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Dahua VTO2000A, SIP Firmware and Asterisk

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You do not need sonia.zip and videodeamon.zip since this are the two main "programs" running on the VTO and are included in the firmware already (off course). No clue why dahua adds them as a zip file in the firmware archives separate.

You can get video via vlc and rtsp://vto_ip:554/

Dunno "MicroSIP", but green screen and dots sound like a video resolution issue to me. Does you client support h.264 video with 800x480px? Use jitsi on windows to check, what resolution is set in VTO for sip calls.

Did you check "CGI Enable" in "Security"-Tab?

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1 hour ago, riogrande75 said:

You do not need sonia.zip and videodeamon.zip since this are the two main "programs" running on the VTO and are included in the firmware already (off course). No clue why dahua adds them as a zip file in the firmware archives separate.

You can get video via vlc and rtsp://vto_ip:554/

Dunno "MicroSIP", but green screen and dots sound like a video resolution issue to me. Does you client support h.264 video with 800x480px? Use jitsi on windows to check, what resolution is set in VTO for sip calls.

Did you check "CGI Enable" in "Security"-Tab?

But sonia and videodaemon are much bigger than firmware. I guess they are not updated if flashing via tftp. Don't know if that's important.

Thanks for jitsi, it works! 

Yes, rtsp works ok

I can't find Security tab in my V4 interface. But as I said, it's not a problem for my setup

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Just a question ... the SIP firmware is it ofering all the same functionalities than the non SIP firmware as well, or I do loose "non SIP" functionalities if I use the SIP version ?

I am asking because I have the non SIP installed at current, and I was planning to install the SIP version ... will that hurt if I do not use the SIP ?

Many thanks !

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Hi

Inside the firmware zip file there are 3 bin files, for instance General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.zip has:

- General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.bin

- General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.language.bin

- Upall_VTOXXX.20181229.bin

After using the first one within VDPConfig app, the upgrade was successful but the VTO2000A webpage was strange (without any text as can be seen in the attached image). After many resets (including factory resets) the problem remained. I believe that was just when I also upgraded the second file (the language.bin) the webpage started to appear normal.

In a future upgrade I must always upgrade both files?

What about the third one (Upall_VTOXXX.20181229.bin)? What is its use?

Regards

new_ko_2.png

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Can you pls. post the download location of this firmware (  General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.zip)?

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10 hours ago, riogrande75 said:

Can you pls. post the download location of this firmware (  General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.zip)?

Hi riogrande75
I got it from https://share.dahuasecurity.com/ from folder: 04 Product Firmware > Intelligent Building > SIP firmware

By the way, I'm trying now to setup my VTO2000A with this firmware in order to link it to my RasPBX. Some of the VTO2000A webpages are a bit different compared with some of the screenshots I find here. Here are my actual settings that still do not work.

As you can see my test setup have a Windows MicroSIP client (extension 1000), an Android CSipSimple App (extension 2000) and VTO2000A (extension 8001 that I accepted as default and set as PJSIP without any certain). MicroSIP and CSipSimple already can connect with each other but not with the VTO2000A.

When I hit the call button of the VTO2000A, what extension it would call? The 9901?

Can I make a call from any of my other SIP clients to the VTO2000A? Calling 8001, right? When I do this I get an error (503 Service Unavailable) which is different of the one I get if I use a non existing extension, so it seems that the VTO2000A is somehow recognised.

Regards and thanks for any help you could give me.

 

Local Config.png

LAN Config.png

Network Config.png

FreePBX Extensions.png

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Hi!
Just installed the fw it on my test vto as well. At first, it looked like yours (empty parameter fields), but I solved it with clearing browser cache (as I was expecting it). So I did not flash the language file at all.

I suggest to read the manual, where all the parameters are described (well, at least most of 'em).

I don't know RasPBX as I youse asterisk in it's pure version (cli).

What I can say right after havin a quick look at the pics: IP and 8001 as registered number seem ok. If you press button on VTO a call to 9901 will be setup (1st pic).

I guess the problem is your SIP server. Is the ext. 8001 really configured correctly? I would deactivate auth just for testing purpous.

Also I would try to add VTO as CHAN_SIP and not PJSIP. I run all devices with chan_sip and everything is fine. In fact I did not play arround a lot with pjsip.

Additionally I would change the villa call number to fit your environment (e.g.1000/2000).

When you setup a SIP call to 8001 the vto should pick up the call automatically and you should have a audio/video connection if everything is setup correct.

If not: User asterisk debugging (sip set debug on) - that will point at the problem.

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@hsmptg So first of all. How do you get a login to that Dahua firmware site?!

Next: Blank web UI pages - you need to delete your browser cached data after a firmware update.

Villa Call Number: Yes this is the number dialled when you press the main call button in the VTO2000A. You will have to either create an extension or call group number in FreePBX to match this, or simply change this number to whatever you want to dial.

Local or Extense: Not entirely sure, but I think this referrers to either the in-built relay, or the "external" (is extense even a word?!) RS485 module.

Server Type: I always set mine to Asterisk for FreePBX, but actually on the most recent Firmware I have, Asterisk is not even an option, so I had to choose VTO. I don't think is matters, because perhaps it is only if you have "Server Enabled", which since we using this VTO as a SIP "CLIENT", then it's irrelevant.

As for the last username and password (and domain possibly), this is also only relevant for the device if it's acting as a SIP server, not as a client.

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6 minutes ago, riogrande75 said:

Hi!
Just installed the fw it on my test vto as well. At first, it looked like yours (empty parameter fields), but I solved it with clearing browser cache (as I was expecting it). So I did not flash the language file at all.

I suggest to read the manual, where all the parameters are described (well, at least most of 'em).

I don't know RasPBX as I youse asterisk in it's pure version (cli).

What I can say right after havin a quick look at the pics: IP and 8001 as registered number seem ok. If you press button on VTO a call to 9901 will be setup (1st pic).

I guess the problem is your SIP server. Is the ext. 8001 really configured correctly? I would deactivate auth just for testing purpous.

Also I would try to add VTO as CHAN_SIP and not PJSIP. I run all devices with chan_sip and everything is fine. In fact I did not play arround a lot with pjsip.

Additionally I would change the villa call number to fit your environment (e.g.1000/2000).

When you setup a SIP call to 8001 the vto should pick up the call automatically and you should have a audio/video connection if everything is setup correct.

If not: User asterisk debugging (sip set debug on) - that will point at the problem.

You beat me to the reply, but I would just add that I use PJSIP as it's now the default on RasPBX and everything works as I'd expect it to.

 

I wonder, why have they gone back to the V3,x branch if this is a newer build of firmware?! What's wrong with the "WEB SERVICE2.0" interface. I think it's much better, particularly the fact that you can see the video feed without using Internet Explorer!

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25 minutes ago, riogrande75 said:

Hi!
Just installed the fw it on my test vto as well. At first, it looked like yours (empty parameter fields), but I solved it with clearing browser cache (as I was expecting it). So I did not flash the language file at all.

I suggest to read the manual, where all the parameters are described (well, at least most of 'em).

I don't know RasPBX as I youse asterisk in it's pure version (cli).

What I can say right after havin a quick look at the pics: IP and 8001 as registered number seem ok. If you press button on VTO a call to 9901 will be setup (1st pic).

I guess the problem is your SIP server. Is the ext. 8001 really configured correctly? I would deactivate auth just for testing purpous.

Also I would try to add VTO as CHAN_SIP and not PJSIP. I run all devices with chan_sip and everything is fine. In fact I did not play arround a lot with pjsip.

Additionally I would change the villa call number to fit your environment (e.g.1000/2000).

When you setup a SIP call to 8001 the vto should pick up the call automatically and you should have a audio/video connection if everything is setup correct.

If not: User asterisk debugging (sip set debug on) - that will point at the problem.

Hi

I tried several things to solve the missing text on the webpages, and one of those things was in fact "clearing browser cache". So I did not had any certain about what had solve the problem. But since you and @rickwookie both point that issue, it seems that is really the way to manage it.

It is curious that you are just using Asterisk from cli since that was my initial idea. I would prefer just to install Asterisk in a clean Raspbian Stretch Lite image instead of having to use a RasPBX image. But for these initial tests it seems to be easier to use RasPBX.

My idea is to build with a Raspberry Pi plus a 7" display a SIP client equivalent to a VTH1550CH to connect to my VTO2000A. That way I could have in the same equipment a Video Intercom and my home automation (home assistant) terminal.

Regards

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3 minutes ago, hsmptg said:

Hi

I tried several things to solve the missing text on the webpages, and one of those things was in fact "clearing browser cache". So I did not had any certain about what had solve the problem. But since you and @rickwookie both point that issue, it seems that is really the way to manage it.

It is curious that you are just using Asterisk from cli since that was my initial idea. I would prefer just to install Asterisk in a clean Raspbian Stretch Lite image instead of having to use a RasPBX image. But for these initial tests it seems to be easier to use RasPBX.

My idea is to build with a Raspberry Pi plus a 7" display a SIP client equivalent to a VTH1550CH to connect to my VTO2000A. That way I could have in the same equipment a Video Intercom and my home automation (home assistant) terminal.

Regards

The only downside you using this in the SIP mode as opposed to the standard VTO-to-VTH mode is that it doesn't then notify the app (iDMSS Plus) when there's a call. Unless it does and I just haven't found a way to do it yet?

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33 minutes ago, rickwookie said:

You beat me to the reply, but I would just add that I use PJSIP as it's now the default on RasPBX and everything works as I'd expect it to.

 

I wonder, why have they gone back to the V3,x branch if this is a newer build of firmware?! What's wrong with the "WEB SERVICE2.0" interface. I think it's much better, particularly the fact that you can see the video feed without using Internet Explorer!

Hi rickwookie

Thanks for your suggestions!

Regarding the use of V4, I was a bit scared of flashing it since I read some posts reporting some bricking problems after doing that.

But since you have this opinion I thing I will try it. It is General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.zip, right?

But my initial question remains: What is the use of the other bin files besides General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin?

About my access to the Dahua firmware site, well I just used an username/password that I found in ipcamtalk forum.

Regards

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8 minutes ago, hsmptg said:

Hi rickwookie

Thanks for your suggestions!

Regarding the use of V4, I was a bit scared of flashing it since I read some posts reporting some bricking problems after doing that.

But since you have this opinion I thing I will try it. It is General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.zip, right?

But my initial question remains: What is the use of the other bin files besides General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin?

About my access to the Dahua firmware site, well I just used an username/password that I found in ipcamtalk forum.

Regards

Oh ok, yes I thought that chuest login had stopped working, but no its still does work!

General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin is what I'm using. The other bin files are just subsets of that main file (e.g. to only update the language if you have that same firmware in another language installed, I assume).

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Dunno. I guess, webservice 2.0 is and will be the future. But even it seems that they changed simple html code only, it's very buggy now.
Somehow they still release brand new versions in "old" web service 1.0 flavour.

Anyhow, don't use General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin as they removed asterisk support.
With that version installed I never got any successful connection setup to asterisk.

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Pay attention to this firmware, I think it is not official ... but there is an official one I will request my vendor to get.

 

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I just found the problem!

I was using copy/paste to copy the FreePBX SIP password to the VTO2000A web interface, and did not realised that it only accepts 15 characters when FreePBX automatically generates passwords with 32 characters. Since the VTO2000A web interface was truncating the password it was therefore incorrect! Using a shorter password, all started working correctly.

Next fight: the language used by the audio of the VTO2000A

I'm from Portugal and Portuguese is not supported. It seems that version 4 allow at least to turn off the sentences in the begin/end of a call. But since version 4 is not compatible with asterisk that is not a solution for me. Any other suggestion?

Regards

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Either you use 1st v4 firmware I posted 20180629, or get back to v3 fw. Btw. I have a pretty new v3 fw with portugese lang (20181229), but non SIP.

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On ‎1‎/‎28‎/‎2019 at 1:22 PM, riogrande75 said:

Anyhow, don't use General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin as they removed asterisk support.

With that version installed I never got any successful connection setup to asterisk.

That's interesting. It's true that Asterisk is not an option in the drop down menu in that firmware, but I just set mine to 'VTO' and it connects without any problem to my RasPBX box. That's why I've assumed that this setting is irrelevant to the VTO as a SIP client.

2019-01-30.png

2019-01-30 (1).png

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Well, it connects to my asterisk too (sip register is simple and does not have a lot of options). But there seems to be a problem when tryin to establish the connection. But maybe thats better with pjsip, my ast is still on chan_sip.

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After flashing General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin firmware I can still flash v3 firmware back if it does not work properly with RasPBX?

When you say "non SIP", you mean a firmware that just work with the VTH monitors and Dahua apps (gDMSS), right?

The General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin firmware is the one you suggest within version 4?

From what I read it still does not have Portuguese language (audio messages I mean) but at least I can disable its use, right?

Regards

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Yes, yes and no. I am suggesting General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.1.R.20180606 if you really "need" V4 on asterisk.

As mentioned, there is a V3 nonSIP firmware with portuguese audio messages available. In the latest english only V3 SIP fw (20181229) single audio messages cannot be disabled - i guess this a V4 "feature".

BTW and once again: You can always move from one to any other firmware with your VTO2000A. If the dahua tools do not support that, just use the tftp method. But you need to have either a lot of experience with embedded devices/networking/electronix or you must read the whole unbricking thread VERY carefully.

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18 minutes ago, riogrande75 said:

Yes, yes and no. I am suggesting General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.1.R.20180606 if you really "need" V4 on asterisk.

As mentioned, there is a V3 nonSIP firmware with portuguese audio messages available. In the latest english only V3 SIP fw (20181229) single audio messages cannot be disabled - i guess this a V4 "feature".

Any reason to use General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.1.R.20180606 and not the newest one?

Well, it seems that I "need" V4 in order to disable the english audio!

Even with a V3 with portuguese audio messages, it is nonSIP which would not allow the integration with RasPBX.

Anyhow, where could I get the "V3 nonSIP firmware with portuguese audio messages" to test?

Regards

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I uploaded it here: VTO+VTH multilingual firmware

I hope the "chinese-portuguese" is better than the "chinese-german"... If you don't like the sounds in your language, you either step back to an old SIP firmware as I did (where you are still able to personalize every sound file!) or you wait for a working V4 firmware, where you are at least able to deactivate the sounds.

Too bad, that dahua does not implement a "sound" upload option. I guess, this would not be hard to implement but it would make this little device the only one on the market with such options.

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