. > > > > Any suggestions on the above scenario. > > I don't think it can be done without making modifications to A...

...: Add option to suppress the Message channel AMI and ARI events In order to reduce the amount of AMI and ARI events generated, the global "Message/ast_msg_queue" channel can be set to suppress it's normal channel housekeeping events such as "Newexten", "VarSet", etc. This can greatly reduce load on the manager and ARI applications when the Digium Phone Module for Asterisk is in use. To enable, set "hide_messaging_ami_events" in asterisk.conf to "yes" In Asterisk versions <18, the default is "no" pre...

Hello List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

...Batch mode: size used as time value after first expire (Reported by Shane Blaser) * ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to softmix sometimes fails to properly re-INVITE remotely bridged participants (Reported by Matt Jordan) * ASTERISK-24415 - Missing AMI VarSet events when channels inherit variables. (Reported by Richard Mudgett) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24122 - Documentaton for res_pjsip option use_avpf needs to be fixed (R...

...Batch mode: size used as time value after first expire (Reported by Shane Blaser) * ASTERISK-24327 - bridge_native_rtp: Smart bridge operation to softmix sometimes fails to properly re-INVITE remotely bridged participants (Reported by Matt Jordan) * ASTERISK-24415 - Missing AMI VarSet events when channels inherit variables. (Reported by Richard Mudgett) * ASTERISK-24063 - [patch]Asterisk does not respect outbound proxy when sending qualify requests (Reported by Damian Ivereigh) * ASTERISK-24122 - Documentaton for res_pjsip option use_avpf needs to be fixed (R...

...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> ConnectedLineNum: <unknown> ConnectedLineName: <unknown> Language: en AccountCode: 11 Context: ABC Exten: 3002 Priority: 8 Uniqueid: 15217243...

...RI host/port to 85.142.148.80:5060 [2016-02-10 22:58:20] DEBUG[25442] netsock2.c: Splitting '185.45.152.174' into... [2016-02-10 22:58:20] DEBUG[25442] netsock2.c: ...host '185.45.152.174' and port ''. [2016-02-10 22:58:20] DEBUG[25553] manager.c: Examining AMI event: Event: VarSet Privilege: dialplan,all Channel: PJSIP/admin-00000254 ChannelState: 4 ChannelStateDesc: Ring CallerIDNum: 77777716 CallerIDName: admin ConnectedLineNum: 34 ConnectedLineName: <unknown> Language: ru AccountCode: 9999 Context: context-demon-int Exten: 34 Priority: 3 Uniqueid: 1455134292.635 Lin...

We are communicating with Asterisk via AMI. Running Asterisk version 13.18.5 on an Ubuntu box. 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

...25257 - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope (Reported by Patric Marschall) * ASTERISK-24934 - [patch]Asterisk manager output does not escape control characters (Reported by warren smith) * ASTERISK-25255 - Missing AMI VarSet events when setting to an empty string. (Reported by Richard Mudgett) * ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park. (Reported by Richard Mudgett) * ASTERISK-25183 - PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite...

...25257 - [patch]channels/sig_pri.h -> sig_pri_span -> force_restart_unavailable_chans in wrong scope (Reported by Patric Marschall) * ASTERISK-24934 - [patch]Asterisk manager output does not escape control characters (Reported by warren smith) * ASTERISK-25255 - Missing AMI VarSet events when setting to an empty string. (Reported by Richard Mudgett) * ASTERISK-25254 - Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park. (Reported by Richard Mudgett) * ASTERISK-25183 - PJSIP: Crash on NULL channel in chan_pjsip_incoming_response despite...

...Marschall) - [ASTERISK-24934 <https://issues.asterisk.org/jira/browse/ASTERISK-24934>] - [patch]Asterisk manager output does not escape control characters (Reported by warren smith) - [ASTERISK-25255 <https://issues.asterisk.org/jira/browse/ASTERISK-25255>] - Missing AMI VarSet events when setting to an empty string. (Reported by Richard Mudgett) - [ASTERISK-25254 <https://issues.asterisk.org/jira/browse/ASTERISK-25254>] - Crash if dialplan sets ATTENDEDTRANSFER to an empty string before Park. (Reported by Richard Mudgett) - [ASTERISK-25183 <https:...