search for: __dynamic_features

Hi, I think I've identified an issue and just want to check before completing a bug report. Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. AgentA answers and is able to use that feature code. If AgentA performs an attended transfer of a call from a queue to AgentB, the feature code no longer works. Cases that do work are as follows... Calls using both Queue() and Dial() applications, prior to transfer, feature code work...

...alls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489", "__DYNAMIC_FEATURES=") in new stack -- Executing [s at macro-stdexten:2] GotoIf("IAX2/6223-10489", "0?5:3") in new stack -- Goto (macro-stdexten,s,3) -- Executing [s at macro-stdexten:3] Dial("IAX2/6223-10489", "SIP/ 6123&IAX2/6123|20|tTkK") in new stack etc... My q...

On Wed, Aug 8, 2018 at 1:53 PM, Daniel Journo <dan at keshercommunications.com> wrote: > > Prior to a call entering a Queue, I set __DYNAMIC_FEATURES=NewRecordApp. > > AgentA answers and is able to use that feature code. > > If AgentA performs an attended transfer of a call from a queue to > AgentB, the > > feature code no longer works. > > > > It only doesn’t work when using Queue() and an Attended transfer is &...

...ines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake?? [macro-stdexten] exten = s,1,Set(__DYNAMIC_FEATURES=${FEATURES}) exten = s,2,GotoIf($["${FOLLOWME_${ARG1}}" = "1"]?5:3) ----adding starts from here---------------------------- exten = s,3,Set(temp=${DB(CFIM/${ARG1})}) exten = s,4,Dial(Local/${temp}@default/n) ; Unconditional forward exten = s,5,Set(DB(lastcaller/${ARG1})=${CALLE...

I have been trying for a very long time to get asterisk to detect and utilize dtmf tones from my sip clients within my dial scripts. I have set automon=>#9 in my features.conf, I have Dial(....,gWw) in my dial scripts. I have Set(DYNAMIC_FEATURES=automon) as the first script in my extension. I can see the dtmf tones on the wire as SIP INFO packets. Using the Read() app I have verified that * is

Hi there, I need some help with a 'custom' feature. I have following feature defined in features.conf : [applicationmap] opnemencallee => #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m In my dialplan : [from-HostAst] exten => s,1,Set(__DYNAMIC_FEATURES=opnemencallee) exten => s,n,Dial(SIP/grandstream,30) I want the callee to be able to press #3 to be able to record the conversation but when I press these keys on my Grandstream phone, the following is displayed on the CLI : [Sep 7 20:33:49] WARNING[10870]: res_musiconhold.c:665 get_mohbyname...

...t DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. Here's the scenario: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb) exten => _X.,n,Set(__DYNAMIC_FEATURES=in-pauseMonitor#in-resumeMonitor) exten => _X.,n,Dial(SIP/myphone,300,tTo) My [applicationmap] in features.conf is setup as follows: in-pauseMonitor => *7,self/callee,Macro,pause-record in-resumeMonitor => *9,self/callee,Macro,resume-record I also have the following conte...

...callee on > how to open the gate/door. I just realized I said one piece wrong in this. 'gate' is not the context, it is the dynamic feature designator. I can illustrate this better by posting my front gate context. [front_gate] exten => <number gate dials goes here>,1,Set(__DYNAMIC_FEATURES=gate) same => n,Goto(frontgate_queue,${EXTEN},1) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150603/d48aae49/attachment.html>

...transfer (default is #) ;disconnect => *0 ; Disconnect (default is *) automon => *1 ; One Touch Record a.k.a. Touch Monitor atxfer => *2 ; Attended transfer ;parkcall => #72 ; Park call (one step parking) ; Set(__DYNAMIC_FEATURES=myfeature1#myfeature2#myfeature3) ;<FeatureName> => <DTMF_sequence>,<ActivateOn>[/<ActivatedBy>],<Application>[,<AppArguments>[, MOH_Class]] testfeature => #9,peer,Playback,tt-monkeys ;Allow both the caller and callee to play pauseMonitor => #1,sel...

...d AGI Script /var/lib/asterisk/agi-bin/cid_lookup.agi -- SEP001121d89b97: Accessory 'Speaker' is 'OffHook' (0) -- <SCCP/203-0000000c>AGI Script cid_lookup.agi completed, returning 0 -- Executing [internal-defaults at home:3] Set("SCCP/203-0000000c", "__DYNAMIC_FEATURES=internal-move") in new stack -- Executing [internal-defaults at home:4] Return("SCCP/203-0000000c", "") in new stack -- Executing [201 at home:2] Dial("SCCP/203-0000000c", "SIP/201,,t") in new stack == Using SIP RTP CoS mark 5 -- Called 201...

...39; -- Hungup 'DAHDI/7-1' When I don't have the first two lines this is in the log: -- Executing [106 at voicemenu-custom-4:1] Macro("DAHDI/7-1", "stdexten|106|SIP/106") in new stack -- Executing [s at macro-stdexten:1] Set("DAHDI/7-1", "__DYNAMIC_FEATURES=") in new stack -- Executing [s at macro-stdexten:2] GotoIf("DAHDI/7-1", "0?5:3") in new stack -- Goto (macro-stdexten,s,3) -- Executing [s at macro-stdexten:3] Dial("DAHDI/7-1", "SIP/106|20|") in new stack What did I do wrong in adding the...

Hi guys, I'm trying to get blind transfer to work and automatically transfer call to another number on key sequence press. Extensions.conf_snippet [from-pstn] exten => _0399377744,1,Set(__DYNAMIC_FEATURES=blindxfer) exten => _0399377744,n,Set(__GOTO_ON_BLINDXFR=to-pstn ^0388924326^1) exten => _0399377744,n,dial(SIP/0399377704 at c5400-02,T) [to-pstn] Exten => _XXX.,1,dial(sip/0388924326@ c5400-01) Features.conf_snippet [featuremap] blindxfer => #1 on #1 all I get is sile...

Hi Is there a way to disable blind and attended transfer during a call. I am trying this configuration but unfortunately with no luck: - in features.conf [applicationmap] disabletransfer => 9*9,self,GoSub(disabletransfer,s,1) - in extensions.conf [incoming] exten => 99,1,Set(__DYNAMIC_FEATURES=disabletransfer) exten => 99,n,Dial(Sip/alice,120,tT) exten => 99,n,Hangup() [disabletransfer] exten => s,1,Set(FEATUREMAP(atxfer)=) exten => s,n,Set(FEATUREMAP(blindxfer)=) exten => s,n,Set(FEATURE(inherit)=yes exten => s,n,Noop(${FEATUREMAP(atxfer)}) exten => s,n,Noop(${FEA...

...indow from Asterisk: > == Using SIP RTP CoS mark 5 > -- Executing [6201 at DLPN_DialPlan1:1] Macro("SIP/6000-00000013", > "stdexten,6201,SIP/6201") in new stack > -- Executing [s at macro-stdexten:1] Set("SIP/6000-00000013", > "__DYNAMIC_FEATURES=") in new stack > * [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyerror: > ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; > Input: > = 1 > ^ > [Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:472 ast_yyerror: If you...

...read my hints and check for line in use that way. The AGI works fine from a prompt, but returns the dreaded "utils.c:966 ast_carefulwrite: write() returned error: Broken pipe" when I try to run it from the dialplan. Here is my dialplan snippet; [macro-stdexten] exten => s,1,Set(__DYNAMIC_FEATURES=${FEATURES}) exten => s,n,GotoIf($[${FOLLOWME_${ARG1}} = 1]?5:3) exten => s,n,Noop(dial $ARG1 after checking sippeer) exten => s,n,Set(LINESTAT=Idle) exten => s,n,AGI(hintcheck.agi|${ARG1}) exten => s,n,Wait(3) exten => s,n,Verbose(status is ${LINESTAT}) exten => s,n,Got...

Hi all, I'm looking for some serious help. :) I couldn't find a better description for my problem... I think it is quite complex! Here's what I would like to achieve: A SIP caller dials into to my Asterisk 10. He will automatically listen to a specific MP3 stream. Other SIP callers dial also into my Asterisk. They all will automatically listen to the same MP3 stream. All

Hi Kevin. Thank you very much for the hint! It worked very well! Your example ' exten => 1234,1,System(echo "This is a test" >> /var/log/asterisk/test.txt) ' executes when the SIP client (my softphone Jitsi) sends a SIP INVITE to asterisk. So, the softphone tries to establish a session with target 1234. Now, lets suppose my softphone rings and I answer a