Dahua VTO2000A, SIP Firmware and Asterisk

[gerdshi 0]

And here some Docu to the links that Foxi found in sonia:

 

ftp://ftp.asm.cz/Dahua/videovratni/_SDK/DAHUA_HTTP_CGI_FOR_VDP.pdf

 

Its only a small number of Links with a little bit more information.

[gerdshi 0]

There is a new (for me) SIP-Firmware for the VTH indoor Station:ftp://ftp.asm.cz/Dahua/videovratni/VTH15xx/firmware/20160930-CZ-SIP.ZIP

[galaksy2001 0]

Hi everybody,

 

I'm a happy owner of the vto2000a. I found this post very interesting : thanks for the great job !

I try to unlock the door with the command i've found on this post, then with the other one found in the API provided in the link via ftp (ftp://ftp.asm.cz/Dahua/ ....pdf) but both don't work :

http://xxx.xxx.x.xxx:yyyy/cgi-bin/accessControl.cgi?action=openDoor

http://xxx.xxx.x.xxx:yyyy/cgi-bin/accessControl.cgi?action=openDoor&channel=x&UserId=xxx&Type=Remote

 

Do you have the same issue ?

 

I don't understand what 's the meaning of "channel" and i suppose that UserId is the SID returned by the following command which don't work correctly for me :

 

http://192.168.0.150:1025/cgi-bin/VideoTalkPeer.cgi?action=attachState

 

Which returns no SID but strings like json :

 

[

{

"id" : x,

"result" : x,

"session" : x

},

{

"id" : x,

"params" : {

"CallID" : "x"

},

"result" : true,

"session" : x

}

]

 

Thanks for your answer and sorry if i'm out of the main subject

And thanks again for the work ! " title="Applause" /> " title="Applause" /> " title="Applause" />

[gerdshi 0]

Hello!

 

Do you tryed the Example:

http://192.168.xxx.xxx/cgi-bin/accessControl.cgi?action=openDoor&channel=1&UserID=101&Type=Remote

It works fine by me.

 

In the PDF i read:

URL-Syntax: http://<ip>/cgi-bin/accessControl.cgi?action=openDoor&channel=<channelNo>[&UserID=<UserID>&Type=<Type>]

Param in: channelNo: channelNo: the index of door. Start from 1;

The followings are optional：
UserID: remote User ID;
Type: the open type; default value is “Remote”.

For example： http://<ip>/cgi-bin/accessControl.cgi?action=openDoor&channel=1&UserID=101&Type=Remote

 

OK, why 101 - i don't know.

I assumed it must be a SIP-DID, or the own 8001 for example... But no.

May be 101 is the Linux-UID from admin?

 

Edit: as User ID works 102 too and 103 and ....

[galaksy2001 0]

Hello!

 

Do you tryed the Example:

http://192.168.xxx.xxx/cgi-bin/accessControl.cgi?action=openDoor&channel=1&UserID=101&Type=Remote

It works fine by me.

 

Edit: as User ID works 102 too and 103 and ....

 

Thanks gerdshi, it works for me too now Strange 'cause i'm sure i've execetuted the same command before ... maybe a syntaxe error

 

Same for me : as User ID works 102 too and 103 and .... 106 ....

 

And what about the SID, did you managed to get it with the following command

 

http://<ip>/cgi-bin/VideoTalkPeer.cgi?action=attachState

[gerdshi 0]

Hi!

 

I'm not sure for what it is good?

In the API PDF stays written:

Subscribe the video talk status.

When the client sent request is offline, auto cancel subscription.

I'm from germany and here exist a Term - Ich verstehe nur Bahnhof (I understand only Station)

Meaning - i don't understand nothing.

 

I don't understand the Context of the API Call....

 

It is may be for the nonSIP-Firmware. Were a Client must register at the VTO?!

The example IP-Adress is a Mutlicast-Addres - such thing doesn't exist usualy by SIP. So i think this is all for the nonSIP-Firmware. But i'm not 100% sure.

I'm sorry it is to high for me

[galaksy2001 0]

Thanks again for your answers gerdshi

 

My firmware is a non-sip one which should accept the command to get the SID ...

 

I'm from france and "Ich verstehe auch nur Bahnhof"

 

Tchüss

[Allodo 0]

After a long time for wait I get my new VTO

 

I've tested some achievements from here and it seems to work well.

 

But I still get no Video on my Android App (Zoiper, Linphone or Bria) and didn't get a call on 3G, only on WiFi.

These two things I'm still missing for turning my VTO to productive.

 

Can anyone help me to solve the problems?

 

I would like to use myfritz.net for DynDns.

[Foxi352 0]

After a long time for wait I get my new VTO

 

I've tested some achievements from here and it seems to work well.

 

But I still get no Video on my Android App (Zoiper, Linphone or Bria) and didn't get a call on 3G, only on WiFi.

These two things I'm still missing for turning my VTO to productive.

 

Can anyone help me to solve the problems?

 

I would like to use myfritz.net for DynDns.

 

I have video running out of the box with linphone, that should work without problems if config is right. To get a call on 3G you need to use TCP and not UDP because linphone need to keep up a connection (stay registered) with your asterisk @ home. That leads us to the next problem: On Fritz!Box you can't redirect port 5060 to an internal computer. The Fritz!Box wants to keep that port for it's own VOIP usage. The solution was to run asterisk on port 5061 and redirect that port to your asterisk.

 

Here is the general section of my sip.conf that works with linphone:

[general]
videosupport=yes
language=de
bindport=5061
realm=<my_realm_name>
externhost=<my_external_dyndns_hostname>
externrefresh=10
context=default
type=friend
externrefresh=30
nat=force_rport,comedia
srvlookup=yes
tcpenable=yes
transport=udp,tcp
localnet=192.168.0.0/255.255.0.0
stunaddr=<my_providers_stun_server>
directmedia=no
srvlookup=yes
allowguest=no
alwaysauthreject=yes

Here is the linphone relevant part:

[linphone]
type=friend
host=dynamic
secret=<mypassword>
context=users
disallow=all
allow=ulaw
allow=h264
nat=force_rport,comedia

On my linphone under proxy i added ":5061" and under Transport i chose "TCP".

[vytasm 0]

Hello. Nice work done! I will try to translate and change voice records later also.. But now have problem finding right SIP firmware for my VTH1560 console. I have found one that works quite well but it is in Czech language only it is here ( ftp://ftp.asm.cz/Dahua/videovratni/VTH15xx/firmware/SIP_20161017-CZ.ZIP ) May be some body have and can share English version? I have tried older version (General_VTH1510_Eng_P_SIP_V1.100.00.0.R.20151120) but it freezes on call: when I press "call" it rings and shows video and buttons to answer the call but when I touch the buttons nothing happens... The Czech version works perfect, but I do not understand it

[Allodo 0]

Well I've not tested the latest Firmware but the older Firmware from this page (which I've used) always are in English and not CZ. Maybe there is a possibilty to change to English in Settings.

 

There are two kinds of firmware, the international one (which has no SIP). There you can choose language and the other one is with SIP and always was in English. Maybe tomorrow I will find time to test the newer Firmware

[vytasm 0]

Allodo said:

Well I've not tested the latest Firmware but the older Firmware from this page (which I've used) always are in English and not CZ. Maybe there is a possibilty to change to English in Settings.

 

There are two kinds of firmware, the international one (which has no SIP). There you can choose language and the other one is with SIP and always was in English. Maybe tomorrow I will find time to test the newer Firmware

 

Yes I know that. But I am talking about not the camera itself, but about the screen panel VTH1560B. I have older firmware (General_VTH1510_Eng_P_SIP_V1.100.00.0.R.20151120.bin) and it is in English. but for some reason when You receive a call the panel screen freezes for some time and I can not pick the call or unlock the door. On the VTO200A I am using General_VTO2000A_Eng_P_16M_SIP_V1.200.1000.0.R.20160505.bin

 

I have found a newer firmware in the link I mentioned before (ftp://ftp.asm.cz/Dahua/videovratni/VTH15xx/firmware/SIP_20161017-CZ.ZIP) it is working and panel does not freeze but that firmware is in Czech only So few days ago following this instruction:

 

I have extracted the Czech firmware and replaced language files and then I packed firmware again and updated panel... Wualia - now I have English language and it is working. I think it is possible to use same method and change language or voice files on VTO2000A to.

SIP protocol works just fine with Asterisk phone system. Just there are few annoying things. First, when call is requested on VTO there is no call signal like in normal phone so the user just hears women voice "Calling now, please wait" and then silence... it would be great to hear some ring signal. On the inside panel screen there should be possible to press Unlock button and the call should be answered, door unlocked and call canceled. Now if I unlock the door and cancel call without answering then other extensions are still ringing. This should be implemented by the person or team who is programming the GUI, but there is no contacts or support for that...

[kameraman 0]

Since my Czech is rusty, looks like I will have to settle for v1.2. After extensive search, it appears that binary is no longer available on the original site. Are there any working links to General_VTO2000A_Eng_P_16M_SIP_V1.200.1000.0.R.20160505.bin still available?

 

Thanks!

[gerdshi 0]

Hello Vytasm!

 

I have found a newer firmware in the link I mentioned before (ftp://ftp.asm.cz/Dahua/videovratni/VTH15xx/firmware/SIP_20161017-CZ.ZIP) it is working and panel does not freeze but that firmware is in Czech only So few days ago following this instruction viewtopic.php?p=293981#p293981 I have extracted the Czech firmware and replaced language files and then I packed firmware again and updated panel... Wualia - now I have English language and it is working. I think it is possible to use same method and change language or voice files on VTO2000A to.

Can you help me please - this instruvtions shows how to change the Language files on the VTO2000. I tryed like you to make it for the VTH15xx but i don't what a destination should set at step 8. Becaus by the VTH i don't become the Message "WARNING Autodetected config: VTO"

Or maybe you can upload you ready Firmware?

 

SIP protocol works just fine with Asterisk phone system. Just there are few annoying things. First, when call is requested on VTO there is no call signal like in normal phone so the user just hears women voice "Calling now, please wait" and then silence... it would be great to hear some ring signal.

There are two-three ways to solve this:

- you set at you Dial() command in asterisks dialplan the option r or R or together.

- look at Progress(), the progressinband setting in sip.conf or Ringing() if you would like to avoid the use of 'r' but have issues with the ringback behaviour of Dial().

- make in you diaplan as first step an answer and play after that ring tones. But this is ...

http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial

 

On the inside panel screen there should be possible to press Unlock button and the call should be answered, door unlocked and call canceled. Now if I unlock the door and cancel call without answering then other extensions are still ringing. This should be implemented by the person or team who is programming the GUI, but there is no contacts or support for that...

I'm sorry, but with the SIP-Protocol you must do this in you Asterisk self. There is no way at the VTO or other dialing SIP-Device to discover this situation - it is simply impossible. The only who can send a message to the VTO back when ringing is canceled, is the Asterisk self. The Panel (SIP-Client) communicates in the Ringing phase only with the Asterisk and with no body else. In this moment the VTO doesn't know that there is a Panel or more then one Clients.... He "see" only the Asterisk, nothing else. The whole Communication is going over the Asterisk.

So you can modify you dialapln in such way, that when in Asterisk occure a disconnect event, the whole call to VTO is canceled.

 

https://de.wikipedia.org/wiki/Session_Initiation_Protocol#/media/File:SIP-B2BUA-call-flow.png

Look that there is no direct connection between Alice and Boris. The only Connection is the SIP-PBX.

 

You have (may be!) direct connection between VTO and VTH ONLY if the Client is going to answer the call. And then ONLY when the RTP-Media Data a redirected between the VTO and the VTH. That is in sip-calls not allways the Case! The option in asterisk for this is canreinvite. If it is set to NO on one or both of the sides - the stream will going only true the asterisk. This is why i wrote may be.

 

Best Regards and excuse me for my bad english!

[unschuldsjuenger 0]

I tryed very long to get on my VTO2000A the SIP Firmware. When i Flashed it, them web interface will start, but no configuration will be possible.

Only flashing back ti V3.1 without SIP will work.

Any Ideas, I just what to use VTO2000A only with my Fritzbox, but I cannot configure it as shown below by Allodo .

I donnot have in A& C Manager "Open Door Command" and Villa Call Comand. Where can i get the Firmware?

Thanks al lot

[gerdshi 0]

Do you did make a reset over ssh? I had too no luck without reset, but with reset it works good.

[jvbts8 0]

Hi,

 

I own a VTO2000A-2 (2-wire version). Is it possible to install SIP-firmware, and if yes, which one?

[kameraman 0]

What are you guys using for custom audio prompts? Is there a repacked firmware anyone can share?

[Allodo 0]

@ unschuldsjuenger

You must reset the VTO by SSH, try another browser or empty your browsercache.

The last two things sometimes works, but when you reset the VTO by SSH I had always success

 

For Reset by SSH you have to use putty oder WinSCP under Windows.

Putty is an standalone program which is only an executable File (nothing to install).

 

Before you can use putty you have to allow Telnet on VTO.

 

Just type this in your browser:

http:///cgi-bin/configManager.cgi?action=setConfig&Telnet.Enable=true

e.g.

http://192.168.1.110/cgi-bin/configManager.cgi?action=setConfig&Telnet.Enable=true

 

After that telnet-Session is enabled and you can use putty to connect to the VTO

 

Login user: admin

Password: 7ujMko0admin

If your password is an other then type as password: 7ujMko0 + your password

 

Then you type the following on console:

rm -rf /mnt/mtd/*

rm -rf /mnt/backup/*

reboot

 

The VTO will reboot an all settings are resetted

[unschuldsjuenger 0]

Hello,

Im still working to get the SIP Version on my VTO. Flash is fine now , but when I want to do the configuration, it seems that all information are empty, no Local Information, no Devicetyp. When i Flash back all Information are avaiable.

 

I only have an VTO2000A (should) running with a Fritzbox.

Any idears?

[unschuldsjuenger 0]

DONE,

USE LAN not WLAN, and delete CASH.

Still not working with my Fritzbox.

I only wont to get the RING Button on my Fritzphones, but SIP will not work.

[mgerste 0]

Hi!

 

Is it possible to get an mjpeg stream from a VTO2000A?

It would be very important for me.

[Grumpy 0]

Hi everybody,

 

I very much appreciate the knowledge you allready shared regarding SIP on VTO2000a.

 

I got my one today and tried to load the SIP-Firmware:

General_VTO2000A_Eng_P_16M_SIP_V1.200.1000.0.R.20160505.bin

 

Currently there is this firmware installed:

General_Multi3_VTO2000A_EngRusItlFreGetDutSpaPor_P_16M_V3.100.0000.0.R.20160622.bin

 

When starting upgrade with ConfigTool progress counts up to 50% and then I get the status "Upgrade succeed" and the message "Device upgrading is complete."

 

ConfigTool has version 4.00.0

 

Unfortunately the new firmware is not loaded. Stil the old one is there.

Here I found a workaround for "older" firmware not installable:

http://www.cctvstoreuk.com/blogs/cctv-faqs/132564995-solved-dahua-vto2000a-no-sip-or-only-with-vth150ch-vth1550ch

 

But the VTO2000a does not keep any other port than 37777 after reboot.

 

So I'm stucked.

I have gone through all posts but couldn't find additional comments or help for this problem.

In case I missed it - I apoligize for that.

 

Anyone who can help or point to the right direction ?

 

Many thanks.

[Allodo 0]

@Grumpy

You have to use Port 3800 in Config Tool for flashing Firmware.

 

Interesting Link, because this is my post, but I don't know this Website.

This is the Original-Post: http://www.cctvforum.com/viewtopic.php?f=6&t=51296&hilit=VTO

Edited January 12, 2017 by Guest

[Allodo 0]

@unschuldsjuenger

Till now you must use Asterisk if you wanna use VTO with Fritz!Fon.

 

That's an Problem about communication between Fritzbox and the VTO.

The VTO sends an INVITE to the Fritzbox. The Fritzbox answers with 183 SESSION PROGRESS, but the VTO don't know this answer. The only answer the VTO understands is 180 RINGING. Because of this the VTO Ends the Call.

 

I've informed the Dahua-Support about this behaviour. There was no answer about it

 

Maybe everyone with this Problem should contact Dahua-Support. So they see that a lot of People wanna use this combination and they could include it in an SIP-Firmware.

 

If the VTO would understand 183 Session Progress, an FritzFon could be used directly. But therefor the Dahua-Support has to build it in Firmware.