There is already a webrtc card, nothing needed anymore…
Sip card is also in place, it’s based on wss, and it will stay that way :+)
I know that. I just meant PBX. Which PBX do you use? I am just learning the PBX world. Never used one. I am trying the andrius/asterisk as a docker.
Any special configuration I should use?
I’m using the asterisk addon for HA, basic use…
I just forward the call to the android users ,because don’t want to use hikconnect…
I use Linhome as a sofphone client, but because there is no video in the stream, I inject it manually with an RTSP stream, just added a few lines of extra code in it
I am not sure I follow you. The Asterisk integration defines a need for a PBX server. On the other hand you say you do not run a stand alone PBX server but only the HA Asterisk integration and the add-on.
I had an impression you use Asterisk on a RPi.
Hi,
can anybody help me with the hikvision-v4l2loopback ?
I Can call Already the Sip Extension but canot Create the Dial Plan.
Dialplan example
exten => 777,1,NoOp()
same => n,Progress()
same => n,Wait(1)
same => n,Originate(PJSIP/1234,exten,default,777,1,,aC(ulaw,alaw,h264)c(1234)n(Doorbell))
same => n,Originate(PJSIP/6000,exten,default,888,1,,aC(ulaw,alaw,h264)c(6000)n(Fabio))
same => n,ConfBridge(1,myconferenceroom,admin_user)
exten => 777,1,NoOp()
same => n,Progress()
same => n,Wait(1)
same => n,ConfBridge(1,myconferenceroom,admin_user)
exten => 888,1,NoOp()
same => n,Progress()
same => n,Wait(1)
same => n,ConfBridge(1,myconferenceroom,marked_user)
but how i Configure it in OpenSips?
unfortunately i can find a tutorial on dialplan and opensips on the internet…
would be great if someone can help me
No, I don’t use a RPI , I dont like it…
Asterisk = PBX
Opensips is indeed not easy, but read a few pages up, it’s not needed anymorw, you can run a simple script now to a call to asterisk…
Why you using the loopback? What softphone are you using?
Also don’t use the loopback , it works , bit it’s very CPU hungry…
I use my own linhome or linphone client by injecting rtsp stream manually
I use the SIp client with the Gira G1 Door Station.
Since unfortunately out of the box the video stram does not run, I need the looper.
Ah ok , then indeed loopback is the option
I copy pasted the info wrong , I did 2 times 777 , offcourse that doesn’t work, just use another number in the extension
Edit: I edited the GitHub page
can you still tell me what to put in in wich field?
perfect would be a screenshot of the Dlplan page.
I edited the dialplan on GitHub?
The opensips info is also up-to-date, just link the incoming trunk to 777
Asterisk is not out of the box, you need to learn it… If it’s to difficult, then stick with hikconnect
I know that. Is it out of the box or does it needs configuration changes?
If yes than where can I find a tutorial?
Offcouse asterisk needs changes, I just post an example, you need to configure it for your needs… Create extensions and stuff…
unfortunately I still have not understood , sorry 
do I need anything else than the OpenSips server for the solution?
Where exactly do I create the dialplan ?
Depends, you don’t need asterisk, you can also register as a user on opensips…
I have Open sips running with a total of 3 numbers.
100 = DS-KD8003
200 = Gira G1
7000 = Video Looper
but how do I build the dialplan in Open Sips?
I don’t have knowledge of creating dialplan in opensips, I forwarded the call to asterisk, and created it there , for me much easier… The wiki pages of asterisk are much better
But I don’t use opensips anymore, just the python script to register… I created opensips proxy before ,but was an overkill… The script is easier
People using my v4l2loopback addon, dont use it , i created a new one
Much better, multi arch, runs on alpine, no ffmpeg anymore, so 0% cpu usage!
here is new one:
this addon is usefull for people that are registering on indoor panel with asterisk, the problem was that the outdoor doesnt send video then, this one injects the RTSP in the call when setting up an conference
enjoy
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