FreePBX with Dahua VTO + 2x VTH

[pedromrg]

Hi there everyone,

I have been trying to setup FreePBX with my VTO and both VTH's still I reach a block.

Calling VTO--->VTH is OK
Calling VTH--->VTO is OK
I can see video on VTH but if I open a call with VTO the video is black.

I'm not an expert in asterix / freePBX, so anyone have any idea on how to troublehsoot this?

thanks

 

[riogrande75]

Set it up with the vto Internal sip server and let freepbx register on that sip server too.

 

[pedromrg]

Set it up with the vto Internal sip server and let freepbx register on that sip server too.

Thank you for the reply riogrande.

Never thought that could work.

How can I register freepbx on VTO Server SIP trunk?

Then you mean register both screens on VTO SIP server and home SIP phone and mobile phones on FreePBX ?

Also crossed my mind installing Asterisk v16 (seems to be the one that is working fine with VTO--VTH)

Thank you

 

[riogrande75]

Simply setup VTO+VTH according to manual/videos and enable group_call.
Add a extra number (e.gh.9901#2) and let Asterisk/FreePBX register a trunk on that number.
Then you'll get all (group-)calls on the PBX side as well and can forward, etc. it.

Video for clients on the PBX side works either with rtsp from the phone directly (e.g. Grandstream and Unify CP phones) or with lots of fine adjustments on PBX sip video part.

 

[pedromrg]

Simply setup VTO+VTH according to manual/videos and enable group_call.
Add a extra number (e.gh.9901#2) and let Asterisk/FreePBX register a trunk on that number.
Then you'll get all (group-)calls on the PBX side as well and can forward, etc. it.

Video for clients on the PBX side works either with rtsp from the phone directly (e.g. Grandstream and Unify CP phones) or with lots of fine adjustments on PBX sip video part.

Good idea, I'm going to try this one later and report back.

Thanks

 

[pedromrg]

Simply setup VTO+VTH according to manual/videos and enable group_call.
Add a extra number (e.gh.9901#2) and let Asterisk/FreePBX register a trunk on that number.
Then you'll get all (group-)calls on the PBX side as well and can forward, etc. it.

Video for clients on the PBX side works either with rtsp from the phone directly (e.g. Grandstream and Unify CP phones) or with lots of fine adjustments on PBX sip video part.

Ok, got the Trunk up to 9901#2

Now created an inbound route



Do I need to take anything esle in consideration?

Can I call from a SIP phone to any VTH?

 

[riogrande75]

You sould get a INVITE to 9901#2 from VTO - what you do with this in your PBX is up to u.

If you route calls to the VTH via your SIP trunk (9901#2), then you should be able to address every single one.

 

[pedromrg]

You sould get a INVITE to 9901#2 from VTO - what you do with this in your PBX is up to u.

If you route calls to the VTH via your SIP trunk (9901#2), then you should be able to address every single one.

So to resume what I have now:

VTO - acting as SIP SERVER 8001
VTH registering to VTO SIP SERVER
FreePBX trunk on VTO 9901#2

I can now receive calls from the VTO to all the VTHs (video OK)
To Linphone/Zoiper sounds is OK but... Video Not ok, still trying to figure this one out.
-----------
Call from Linphone/Zoiper to VTH/VTO not working, need to figure out the routing.

Thanks for the help so far

 

[riogrande75]

I configured my asterisk (chan_sip) this way:
sip.conf:

register => 9901#2:password@vto.ip/SIP_VTO

extensions.conf:

exten => 8001,1,Set(CALLERID(all)="Asterik2VTO" <9901#2>); VTO productive
same => n,Dial(SIP/SIP_VTO/${EXTEN})

exten => 9900,1,Set(CALLERID(all)="Asterik2VTH" <9901#2>); VTH kitchen
same => n,Dial(SIP/SIP_VTO/9901#0)

Dialing 8001 on my sip-client (jitsi) reaches my VTO, 9900 my VTH.

 

[pedromrg]

I configured my asterisk (chan_sip) this way:
sip.conf:

register => 9901#2:password@vto.ip/SIP_VTO

extensions.conf:

exten => 8001,1,Set(CALLERID(all)="Asterik2VTO" <9901#2>); VTO productive
same => n,Dial(SIP/SIP_VTO/${EXTEN})

exten => 9900,1,Set(CALLERID(all)="Asterik2VTH" <9901#2>); VTH kitchen
same => n,Dial(SIP/SIP_VTO/9901#0)

Dialing 8001 on my sip-client (jitsi) reaches my VTO, 9900 my VTH.

Thank you again.

I have this registered on VTO.... now I need to figure out how to forward 8001 to SIP_VTO in FreePBX.

I will try and make a test tonight.

 

[pedromrg]

ok bingo. I can now call VTO and VTH

Missing the video, but a good thing is that I can now turn video ON linphone and see some data, but no video up.

Trunk configs, can you check and see if you spot anything ?





linphone:





Thank you

 

[riogrande75]

Regarding video from VTO on 3rd pty devices: Have a look at this thread, here we discussed this issue too.
I get VTO video on my devices via rtsp (my sip phone is able to connect to a rtsp source) so I don't need sip video for now.

Anyway, if you spend some time on that, you might be lucky to get video either via sip or the multicast address I noticed (224.0.2.14:3001).

 

[Frank89]

Hi @riogrande75 ,
you are my last hope. I have read this thread and what you mentioned in the post just above. But I have not been able to find a suitable solution for my system.
I have a pbx 3cx and a dahua vto 3211d with fw 4.4 . The vto is registered as an extension on the 3cx and the 4 buttons on the vto call 4 different call groups on the 3cx. Now I would like to add the VTH but I can't do this by putting them as extensions on the 3cx because when I set them up like this the call comes to the VTH but it can't be answered, nor does it seem to be able to see the VTO video. I don't point to the available antemps of the vto on 3cx or third party apps. It is sufficient for me to have the audio of the Vto on the various ip phones and smartphones where I have the 3cx app installed. And this works well for me so far. But at this point I can't add VTH. I don't know if I should set up the VTO as a sip server, at which point the VTH would work fine but I don't know how to get the VTO, if it also acts as a sip server, registered on the 3cx PBX (I've tried setting it up as a gateway or as a sip trunk, but to no avail). How can I do this?

 

[riogrande75]

Just read my posts above in this thread. Setup VTO+VTH with vto sip server like it's described in manuals/video of dauha.
Add an extra extension for your pbx on VTO's sip server and activate group_call.
Then let your freepbx,etc. register on this server (add a sip trunk from pbx to vto, just like any other trunk).

If this was completed successfully you should receive a INVITE message in your pbx every time someone rings you doorbell and you can then add any further actions with this.

 

[Frank89]

I have tried but 3cx doesn't accept hastagh in the number of trunk sip. Also if it write that hastagh is accepted: "Only the following characters are permitted A-Z a-z 0-9 + #". But for the VTO is necessary to put #1 #2 etc. How could resolve? The problem is 3cx

 

[riogrande75]

I've no idea as group_call on vto's works with sip_number+hashtag+device_number only, as far as I understand.
So this question doesn't fit here but in a 3cx forum.