VTO2000A and SIP again...I can register my SIP Phone, my SIP phone rings when VTO calls but answering doesn't work

[Antori91]

Hello,

I know that this VTO2000A and SIP topic has been already covered but I didn't find a specific solution to the issue I'm having.

I've installed a few days ago the Dahua VTO/VTH (firmware v4.3) intercom solution and everything works fine even answering calls using my Android smartphone with the gDMSS lite App (yes, I receive the Google notifications when the VTO calls).

But to answer calls with my smartphone , I would like to use SIP rather the Dahua P2P. I'm able using Linphone to register my Android client to the VTO SIP server, even more when the VTO calls, my smartphone rings. But when I answer the call, nothing more happens and the Linphone from the green status (Connection OK) goes to orange status (Connecting).

Here is the Linphone configuration parameters I use:
User name : 9901#1
User authentication : Default VTO SIP password
Password : VTO password
Domain : VTO IP address
Protocol : UDP
Audio : PCMU, PCMA andt 6722 enabled
Video : H264 enabled
Call : Send DTMF's outside flux (SIP INFO)

I emphasize that :
- SIP registration is OK (I can see my smartphone connected to the VTO SIP server using the web interface of the VTO),
- my smartphone rings and the Linphone page to accept/decline the call appears,
- but if I try to accept the call, it doesn't work !

Do you have an idea? Thanks in advance

PS: I 'm able also to make ringing my smartphone when calling my smartphone from the VTH. But as previously described, I can't answer.

 

[Antori91]

I have progressed with the following SIP parameters. According to Dahua tutorial, User name and User authentication have to be the same and the password is the SIP user password and not the VTO password.
Now I can take the call. For example, when I call my smartphone from the VTH and answer the call, the VTH goes to "conversation in progress"

User name : 9901#1
User authentication : 9901#1
Password : SIP user password
Domain : VTO IP address
Protocol : UDP
Audio : PCMU, PCMA and G722 enabled
Video : H264 enabled
Call : Send DTMF's outside flux (SIP INFO)

 

[Justjason]

This is very interesting. I am happy to hear that you got this working. I have been trying to do the same thing and have not been able to get it working.
Could you I ask if you would share the config settings that you used on the VTO?

I have progressed with the following SIP parameters. According to Dahua tutorial, User name and User authentication have to be the same and the password is the SIP user password and not the VTO password.
Now I can take the call. For example, when I call my smartphone from the VTH and answer the call, the VTH goes to "conversation in progress"

User name : 9901#1
User authentication : 9901#1
Password : SIP user password
Domain : VTO IP address
Protocol : UDP
Audio : PCMU, PCMA and G722 enabled
Video : H264 enabled
Call : Send DTMF's outside flux (SIP INFO)

 

[Antori91]

Hello Justjason,
Here are my VTO main parameters:

• VTO firmware : 4.3
• SIP server : Enabled
• Device type : Villa station
• Villa Call No. : 9901 (VTH number)
• Group Call : Enabled
• VTO Rooms declared : from 9901#0 (VTH room) to 9901#9 (for mobile users)
• Video MainStream : WVGA 800 x 480 pixels, 25 images/s, bandwidth Max = 2 Mbps
• Video SubStream : CIF 352 x 240 pixels, 25 images/s, bandwidth Max = 256 Kbps

 

[Justjason]

Thanks, that is perfect !! Can i also ask if you are having any issues using the unlock buttons using FW 4.3 ?
I saw this mentioned in another post while looking for info

Dahua VTO2000A, SIP Firmware and Asterisk

hello, I updated a vto 1210x with firmware v.4, now I would like to downgrade, with ipconfigtool, by logging on port 3800 tells me login failed, by logging on 37777 tells me system in busy. Help. Thank you.

www.cctvforum.com

 

[Antori91]

Thanks, that is perfect !! Can i also ask if you are having any issues using the unlock buttons using FW 4.3 ?
I saw this mentioned in another post while looking for info

Dahua VTO2000A, SIP Firmware and Asterisk

hello, I updated a vto 1210x with firmware v.4, now I would like to downgrade, with ipconfigtool, by logging on port 3800 tells me login failed, by logging on 37777 tells me system in busy. Help. Thank you.

www.cctvforum.com

I didn't notice any issue.

 

[_lucas_]

Hello Justjason,
Here are my VTO main parameters:

• VTO firmware : 4.3
• SIP server : Enabled
• Device type : Villa station
• Villa Call No. : 9901 (VTH number)
• Group Call : Enabled
• VTO Rooms declared : from 9901#0 (VTH room) to 9901#9 (for mobile users)
• Video MainStream : WVGA 800 x 480 pixels, 25 images/s, bandwidth Max = 2 Mbps
• Video SubStream : CIF 352 x 240 pixels, 25 images/s, bandwidth Max = 256 Kbps

hello mate, I have vto / vth in firm 4.3 and they connect me both by chan_sip and with pjsip to asterisk.
but neither linphone nor voip by Antisip works for me in chan_sip.
can you tell me how you do this
I am testing with chan_sip in vto / vth and phone in pjsip, something always fails and I don't know if I'm in the right direction

 

[Antori91]

Hello Lucas,
I'm sorry but currently I don't use Asterisk in my Dahua environment. I don't know how to help you. Maybe someone else.

 

[runraid]

Here's how I got linphone to work on my PC...



When it request the username/password, I just put:
Username: 9901#0
Password: 123456 (which is the default)

Note in the SIP Address above: sip:9901%230@192.168.1.210 the %23 is a URL encoded #.

 

[Justjason]

Here's how I got linphone to work on my PC...

View attachment 84632

When it request the username/password, I just put:
Username: 9901#0
Password: 123456 (which is the default)

Note in the SIP Address above: sip:9901%230@192.168.1.210 the %23 is a URL encoded #.

Perfect, thanks

 

[Motionaire]

It always says Connection Failed at the top of my app. Can someone help me please how to set this up perfectly?

 

[Mauro70]

Hello Justjason,
Here are my VTO main parameters:

• VTO firmware : 4.3
• SIP server : Enabled
• Device type : Villa station
• Villa Call No. : 9901 (VTH number)
• Group Call : Enabled
• VTO Rooms declared : from 9901#0 (VTH room) to 9901#9 (for mobile users)
• Video MainStream : WVGA 800 x 480 pixels, 25 images/s, bandwidth Max = 2 Mbps
• Video SubStream : CIF 352 x 240 pixels, 25 images/s, bandwidth Max = 256 Kbps

Hi Antori91, hope you are still around, just to better understand: with this config, were you able to have a group-call with the VTO, VTH and mobile device via SIP (embedded into dahua VTO) without P2P? and is it group-video early preview working on the mobile device and VTH at the same time during the ringing?
Thanks

 

[Mash203]

Hi!

I didn't purchase a VTH, and would like to use an old android phone as door chime, using the VTO2211G-WP as SIP server.
The notification from the DMSS app is not enough to do proper ringing, so I'd prefer to use SIP with a client that offers some more customisation.
I'm not ready yet to put in the effort to go with Asterisk, so I'd like to try Linphone first. (or another client if that works as well)

Currently, I'm struggling to get Linphone setup on my android device. It's unclear to me which SIP user password/user name/server I should set.

What makes sense, is that I just use the SIP server IP (192.168.1.111 in my case) as server, but then I don't get a connection. (see screenshot attached of web interface)
If I use the VTO web interface IP of 192.168.1.108, I'm getting a connection in linphone, but nothing happens when I start a call using the call button on the VTO.

What would also make sense, is that I choose one of the room numbers (I added one with name 9902, to get rid of the #), and use that as username, and default password.
Though as you can see in the screenshot, there's another username mentioned in the web interface.

There's also another SIP Server username & password mentioned.

I'm very confused as to what I should use? I assume I shouldn't use the server username, nor the other username mentioned there, but need to pretend I'm one of the rooms?
Though I don't get a connection if I use 111 as an IP adress.


Is anyone brave enough to advise?

 

[hansmagnus]

What makes sense, is that I just use the SIP server IP (192.168.1.111 in my case) as server, but then I don't get a connection. (see screenshot attached of web interface)

These are the settings if you do not use VTO as SIP server. You can confugarate an external SIP server.
If you enable SIP Server, the server is running on your VTO with same IP adress as the web interface.

If I use the VTO web interface IP of 192.168.1.108, I'm getting a connection in linphone, but nothing happens when I start a call using the call button on the VTO.

I have the same problem. Connetion with linphone works fine. I can also call the VTO (8001) but without response.
And linphone does not ring when I press the call button.

Call Logs look like this:




Edit:
VTO2202F, Firmware 4.511.