Archives : June-2023
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Im learning about WebRTC clients, and am wondering why Asterisk treats them differently from any other SIP client. The media (RTP) should be no different, so the only difference should be on the signaling side.I noticed that the Asterisk wiki menti..
Im looking at using Asterisk 20 with WebRTC clients (sipjs).I know the media runs over TCP, but what about the signaling? I read something about signaling over UDP was proposed as part of a webrtc standard, but cant find if that was ever ratified..
Howdy, Has anyone worked on a Mitel-2000 emulation for PMS integration (Hotel mgmt systems)? Hoping to get my hands on the protocol definition (RS-232!!) for check-in/check-out/housekeeping/CDR, but if someone has already done I would totally buy ..
Ive split this thread off from another (PJSIP authentication) because I think the root cause is something different.I think the problem is the following FROM line in my SIP INVITE transaction: From: MYNAME ;tag=773a3e6a-a677-4fb1-95fc-54b379b650a4 ..
I am using Asterisk 20.3.0 with PJSIP.I have setup a trunk to my ISP (Twilio) who requires outbound authentication.My pjsip.auth.conf contains: [Twilio] type=auth auth_type=userpass password=mysecret username=myun However, my calls using the trunk ..
I am creating a dialplan where a single user (Alice) has two offices.Both of her phones should ring if her extension is called. I could use a ring group, but Im wondering can both phones use the same PJSIP extension account (username/secret)?ThanksBr..
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..