Archives : June-2023
Based on postings it should be possible to get the SIP Call-ID header value from the ARI.At what point is this value available ?As well, how do Iretrieve that value – something like GET /channels/{channelId}/pjsip_..
You all know the story–give the customer/client what they ask for, and if they like it, theyll be back for more. Such is just so with my one-trick-pony answering-machine project. Now the other two musicians in my virtual band want the following capabiliti..
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..