Jump to content
Allodo

Dahua VTO2000A, SIP Firmware and Asterisk

Recommended Posts

There is only thing I was not able to figure out so far: How does the sabotage contact work? Is it just this noisy sound outside my door, or does anything happen on the network too (FTP-Upload, SIP-call,...)?

As far as I can remember it calls the "Centre Control Number" number

Share this post


Link to post
Share on other sites

Help!

 

I received my vto2000a today. I was able to connect via Webinterface. After i connected via cgi to the mjpeg stream, the Webgui was not reachable any more. I did an factory reset, ip changed but i can't connect via Browser or Configtool. Login via ssh is fine, but i can't do anything on the commandline? Any ideas?

Share this post


Link to post
Share on other sites

Hello,

 

I'm new to this forum and was reading the whole thread.

However one question is still open for me.

You all use asterisk to call your phones, when someone rings the door bell.

My question is: Can I use a simple android tablet without sim card to be called?

 

Many thanks.

Share this post


Link to post
Share on other sites
My question is: Can I use a simple android tablet without sim card to be called?

Off course. Just connect it to your wifi and install/configure a sip client app. I prefer BriaMobile ($$), but there are some good ones for free (e.g. zoiper).

Share this post


Link to post
Share on other sites

Hello all,

 

 

Dahua has removed the Telnet-Access in newer Firmware

I know. It is just uncomment in the start script

Help me, How do you uncomment it?

Share this post


Link to post
Share on other sites

Hi!

Is there anyone out there, who still has a old SIP firmware, where telnet is still included?

I would extract this one and check the difference to newer ones - then I am able to help.

Share this post


Link to post
Share on other sites

Hi together,

 

I have a VTO2000a and a VTH1510CH (IP/LAN Version) and would like to update my firmware. (some problems with the communication between both devices, which I think to be solved by updating the firmware)

 

Attached Firmwares (picture) have been provided by the support of my seller. But which Firmware should I install on which device? Maybe even better - do you know some link or so to download the actual firmware for my devices?

 

Best regards

Florian

1510174257_Bildschirmfoto2018-01-13um17_30_12.png.f1249668f8337a2fef3e77fb169612a4.png

Share this post


Link to post
Share on other sites

Hello,

 

I'm very interested by a VTO2000A from Aliexpress.

I read some posts in this thread and I would like confirmations...

 

- It needs a special firmware to use SIP ?

- If I have a SIP Phone which can handle SIP VIDEO (like GRANDSTREAM GXV3240) is the video will work ?

- In parallel, I would like to handle the video stream in my Surveillance Station system, is it working too ? Which protocol is it ?

- Is the VTO2000A is PoE compliant or not ? 802.3af/at ?

 

Thanks for your help !

Share this post


Link to post
Share on other sites
It needs a special firmware to use SIP ?

Yes, but this should not be a problem. The SIP FW comes officially from dahua.

If I have a SIP Phone which can handle SIP VIDEO (like GRANDSTREAM GXV3240) is the video will work ?

Yes, this should work. A long as the phone supports h.264 codec (like GXV3240 does), you do not have to transcode it.

In parallel, I would like to handle the video stream in my Surveillance Station system, is it working too ? Which protocol is it ?

Yes - I do it as well.

Is the VTO2000A is PoE compliant or not ? 802.3af/at ?

No - it needs 12-24V DC, which can off course be delivered via the spare wires of a CAT5/6/7 cable like many other "cheap" devices do.

Share this post


Link to post
Share on other sites

@all

 

Please help me!

 

I have the same problem as "jajecek".

 

My "DH-VTH1550CH" shows after the firmware update ("General_Overseas_VTH1510_Eng_P_SIP_V1.000.00.0.R.20170425.bin")

only "Welcome" and then a black screen.

 

A ping to the old IP address works.

 

What can I do?

 

 

Marcel

Share this post


Link to post
Share on other sites

For sip video calls I got a used Grandstream GXV3140 phone.

It supports h.264 codec, but unfortunately I am not able to get it working fine with my VTO2000A.

Call setup is ok, audio is good, but video is shown as a "grey pixel soup".

I guess, the phone can't handle VTO's "high-res" h264 video (800x480) from main profile.

I expect also, VTO's extra format when set to QVGA (320x24) might work on my phone.

 

Any Idea, where I might be able to set the used video profile for SIP calls to "extra"?

Share this post


Link to post
Share on other sites

Hello,

 

I'd like to share my current state of my setup.

I have a VTO2000A-C, FreePBX asterisk server on a Raspberry PI 3 and I'm using Linphone to get the calls from the door station.

The VTO is making a ringing sound, while the call is not responded. As soon I take the call on my Linphone the video stream is shown there.

I can open the door with 777 command.

 

The base for my configuration was this thread and this: https://ab-log.ru/smart-house/asterisk/sip-vto2000a

 

Firmware Upgrade

First of all I updated the firmware to SIP 1.2 in from the russian site. I did not found V1.2 on the Dahua servers. The only modification in this firmware is that the woman is talking russian.

I used config tool 4.05 to update the firmware of the VTO2000A. In the config tool you have to push the icon in the down left corner, then select VDP config.

I connected the VTO directly to my notebook and changed the IP of the interface to 192.168.1.100, as the standard IP of the VTO is 192.168.1.110. Then the config tool could connect to VTO.

 

Asterisk Installation on RPi3

On this site you can download a FreePBX image: http://www.raspberry-asterisk.org/downloads/

Use Etcher to burn it to the sd Card.

Don't forget the raspbx-upgrade. I used Putty to connect to the Raspberry.

 

Asterisk Config via SSH

The asterisk files are located under /etc/asterisk. The two files below were the only once I edited via SSH. All the rest I did via the FreePBX web interface (see next chapter)

 

My extensions_custom.conf:

[from-door]

exten => 9901,1,Ringing()
exten => 9901,n,Answer()
exten => 9901,n,Set(__DYNAMIC_FEATURES=door-open1)
exten => 9901,n,Dial(SIP/6001,30)  ;6001 is the linphone client on my smartphone
exten => 9901,n,Hangup()

[macro-door-open1]
exten => s,1,TrySystem(curl --user admin:admin --digest "http://192.168.0.90/cgi-bin/accessControl.cgi?action=openDoor&channel=1&UserID=101&Type=Remote")

 

192.168.0.90 is the IP of my VTO

192.168.0.132 is the IP of the RPi

 

The features_applicationmap_custom.conf contains the macro to open the door via phone code:

door-open1 => 777,self/callee,Macro,door-open1

 

 

Asterisk Config in the FreePBX WebInterface

At first start you will need to register.

 

At "Settings -> Asterisk SIP settings" you should set your network settings. At beginning I was trying to use PJSIP, but couldn't get this to work. So use normal SIP.

There it's important to enable Video Support and select H264 codec

https://ibb.co/i4yx87

 

Under Applications ->Extensions you can add the extensions.

In my case it's 6001 for the smartphone. THe context is from-internal (see Advanced tab). The video support needs to be on or inherited from the main settings.

DTMF Signaling needs to be on RFC2833 to make the unlock code work. Secret should be the password you use to connect from the linphone.

The other extension is the VTO 01018001 in my case. I could not change this in the VTO webinterface. The context is from-door. Also enable the video support.

https://ibb.co/gbjR1S

https://ibb.co/gwtm1S

https://ibb.co/eQqoFn

https://ibb.co/if3m1S

https://ibb.co/gtsDgS

 

VTO-2000A settings

https://ibb.co/kb7787

 

https://ibb.co/fb2UMS

https://ibb.co/f9DEo7

https://ibb.co/miKXan

https://ibb.co/ifg0T7

https://ibb.co/gHCw1S

 

 

Logfile of operation:

The FreePBX allows to view the logs in a convenient way. This is what I see when I push the door button:

2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip.conf': Found
[2018-02-12 14:46:47] VERBOSE[31216] loader.c: Reloading module 'res_pjsip_notify.so' (CLI/AMI PJSIP NOTIFY Support)
[2018-02-12 14:46:47] VERBOSE[31216] loader.c: Reloading module 'res_adsi.so' (ADSI Resource)
[2018-02-12 14:46:47] VERBOSE[31216] loader.c: Reloading module 'app_confbridge.so' (Conference Bridge Application)
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_general_additional.conf': Found
[2018-02-12 14:46:47] VERBOSE[31216] loader.c: Reloading module 'cel_manager.so' (Asterisk Manager Interface CEL Backend)
[2018-02-12 14:46:47] VERBOSE[31216] loader.c: Reloading module 'cel_odbc.so' (ODBC CEL backend)
[2018-02-12 14:46:47] VERBOSE[31216] config.c: Parsing '/etc/asterisk/cel_odbc.conf': Found
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_general_custom.conf': Found
[2018-02-12 14:46:47] VERBOSE[31216] config.c: Parsing '/etc/asterisk/cel_odbc_custom.conf': Found
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_nat.conf': Found
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_registrations_custom.conf': Found
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_registrations.conf': Found
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_custom.conf': Found
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_additional.conf': Found
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_custom_post.conf': Found
[2018-02-12 14:46:47] VERBOSE[31216] cel_odbc.c: Found CEL table cel@asteriskcdrdb.
[2018-02-12 14:46:47] VERBOSE[31216] loader.c: Reloading module 'codec_speex.so' (Speex Coder/Decoder)
[2018-02-12 14:46:47] VERBOSE[31216] loader.c: Reloading module 'app_voicemail.so' (Comedian Mail (Voicemail System))
[2018-02-12 14:46:47] VERBOSE[31216] config.c: Parsing '/etc/asterisk/voicemail.conf': Found
[2018-02-12 14:46:47] WARNING[31216] app_voicemail.c: maxsilence should be less than minsecs or you may get empty messages
[2018-02-12 14:46:47] VERBOSE[31216] loader.c: Reloading module 'app_playback.so' (Sound File Playback Application)
[2018-02-12 14:46:47] VERBOSE[31216] loader.c: Reloading module 'app_queue.so' (True Call Queueing)
[2018-02-12 14:46:47] NOTICE[31216] app_queue.c: No queuerules.conf file found, queues will not follow penalty rules
[2018-02-12 14:46:47] VERBOSE[6257] netsock2.c: Using SIP TOS bits 96
[2018-02-12 14:46:47] VERBOSE[6257] netsock2.c: Using SIP CoS mark 4
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_notify.conf': Found
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_notify_custom.conf': Found
[2018-02-12 14:46:47] VERBOSE[6257] config.c: Parsing '/etc/asterisk/sip_notify_additional.conf': Found
[2018-02-12 14:47:04] VERBOSE[6257][C-0000001b] netsock2.c: Using SIP VIDEO TOS bits 136
[2018-02-12 14:47:04] VERBOSE[6257][C-0000001b] netsock2.c: Using SIP VIDEO CoS mark 6
[2018-02-12 14:47:04] VERBOSE[6257][C-0000001b] netsock2.c: Using SIP RTP TOS bits 184
[2018-02-12 14:47:04] VERBOSE[6257][C-0000001b] netsock2.c: Using SIP RTP CoS mark 5
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] pbx.c: Executing [9901@from-door:1] Ringing("SIP/01018001-00000031", "") in new stack
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] pbx.c: Executing [9901@from-door:2] Answer("SIP/01018001-00000031", "") in new stack
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] pbx.c: Executing [9901@from-door:3] Set("SIP/01018001-00000031", "__DYNAMIC_FEATURES=door-open1") in new stack
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] pbx.c: Executing [9901@from-door:4] Dial("SIP/01018001-00000031", "SIP/6001,30") in new stack
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] netsock2.c: Using SIP VIDEO TOS bits 136
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] netsock2.c: Using SIP VIDEO CoS mark 6
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] netsock2.c: Using SIP RTP TOS bits 184
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] netsock2.c: Using SIP RTP CoS mark 5
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] app_dial.c: Called SIP/6001
[2018-02-12 14:47:04] VERBOSE[31227][C-0000001b] app_dial.c: SIP/01018001-00000031 requested media update control 26, passing it to SIP/6001-00000032
[2018-02-12 14:47:05] VERBOSE[31227][C-0000001b] app_dial.c: SIP/6001-00000032 is ringing
[2018-02-12 14:47:05] VERBOSE[31227][C-0000001b] app_dial.c: SIP/6001-00000032 is ringing
[2018-02-12 14:47:06] VERBOSE[31227][C-0000001b] app_dial.c: SIP/6001-00000032 answered SIP/01018001-00000031
[2018-02-12 14:47:06] VERBOSE[31228][C-0000001b] bridge_channel.c: Channel SIP/6001-00000032 joined 'simple_bridge' basic-bridge <513c67dd-2239-4e95-ada7-39891d087f9a>
[2018-02-12 14:47:06] VERBOSE[31227][C-0000001b] bridge_channel.c: Channel SIP/01018001-00000031 joined 'simple_bridge' basic-bridge <513c67dd-2239-4e95-ada7-39891d087f9a>
[2018-02-12 14:47:28] VERBOSE[31228][C-0000001b] pbx.c: Executing [s@macro-door-open1:1] TrySystem("SIP/6001-00000032", "curl --user admin:admin --digest "http://192.168.0.90/cgi-bin/accessControl.cgi?action=openDoor&channel=1&UserID=101&Type=Remote"") in new stack
[2018-02-12 14:47:32] VERBOSE[31228][C-0000001b] bridge_channel.c: Channel SIP/6001-00000032 left 'simple_bridge' basic-bridge <513c67dd-2239-4e95-ada7-39891d087f9a>
[2018-02-12 14:47:32] VERBOSE[31227][C-0000001b] bridge_channel.c: Channel SIP/01018001-00000031 left 'simple_bridge' basic-bridge <513c67dd-2239-4e95-ada7-39891d087f9a>
[2018-02-12 14:47:32] VERBOSE[31227][C-0000001b] pbx.c: Spawn extension (from-door, 9901, 4) exited non-zero on 'SIP/01018001-00000031'

 

 

I'm wondering what the spawn extension at the end does mean. The hangup application does not occur there, maybe it's this one.

 

Currently working on

What I'm still searching for is way too show the camera picture before the call is accepted. Video or snapshot.

Also I have two problems with the wiring.

1) I couldn't start the VTO with only one CAT wire attached. I have bought a NETgear POE switch, but does not work. Then I bought some passive POE adapters (https://www.amazon.de/DIGITUS-Professional-Kabelset-Splitter-Steckerdurchmesser/dp/B002NBN72C/ref=pd_sim_147_6?_encoding=UTF8&psc=1&refRID=Y930XJ3F30091BTZ1G1R). I used only one of them, but does also not work.

The other problem is that the E-Opener is controlled by my SPS inside. I would need an additional cable from the VTO to my SPS to get the Open Door Command or use some spare wires of the CAT, but these are used for the power supply as I understood.

 

I think I will need to get the command from the RPi somehow.

Share this post


Link to post
Share on other sites
What I'm still searching for is way too show the camera picture before the call is accepted. Video or snapshot.

Did you have a look at this:

https://www.ip-phone-forum.de/threads/im-dialplan-per-script-ein-snapshot-erzeugen-f%C3%BCr-die-fritzbox-t%C3%BCrsprechanlage.288096/#post-2185372

Also I have two problems with the wiring.

I just did wiring of the 24V via wires 4/5 and 7/8 by myself. I pulled out the power wires and then I crimped a RJ45 at the remaining ethernet wires (1/2+3/6).

The other problem is that the E-Opener is controlled by my SPS inside. I would need an additional cable from the VTO to my SPS to get the Open Door Command or use some spare wires of the CAT, but these are used for the power supply as I understood. I think I will need to get the command from the RPi somehow.

Do you know the HTTP API for Dahua devices: https://support.amcrest.com/hc/en-us/articles/232310368-HTTP-API-SDK-for-Amcrest-products

This might help you.

Share this post


Link to post
Share on other sites

Hi rio,

 

thanks for the links. I already had a look on it.

 

I did some updates to my config and like too share it:

 

[from-door]
;;;;;;;;;;;;;;;;;;
exten => 9901,1,Ringing()
exten => 9901,n,Answer()
;Get a snapshot and save locally, -y option is needed, otherwise Asterisk will hang because the system asks, whether it shall overwrite the old file
exten => 9901,n,TrySystem(avconv -i 'rtsp://admin:admin@192.168.0.90' -f image2 -vframes 1 -pix_fmt yuvj420p /tmp/snapshot1.jpg -y)
;Send photo to telegram chat group "Haus"
exten => 9901,n,TrySystem(/home/pi/tg_photo.sh Haus "/tmp/snapshot1.jpg")
;With this a text message can be sent to telegram chat group "Haus"
;exten => 9901,n,TrySystem(/home/pi/tg.sh Haus "Hello World")
;enable door open macro
exten => 9901,n,Set(__DYNAMIC_FEATURES=door-open1)
;Call SIP clients
exten => 9901,n,Dial(SIP/6001&SIP/610&SIP/6003,30)
;quit call
exten => 9901,n,Hangup()
;;;;;;;;;;;;;;;;;;

[macro-door-open1]
;Open door by using cgi
exten => s,1,TrySystem(curl --user admin:admin --digest "http://192.168.0.90/cgi-bin/accessControl.cgi?action=openDoor&channel=1&UserID=101&Type=Remote")

 

I tried to implement the Telegram solution, meaning the pi sends a snapshot too a telegram account, when someone pushes the door bell.

Therefore I needed to install the telegram-cli on the pi. I was doing this according to this guideline https://pimylifeup.com/raspberry-pi-telegram-cli

I installed the tool as user pi in the home directory.

I also discovered, that in the line

exten => 9901,n,TrySystem(avconv -i 'rtsp://admin:admin@192.168.0.90' -f image2 -vframes 1 -pix_fmt yuvj420p /tmp/snapshot1.jpg -y)

the -y option is very important. It will make the shell overwrite the old file without asking. WIthout this option the asterisk was hanging.

 

I also needed some time too realize that asterisk is run by the user asterisk and not pi. As I did all installation via SSH and user pi it's important too chmod the scripts tg_photo.sh and tg.sh to the user asterisk. A good way is to test the commands on the command line by using

sudo -u asterisk BEFEHL

where BEFEHL should be replaced by the command that you want to use.

 

By this I also discovered that Telegram needs to be registered by the user asterisk, otherwise it will not work:

You can do this by calling

 sudo -u asterisk /home/pi/tg/bin/telegram-cli -k tg-server.pub -W

Then your going to be asked for your telephone number and so on...

 

Next issue that I see with the telegram:

The pi is sending the snapshot to my own telegram account, meaning I don't get any notification. Maybe someone has a workaround for this, I didn't find one so far.

The last solution would be to get a separate cell phone number for this.

Share this post


Link to post
Share on other sites
Any Idea, where I might be able to set the used video profile for SIP calls to "extra"?

I guess I found the solution by myself!

 

I found out, that the file that gets created with "Export Config" is a simple textfile.

After analyzing it, the parameters regarding the "TalkbackDevice" seemd to me to have something to do with the SIP settings.

"TalkbackDevice": {"AudioChannel":0,"AudioEnable":true,"AudioStream":"Main","VideoChannel":0,"VideoEnable":true,"VideoStream":"Main"},

So I changed the VideoStream for TalkBackDevice from "Main" to "Extra1" and uploaded the config again. The "AudioStream" parameters changed itself then to "Extra1".

And now I get a resolution of 320x240 in SIP video calls. It really seems that did the trick!

 

Maybe this is helpful for other issues regarding video as well - just edit the config manually.

Share this post


Link to post
Share on other sites

While checking the config.export, I found the parameter "SIPServerRedundancy:x.x.x.x" and some corresponding variables at the very end of the file:

SIPServerRedundancy:192.168.1.39,
UserEnable:true
UserID:19
UserType:0}
Web:{Enable:true,Port:80}
DialRule:{BuildingModeEnable:false
DiaSeperator:#
ExtentionModeEnable:true
UnitModeEnable:false}
...
SIPKeepAlive:{FailTimes:3,KeepAliveTime:60,SipServerEnable:true}}

It looks like the VTO2000A might be able to handle a redundant SIP server as soon as the keepalive mechanism detects a fault. At least DAHUA tried to implement something like this.

I put the IP of my backup asterisk there, but so far it don't want to register there (main server disable, off course).

It would make sense for a security device like a doorbell to have such feature.

Share this post


Link to post
Share on other sites

Since a few days there is a brand new SIP fw out:ftp://ftp.asm.cz/Dahua/videovratni/VTO2000A,VTO2000A-C/firmware/SIP_20180105/General_VTOXXX_Eng_P_16M_SIP_V3.300.0000001.0.R.20180105.bin

No info regarding changes so far (as usual).

 

Any testers arround?

Share this post


Link to post
Share on other sites

Dear all,

I am just struggling with my new VTO2000A. Received it out of the box with the non SIP firmware, so I changed the FW to the V3.300.0000 which was no problem at all using the Config Tool V3.20.0. I could log in using the port 3800 and also the upgrade was no problem at all. so fine so good.

But then I noticed that with this FW version I could not register new RFID tags with the VTO2000 RFID module (the fingerprint module still works with this FW). So I wanted to "fall back"to the FW version V1.000.0 (R20170425). But now suddenly I have no chance any more to log in with the configuration tool on port 3800. The "normal"port 37777 still works with the config tool, but using this port neither a flash nor a FW update is possible.

Can anybody confirm this behaviour, and much more important, does anybody know how to get the Firmware changed again?

Thank you very much and best greetings

Chris

Share this post


Link to post
Share on other sites

@Chris_Allgaeu

Having the exact same behaviour, also have the fingerprint reader, but adding new fingerprints does not work for me with SIP firmware 3.3

I also want to go back to the SIP firmware from April 2017 - tcp port 3800 is not listening on the VTO, tcp/37777 is open.

ssh access works, but limited only to some commands. idea was to start upgraded daemon so we have tcp/3800 back in action, but no luck..

I read that telnet seems to be disabled in a startup script, no idea how to get this enabled.

 

Were you able to downgrade the firmware lately?

Just saw you on some other forums with the same question

Share this post


Link to post
Share on other sites

Huh, Just thought on upgrading.

Chris, Flexstarr you've saved me from making community owned VTO unusable. Thanks guys!!

Wish you resolve your issues soon.

Hope Dahua will release usable SIP firmware someday. Currently using VTO-embedded SIP server but without video and issues with routing...

Share this post


Link to post
Share on other sites

Hi,

I have setup 2 vto2000a + raspbx + vth1550 + Cisco PAP2T (ata adapter) + GSWave mobile client

 

Basically when someone call the vto all mobiles ring + all home phones thru the ata which is connected to my analog PBX

 

All works well except the vth1550 showing video when someone call but when I answer the video disappear - audio and unlock OK. Any idea? Thanks.

Share this post


Link to post
Share on other sites

BTW - there are new 2018 sip versions for vth and vto and now support vth5221dw as well....I will try and keep you posted.

Share this post


Link to post
Share on other sites

hi eveybody

 

i've just upgraded my vto2000 to latest sip firmware posted above but i cannot login anymore neither from web and from config tool telnet or wathsoever.

The vto respond to ping to the right ip and the only open port is the tcp 3800 but seems to bring me nowhere (login overtime via configtool), close immediatly

 

I've just tryed to reset pressing ring button and gave it power (with blinking panel) but the situation rest that the vto is completely unaccessible.

 

What can i do now?

 

thanks

Share this post


Link to post
Share on other sites

Just have everything running with SIP v3.3:

- Videocall to GXV3240 with h264

- Call to Fritzbox phones with preview picture snapshoted when someone pushes the doorbell, also ffmpeg mjpeg solution with low framerate works.

- Push notifications with Pushover and Telegram

 

Only thing I really would like is disabling the stupid sounds like "calling now, please wait.." or "end of the call".

Was that possible in the SIP fw from April 2017?

 

Another thing is the preview video before answering the call on the GXV3240. Screen remains black until I push "video answer" button on screen.

Does that someone have running? Maybe to specific..

Share this post


Link to post
Share on other sites
Only thing I really would like is disabling the stupid sounds like "calling now, please wait.." or "end of the call".

Was that possible in the SIP fw from April 2017?

In fact yes, if you are experienced enough to modify the firmware with the toolset mentioned above (BotoX/Dahua-Firmware-Mod-Kit). I dunno the changes of V3.3, but I guess everything of your setup should work with 201704 too.

Another thing is the preview video before answering the call on the GXV3240. Screen remains black until I push "video answer" button on screen.

Does that someone have running? Maybe to specific..

I have a similar setup like you, but my phone is a GXV3140 and server is asterisk. By lowering the video resolution for voip calls (not the rtsp stream!), I got it working just like you.

To have video in the phones displayed before actually answering the call, you need to activate a SIP function called early media. Currently I am working on this issue as well.

Share this post


Link to post
Share on other sites

Create an account or sign in to comment

You need to be a member in order to leave a comment

Create an account

Sign up for a new account in our community. It's easy!

Register a new account

Sign in

Already have an account? Sign in here.

Sign In Now

×