I started messing with this recently and got things working between my Home Assistant users, but have some questions.
I also run go2rtc and frigate. My doorbell does not have SIP support by itself so I don’t think I can follow the normal doorbell guides here because I have no way to tell the doorbell to register and connect. Is it possible to create the extension for it and somehow have the extension connect to the webrtc feed created from go2rtc for two way audio? I have two way audio working directly with go2rtc so that part already works as expected. I know I can set the camera in the sip card, but I’m unclear if that actually ends up bridging my audio from the caller over to the webrtc feed.
Basically what I’m trying to accomplish is to SIP enable all my cameras that don’t support it natively.
It probably won’t take much to add those two #defines to the app_rtsp_sip.c file to the docker build process with a dependency on the arch. The one person who said the fix worked for them was using an rpi4 model B which I believe is an aarch64, so should work for that arch. I’m just not sure about armhf and armv7.
[EDIT] A fix has been put in place for ARM based builds starting 4.2.0 (see post below)
Seems like you put a incorrect person entity in the config. You need to set a actual person entity id in the card config.
As for the RasPBX, I’m not sure if that will work. The current setup does need a valid SSL on your Asterisk server for it to work. If you run HA-OS you can try the asterisk add-on, which has some settings pre-configured for the sip card.
There is a new card coming, that does not require SSL or port forwarding, and works with every PBX. It’s still a WIP and the first proof of concept got released for testing. But it might we worth waiting for. You can follow progress in the discord server.
Thanks! So if I put an incorrect person entity in, NOTHING shows up? It is just totally blank? In that case sounds like I have the lovelace bit right it just has no way of handling that error?
I wonder if I used the SIP-Hass Addon whether I could somehow call using IAX between the two asterisk instances…quite keen to still have FreePBX but could maybe bridge connections? Hmmmm
Thanks, in the end I found (for my use case) this addon (DSS Notifier for Arm) does exactly what I need. It’s a liiiiiitle beta-y but it’s great when it works!
using an app like SIPnetic, what should my network settings be to work with Asterisk? my HAOS machine is not accessible via HTTPS, so i disabled TLS as seen here. I believe the port is 5160.
still the android app on tablet 1 (ext 101) cannot call tablet 2 (ext 102)
Hi,
I’ve a Dahua doorbell VTO2211 at 192.168.1.8. It has internal SIP server enabled on port 5060.
I want to use my wall tablet with HA to answer to the doorbell, see the video and unlock the gate.
I only want a SIP client card to do this.
Can you help me? Can you also tell me the correct values to insert in the configuration?
Hi,
Asterisk is having trouble setting up the SSL when using the Nabu Casa Cloud. Can anyone give me some hints how to setup?
I wanted to switch from open ports to the cloud solution because of security reasons. The connection worked perfectly when using duckdns with an open Port.
I generate wildcard ssl from nginix proxy manager, when I try to use this generated ssl I get error “FATAL: Certificate file at /addon_configs/a0d7b954_nginxproxymanager/letsencrypt/live/npm-15/fullchain.pem was not found”, it is the soft link from this path /addon_configs/a0d7b954_nginxproxymanager/letsencrypt/archive/npm-15/