search for: newexten

...terisk. I have a dialplan, [default] exten => 111222,n,Set(fu_callerid=141688xyxzz) exten => _X.,n,NoOp(Callerid ${fu_callerid}) exten => _X.,n,wait(2) exten => _X.,n,Answer() ? When, ?Answer Application is called AMI Event is triggered like this.. ? ? ? ? ? 'Event' => 'Newexten', ? ? ? ? ? 'Privilege' => 'dialplan,all', ? ? ? ? ? 'Channel' => 'IAX2/X.X.X.X:4572-5011', ? ? ? ? ? 'Context' => 'default', ? ? ? ? ? 'Extension' => '111222', ? ? ? ? ? 'Application' => 'Answer', ?...

...Here is an example (assuming the asterisk-xmpp account is configured) : context gtalk-in { s => { NoOp(Caller id : ${CALLERID(all)}); Answer(); JabberSend(asterisk-xmpp,${CALLERID(name),Please enter the number you wish to call); JabberReceive(${CALLERID(name)},NEWEXTEN); JabberSend(asterisk-xmpp,$(CALLERID(name),(Calling ${NEWEXTEN} ...); Dial(SIP/${NEWEXTEN); Hangup(); } } In this example, when Asterisk receives a GoogleTalk voice call request from a GoogleTalk buddy, it answers the call, and asks the buddy to enter a number over an...

...IAX2/teenlighthouse/16384 Callerid: DAQE Dialer Uniqueid: 1106323559.4 Event: Newchannel Channel: IAX2/teenlighthouse/16384 State: Ringing Callerid: DAQE Dialer Uniqueid: 1106323559.4 Event: Newstate Channel: IAX2/teenlighthouse/16384 State: Up Callerid: DAQE Dialer Uniqueid: 1106323559.4 Event: Newexten Channel: IAX2/teenlighthouse/16384 Context: askdaqe Extension: 100 Priority: 1 Application: Playback AppData: vm-dialout Uniqueid: 1106323559.4 Event: Newexten Channel: IAX2/teenlighthouse/16384 Context: askdaqe Extension: 100 Priority: 2 Application: Dial AppData: Zap/g2/14356355785|10|tT Uniquei...

...gt; Paused: 1 Event: QueueMemberPaused Privilege: agent,all Queue: 140 Location: Local/402 at default/n <mailto:%20Local/402 at default/n> Paused: 0 Event: Newchannel Privilege: call,all Channel: SIP/401-08197170 State: Ring CallerID: 401 CallerIDName: JOSEP Uniqueid: 1196854142.4 Event: Newexten Privilege: call,all Channel: SIP/401-08197170 Context: default Extension: 140 Priority: 1 Application: Answer AppData: Uniqueid: 1196854142.4 Event: Newstate Privilege: call,all Channel: SIP/401-08197170 State: Up CallerID: 401 CallerIDName: JOSEP Uniqueid: 1196854142.4 Event: Newexten Privilege...

...quot;, and the data asterisk needs to fetch from the AGI are set on time. But sometimes not, especially in cases like above, when there are only a few dialplan lines between UserEvent and AGI ... In order to handle that, I thought "let's make a sync/meeting point, with the help of the AMI NewExten event, when the app is AGI". The idea would be to keep the AGI connection open as long as the good AMI NewExten event is not received, then to reply and close it, in order for the dialplan to proceed. However, when trying to do this, nothing more occurs on the AMI connection, thus I come to a...

...nt back to him, asking him to enter a number to call. And that's it, Asterisk just relays the call to the configured destination (here, a registered SIP phone). context jingle-in { s => { Answer(); SendText(Please enter the number you wish to call); Set(NEWEXTEN=${JABBER_RECEIVE(asterisk-xmpp,${CALLERID(name)})}); SendText(Calling ${NEWEXTEN} ...); Dial(SIP/${NEWEXTEN); Hangup(); } } Thanks, Philippe

...ch Dialplan function right after the channel is open. Is it safe to launch JACK_HOOK to a channel that just started to ring? (from AMI)? I doing it from the dialplan, because there is no interaction (yet) from the AMI with our asterisk. Version 1.6.2.7 human_now: 2010-05-20 01:42:03.567385 Event: Newexten Privilege: dialplan,all Timestamp: 1274308923.567385 Channel: SIP/Prov6-000001be Context: NPDB2 Extension: 37062646666 Priority: 75 Application: Dial AppData: SIP/GW1/00737062646666,60,M(connect-jack,737219) Uniqueid: 1274308923.446 human_now: 2010-05-20 01:42:03.568501 Event: Dial Privilege: call...

...me: Fake Name Uniqueid: 1175271459.2288 CID-CallingPres: 0 (Presentation Allowed, Not Screened) Event: Newcallerid Privilege: call,all Channel: Local/201@from-sip2-3974,2 CallerID: 201 CallerIDName: Fake Name Uniqueid: 1175271459.2289 CID-CallingPres: 0 (Presentation Allowed, Not Screened) Event: Newexten Privilege: call,all Channel: Local/201@from-sip2-3974,2 Context: from-sip2 Extension: 201 Priority: 1 Application: SIPAddHeader AppData: Alert-Info: AA Uniqueid: 1175271459.2289 Event: Newexten Privilege: call,all Channel: Local/201@from-sip2-3974,2 Context: from-sip2 Extension: 201 Priority: 2 Ap...

...': 'IAX2[kamyk]/3', 'Uniqueid': '1089842404.75'} 2004-07-14 23:59:36,909 DEBUG Hangup, {'Cause': '0', 'Event': 'Hangup', 'Channel': 'IAX2[kamyk]/3', 'Uniqueid': '1089842404.75'} 2004-07-14 23:59:36,918 DEBUG Newexten, {'Uniqueid': '1089842434.76', 'Extension': 'failed', 'Priority': '1', 'Context': 'meetme', 'Event': 'Newexten', 'Channel': 'OutgoingSpoolFailed'} 2004-07-14 23:59:36,919 DEBUG Hangup, {'Cause':...

...rceforge.net/projects/astrxtools4osx/), for which I am trying to get my head around the Asterisk Manager Interface. I had been using the Event: NewCallerid to detect a new call which my Asterisk server doesn't seem to send to the socket anymore, because of which I have reverted to using Event: Newexten. Which is the most efficient way of monitoring if a new phone call is coming my way? Also my application will only monitor a single extension, should I filter the requests on the client side or can a manager interface user be restricted to a single extensions events. Thanks for your time. -- &q...

...outgoing calls. The callerid for the *called* person got changed to one of my own numbers, after the channels git linked. After looking into the flow of events i saw that asterisk keeps sending an NewCallerID Event *after* the Linked Event. See below: 1) NewChannel: Myself the Caller 2) some NewExten Events 3) NewCallerID: I set my CallerID to e.g. 111 4) NewExten: The Execution of the Dial App 5) NewChannel: The Channel for the Callee, the Callerid is set to an value from a former Call 6) NewState: Channel of the Callee is in Dialing State 7) Dial: The two Channels gets to know each other 8)...

...gt; know if anyone's using them first. > > > > Thanks! > > Hi George, > > could you give us a summary list of the impacted AMI messages? More > specifically, are there AMI messages explicitly generated by > Message/ast_msq_queue? Or are we talking about Newchannel, NewExten and > other messages implicitly sent on the AMI because Message is a channel > like any other? > > Note: please keep me in CC, I am not subscribed to asterisk-users > mailing list. > Here's a copy of the commit message which should explain things... message.c: Add option...

I used to be able to pass dial strings to IAX2 providers with # characters at the end of the string. This is how we end dial strings for international calls. So, I would like to be able to selectivity chop off any # characters at the end of string, only if they exist. Basically as follows (chopping off the leading '9' with ${EXTEN:1} syntax: EXTEN from Phone EXTEN for Dial String

...RISK-24825 - Caller ID not recognized using Centrex/Distinctive dialing (Reported by Richard Mudgett) * ASTERISK-24739 - [patch] - Out of files -- call fails -- numerous files with inodes from under /usr/share/zoneinfo, mostly posixrules (Reported by Ed Hynan) * ASTERISK-23390 - NewExten Event with application AGI shows up before and after AGI runs (Reported by Benjamin Keith Ford) * ASTERISK-24786 - [patch] - Asterisk terminates when playing a voicemail stored in LDAP (Reported by Graham Barnett) * ASTERISK-24808 - res_config_odbc: Improper escaping of backslas...

; ; Display certain channel variables every time a channel-oriented ; event is emitted: ; ;channelvars =3D var1,var2,var3 So if you want fu_callerid, set: channelvars =3D fu_callerid And, once that variable is set, you should get a NewExten event, you should see the following key/value pair: ChanVariable(SIP/1234-00000001): fu_callerid=3Dfoobar --=20 Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org

...el variables every time a channel-oriented<br> ; event is emitted:<br> ;<br> ;channelvars =3D var1,var2,var3<br> <br> So if you want fu_callerid, set:<br> <br> channelvars =3D fu_callerid<br> <br> And, once that variable is set, you should get a NewExten event, you<br> should see the following key/value pair:<br> <br> ChanVariable(SIP/1234-00000001): fu_callerid=3Dfoobar<br> <br> <br> --<br> Matthew Jordan<br> Digium, Inc. | Engineering Manager<br> 445 Jan Davis Drive NW - Huntsville, AL 35806 -...

Hi all, what's your opinion about CDR-Event (like Hangup or Ring etc.) on AstManager ? Or, is something like this already implemented ? Regards, Thomas

...her context: [play_response] exten = s,1,Playback(/etc/asterisk/sounds/response-${RESPONSE}) exten = s,2,Hangup My service program first uses Setvar to set RESPONSE to the name of a recording, then Redirect to transfer the channel to that context. This is where it fails. I see no "Newexten" events indicating the beginning of playback, and the channel (as seen from output of the Status action) does not appear to change state. The only thing that follows is a hang up at timeout. Any ideas? David.

...e, everything seems perfect! After the incomming call, i fire outgoing call using originate again, it works now, my phone can ring, i also can pick up it(I seems originate did not create a new Zap channel,just used an exsiting channel?). But the second problem produced, i received the Dialing, UP, Newexten events before my phone ringing. It is supposed that i send an originate command (like Dial application), the last state should be Dialing... until i pick up my phone or timeout. These problems only for Zap channel, if i fire a outgoing call to SIP channel, it works well. What wrong with me ? Here...

hi i logged on to manager API from other terminal by telnet IPADDR 5038 now logged in with username mark let's say this connection Window A now i opened another connection with Manager API with same usename lets say this window B now if i give a command like originate,Redirect through window A connection , can i able to see its response:success/failure Originate:failed/succesfully