Category : Asterisk Users
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Im trying to join a user (at SIP/99) into a conference via REST/ARI.Iwant the PBX to call the user, and then join him into an existing conference. I have created a conference in FreePBX with number 1234, and name conf. Conceptually the steps I have..
Im monitoring the ARI, and if extension 1 calls extension 2, it seems that extension 2 enters the bridge first, then extension 1 enters the bridge. Can I safely (always) determine who initiated the call by who is the latest endpoint to enter the bridge..
everyone. I allow myself to submit a problem that I can not solve with my VOIP provider Orange in France [2023-06-08 13:19:03] ERROR[185091]: res_pjsip/pjsip_configuration.c:1044 from_user_handler: Error configuring endpoint Biv_Sortie – from_user fi..
Im setting up voicemail on my answering-machine project. Since the directory for voicemail messages for an extension doesnt exist until theres a message to be saved therein, how can I create a custom greeting since it goes in that directory? Thats w..
I have the ARI enabled on my Asterisk test box, and want to listen to all events.I cant find the syntax to do that.Can I only listen to events related to a stasis app? I was hoping that a simple wscat command like this would show me all events: ws..
all Using asterisk 16.25 I was trying to stop Mixmonitor using features. The code is executed but I realized that I was executing StopMixmonitor from another channel so I opted to use AMI. When I call MixMonitor I store the channel name in a var ..
Who controls how many times an incoming call from an external (DID) provider will ring before Asterisk picks up the call and handles it internally–the provider or Asterisk? If its the DID provider, Ill work on that with them; if its Asterisk, I di..
Acording to the book, Im supposed to put things into what Asterisk thinks is its default audio file location, /var/lib/asterisk/sounds, and Im supposed to be able to create a custom directory off of that path and use it in a relative-syntax way in ..
Doug from this list got me to change my connectivity to my DID provider from SIP to IAX, and bingo, it all just worked instantly. For my next trick: setting up voicemail. The book does it all with smoke and mirrors (SQL), but Im fresh outa those,..
when I call my conference, I see this error in my logs: ERROR: Function DENOISE not registered here is snippet from extensions.conf … same => n,Set(CONFBRIDGE(user,announce_join_leave)=yes) same => n,Set(CONFBRIDGE(bridge,record_conference)=yes) s..