Archives : January-2021
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I am unable to figure out why I am not able to blind transfer when I am the caller and I call the extension defined by gosub. When running asterisk -rvvv I can see: — Channel PJSIP/-00000009: Dialed @gosub-stdexten does not exist. It is evident th..
how can I separate individual calls in my asterisk log? I noticed that a call always starts with following line: VERBOSE[C-0000000c]: Using SIP RTP CoS mark 5 How reliable is this as a separator? It seems it appears twice during every call. And, w..
after upgrade from asterisk 11 to 16 i have problem with ForkCDR app (probably) snippet of dialplan EXTEN=800800800 backup_number=666777888 exten => _X.,n(forward800),noop(forward800) exten => _X.,n,Gosub(routing800,s,1(${EXTEN})) exten => _X.,n,goto(pstn,${backup_number}..
Did execution of macro changed in Astersik-16.15 ? When I try to dial an extension that call macro I get an error: app.c:280 ast_app_exec_macro: Cannot run Macro(atb).The application is not available. Dial(SIP/718xxxxxxxxxx@pstn-5665,20,m(default)M(at..
Following the playback.js ari-client example, I now need to store the current playback offsetms, either when it was skipped or hung up on. But I cant seem to find it. I know that https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Application_ControlPlayb..
Very simply, I want to pipe some external audio into a channel (bridge)using the externalMedia channel option. Running Asterisk 18 on ubuntu, heres what I did to try and test things out:open a console tab vlc -vvv https://media-ssl.musicradio.com/LB..
I configured MWI with pjsip. The aors section contains: mailboxes = 101@ The endpoint section contains: context = internal mailboxes = 101@ The dialplan leaves the voicemail by: exten => s-NOANSWER,2,VoiceMail(${vmbox}@|u) or: exten => s-BUSY,2,VoiceMail(${vmbox}@..