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Dahua VTO2000A, SIP Firmware and Asterisk

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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.

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hello naponet,

I also own a vto1210c-x and like to know why do you want to downgrade? Are there features missing, functions not working, language not good,... ?

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Hello, the firmware consists of two parts, the second language, I could not even flash that ...

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I finally had time to test version 4 of the firmware trying not only to confirm the possibility of disable the audio messages but also to solve an issue that I have in version 3.

I can confirm that in version 4 we have the possibility of disable the audio messages. Although I could live with the Chinese accent of the used English, the other users of my VTO (mailman, visits, etc.) would be confused with audio messages in another language besides the Portuguese. So it is better to remove them at all! This is possible in version 4 which I appreciate.

But in version 4 I lost the possibility of opening the lock using command #123#. I see that option in the menus of the web interface but I never was able to work with it as I did In version 3. Fortunately, I have other means of opening the lock besides using DTMF. 

Another strange thing in version 4 is when I call VTO (extension 8001) from another phone the call started but without sound. Call from the VTO, hitting its button, are received in extension 9901 with sound. I only managed to solve that configuring extension 8001 as chan_sip instead of pjsip. In version 3, with pjsip I have sound in either direction.

Finally the issue that I was hoping to solve when I moved to version 4 but it remains as in version 3: when a make a video call TO VTO (extension 8001) I am able to get sound and the video of the VTO camera (second inserted image), but when I accept a call FROM the VTO (initiated hitting the button of the VTO) I can get the audio but not video, instead I see a strange pattern like the one seen in the first inserted image. Note that since I can see the correct video when I call the VTO it seems that is not a codec problem, right? 

Any suggestions?

video_FROM_vto.png

video_TO_vto.png

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Regarding video: It is for sure a codec/resolution issue. Had the same with my sip phone initially. Dunno MicroSIP, but is there a way to check actual used codec like jitsi does?

JitsiVTO.jpg.af8f9c5ddf3c939b2ee115bfaaedc329.jpg

Try to lower main resolution to lowest (CIF/QCIF) and give it a shot. Unfortunately the resolution set in main gets used for sip calls too, better would be extra stream.

 

 

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1 hour ago, riogrande75 said:

Regarding video: It is for sure a codec/resolution issue. Had the same with my sip phone initially. Dunno MicroSIP, but is there a way to check actual used codec like jitsi does?

Try to lower main resolution to lowest (CIF/QCIF) and give it a shot. Unfortunately the resolution set in main gets used for sip calls too, better would be extra stream.

 

 

Thanks riogrande75 to put me again back in track!

Using jitsi instead of microsip I solve the video problem. Now I can also see the video when the call is initiated by the VTO.

Version 4 still gives me 2 issues:

- when I make a call to VTO (using 8001) I connect but without audio and video. To have both I need to set that extension as chan_sip at RasPBX (not pjsip)

- the unlock command (using #123# DTMF) still do not work as in version 3 so I need to use a cli command

Fortunately both issues have workarounds!

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What V4 firmware do you use? There are 2 available for VTO so far.

Are you sure you setup your SIP server correctly? No audio could be a a-law/u-law issue too. To be sure, start a SIP trace (either on asterisk with "sip set debug on" or with wireshark on PC with client software).

Do any other door opening code work (e.g.simply "1")? If not, do you send it as SIP INFO for sure? A trace would proof that as well.

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On 2/8/2019 at 3:00 PM, riogrande75 said:

What V4 firmware do you use? There are 2 available for VTO so far.

Are you sure you setup your SIP server correctly? No audio could be a a-law/u-law issue too. To be sure, start a SIP trace (either on asterisk with "sip set debug on" or with wireshark on PC with client software).

Do any other door opening code work (e.g.simply "1")? If not, do you send it as SIP INFO for sure? A trace would proof that as well.

I tested with two version 4 firmwares:

  • General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.1.R.20180606.bin
  • General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin

But in both cases I have the same problem: to be able to have sound and video in the call TO the vto I must use chan_sip (not pjsip). Note that in the case of calls FROM the vto the pjsip works great with audio and video.

Using the debug I found a line that could give some clues of what is wrong: No joint capabilities for 'video' media stream between our configuration((h264|mpeg4)) and incoming SDP((ulaw|ilbc))

Googling with this hint I found the thread that describes precisely my problem (by the way it seems there is no solution): https://community.asterisk.org/t/video-works-when-a-calls-b-not-when-b-calls-a/68375/8

Regarding the unlock, I tested several DTMF methods: Auto, RTP (RFC2833 / RFC4733), SIP INFO, Inband; and all of them did not allowed the unlocking (with the 123 code defined in the web interface or others that I also used instead). It seems a version 4 problem since version 3 works great: I just have to hit the #123# sequence and immediately I hear the relay unlocking.

But for me these 2 problems are not a big deal since as I wrote before I have workarounds (using chan_spi in the first problem, and using cli commands in the second).

So now I am beginning the next war: trying to build a web client in order to have a kind of VTH1550CH using a Raspberry Pi with an LCD.

For that I am trying SipML5 which uses WebRTC. Since WebRTC requires a secure connection, I am fighting now with certificates, TLS, RTC, WS, and some other monsters! :D

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Yes, I am using chan_sip as well, spares you a lot of trouble. Use v4 from 20180606 since the latter one does not even support asterisk.

I just use a simple SIP video phone as my indoor station. Acutally it almost works out of the box and is cheaper than a RPi with a display module (~60-80$). My plan was to use a cheap android tablet but I was not happy with any of the heavy loaded sip clients available. There are some good ones, but they don't support h.264 then.

If you really wanna develop it by your own, i suggest either linphone or DoorPi_Phone.

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4 hours ago, riogrande75 said:

Yes, I am using chan_sip as well, spares you a lot of trouble. Use v4 from 20180606 since the latter one does not even support asterisk.

I just use a simple SIP video phone as my indoor station. Acutally it almost works out of the box and is cheaper than a RPi with a display module (~60-80$). My plan was to use a cheap android tablet but I was not happy with any of the heavy loaded sip clients available. There are some good ones, but they don't support h.264 then.

If you really wanna develop it by your own, i suggest either linphone or DoorPi_Phone.

Right now I am using v4 from 20181030 and it is working with Asterisk (at least with the lastest version of RasPBX that has Asterisk 13.20.0).

In the SPI settings (as you can check in the attached image), I just had to:

  • disable the SPI server
  • set the IP address of RPi that has Asterisk
  • set the port to 5160 (since I am using chan_sip and not pjsip)
  • set the password
  • leave the SIP Domain empty

Regarding your "simple SIP video phone", I did not find a cheaper solution than the RPi+LCD (specially because I already have both). Can you tell me what video phone are you using?

Furthermore, a custom solution would allow the integration of other capabilities of my home automation system.

Using SipML5 I already manage the login of the web SIP phone client in Asterisk, but for now I still have errors when I try to initiate a call (Media stream permission denied).

For the initial steps I followed these links:

Regards

version4.png

SIPsettings.png

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Did not get 20181030 to work with my asterisk, moved back to a v3 to get it working fine.

I use a old Grandstream GXV3140 (40 bucks on ebay), so that I have all home phone issues on one device. Off course any SIP client could/might work with asterisk, but that belongs to another forum and has not a lot to do with VTO2000A.

 

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On 2/11/2019 at 11:59 AM, riogrande75 said:

Did not get 20181030 to work with my asterisk, moved back to a v3 to get it working fine.

I use a old Grandstream GXV3140 (40 bucks on ebay), so that I have all home phone issues on one device. Off course any SIP client could/might work with asterisk, but that belongs to another forum and has not a lot to do with VTO2000A.

Which version are you using? I just discovered this General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.zip on https://share.dahuasecurity.com

Do you have FreePBX running or is it enough, when I install  asterisk via apt-get asterisk on my raspberry?

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As far as I understand, FreePBX is simply a project adding a working webIf to asterisk. So everything should work as it does with a base asterisk.

I run a asterisk on raspbian, no special version, just the stuff that you get with apt-get.

  • Thanks 1

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I set up an asterisk server under 192.168.2.34

it has the following config files:

sip.conf

[general]
language=de
bindport = 5060
bindaddr = 0.0.0.0
type=friend
context=eingehend
externrefresh=30
nat=force_rport,comedia
srvlookup=yes
transport=udp
localnet=192.168.2.34/255.255.255.0
;directmedia=no

[8001]
host=dynamic
username=VTO2000A
type=friend
secret=qwertz
context=ausgehend
disallow=all
allow=ulaw
allow=h264

[videodoorgateway]
context=eingehend
type=friend
insecure=invite
nat=force_rport,comedia instead
username=621
fromuser=621
fromdomain=fritz.box
secret=AnyPassword
host=192.168.2.34
dtmfmode=rfc2833
disallow=all
allow=ulaw
;allow=h264

extensions.conf


[general]
static=yes
writeprotect=no

[ausgehend]
exten => _9901,1,Set(CALLERID(num)=9901)
exten => _9901,n,Dial(Local/alle@tfe-zuhause,50,w)

[tfe-zuhause]
exten =>  alle,1,Ringing()
exten =>  alle,n,Dial(SIP/9901@videodoorgateway,50,w)

[default]
include => ausgehend
include

When I press the button on the VTO I get:

[Feb 16 18:47:45] NOTICE[5538][C-00000001]: chan_sip.c:26273 handle_request_invite: Call from '621' (192.168.2.34:5060) to extension '9901' rejected because extension not found in context 'eingehend'.
[Feb 16 18:47:45] ERROR[5506]: cdr_csv.c:315 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory
[Feb 16 18:47:45] ERROR[5506]: cdr_csv.c:315 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : No such file or directory

 

Furthermore I can not register my MicroSIP client:

[Feb 16 18:48:55] NOTICE[5538]: chan_sip.c:28499 handle_request_register: Registration from '<sip:621@fritz.box>' failed for '192.168.2.142:53189' - Wrong password


Does anyone can help me?

MicroSIPSettings.PNG

VTO_SIP_Settings.PNG

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Pls. dont flood this VTO thread with asterisk basic questions! Pls. read either a asterisk tut, this thread from the beginning or post a new thread with your problem.

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On 11/22/2018 at 3:17 PM, JM170 said:

By the way, with the old SIP firmware, you could not read RFID cards with the VTO2000A-R module, now with the new SIP, yes i can !!!

How did you manage to get the RFID running? I only get entries in the unlock log. I can not find any register to teach my cards.

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