Archives : June-2023
I have an AGI script written in PHP that worked great with Asterisk 13.Im porting it to an Asterisk 20 site and have a strange problem.I tried running the script from the command line and it works fine; I see the script commands written to stdout likeVERB..
I finally got to look at chan_sip to chan_pjsip migration again. This time I’m having problems with influencing codec selection on originating (calling) channel. It looks like PJSIP_MEDIA_OFFER only works on outbound (called) channel and has no aff..
Ive been having a serious issue the past couple weeks where many users devices show up as Unavailable according to PJSIP. The underlying issue is that res_pjsip thinks there are no available contacts for the device, and in the normal course of operati..
I am connecting to the ARI with subscribe all, so I can see channels being created.I now want to extract a variety of header variables (at the moment the from and to tag).I tried to read them from the ARI but Asterisk refuses since the channel is ..
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..