Im giving HangupCauseClear() a try on a Debian Stretch / Asterisk 13.18.3stack.My dialplan is:exten = 1234,1,Set(CHANNEL(hangup-handler-push)=myhandler,s,1)same = n,Dial(SIP/foo/1234)same = n,Gosub(myhandler,s,1)same = n,HangupCauseClear()same = n,Dial(SIP/bar/1234)[myhandler]ex..
AllI have this system here where:# dahdi_hardware pci:0000:07:04.0 wctdm24xxp+d161:8005 Wildcard TDM410Ppci:0000:07:09.0 wcte11xp+e159:0001 Digium Wildcard TE110P T1/E1 Boardchannels are 0-31 for TE110Pand 32-35 for TDM410PI want to insert a new TE11..
HiIs there any way to timeout AsyncAGI if there is no activity on channel for defined period? I wish to send call to alternate route if there is no activity on channel for defined period.Thanks & Regards, A..
While I am executing a Macro on the called channel, right after the call connects, I need to execute an app on the master channel, from inside that macro, specifically, SendDTMF. If I execute it now, it send a text message to the Callee, when..
I run in a weird issue with a BLF application I have written… this application is just receiving events from Asterisk Manager Interface and blink the lights accordingly. All almost work perfectly, except when a pickupexen is used when multiple extensi..
all,im writing because going crazy on this issue im unable to solve. My VoIP system is based on OpenSIPS router that forward calls to an Asterisk BOX to have IVR and Queue services.If a call was directed to a queue and operator answer, on transfer..
Greetings everyone, I am attempting to adjust the volume of a call using Set(VOLUME) in my extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) have no discernable effect on my endpoints (Snom 300 IP phones). I have tried sett..
againId like to configured my Asterisk to use german sounds for theSay-commands…I installed the sounds-files and I tried them withPlayback(de/demo-echodone) and it works.Now I tried to add an extension to say the current time:exten => 24,1,Verbose(2,T..
, Im trying to use CHANNEL(aor) and CHANNEL(contact) on PJSIP channel, on asterisk-13.3.2, but they dont return anything. Is this a bug, or did I miss something? Here is my test dialplan: exten => *98,1,Answer same => n,NoOp(Channel=,type= ) same..
It seems this variable was removed, since I get an error when I execute this code Set(CHANNEL(SAY_DTMF_INTERRUPT)=1)Please clarify what happened to this option and if the behavior can be achieved some in ..