naponet 0 Posted February 6, 2019 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. Share this post Link to post Share on other sites
awschmidt 0 Posted February 6, 2019 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,... ? Share this post Link to post Share on other sites
naponet 0 Posted February 7, 2019 Hello, the firmware consists of two parts, the second language, I could not even flash that ... Share this post Link to post Share on other sites
hsmptg 0 Posted February 8, 2019 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? Share this post Link to post Share on other sites
riogrande75 3 Posted February 8, 2019 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. Share this post Link to post Share on other sites
hsmptg 0 Posted February 8, 2019 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! Share this post Link to post Share on other sites
riogrande75 3 Posted February 8, 2019 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. Share this post Link to post Share on other sites
hsmptg 0 Posted February 9, 2019 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! Share this post Link to post Share on other sites
riogrande75 3 Posted February 10, 2019 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$). My plan was to use a cheap android tablet but I was not happy with any of the heavy loaded sip clients available. There are some good ones, but they don't support h.264 then. If you really wanna develop it by your own, i suggest either linphone or DoorPi_Phone. Share this post Link to post Share on other sites
hsmptg 0 Posted February 10, 2019 4 hours ago, riogrande75 said: Yes, I am using chan_sip as well, spares you a lot of trouble. Use v4 from 20180606 since the latter one does not even support asterisk. I just use a simple SIP video phone as my indoor station. Acutally it almost works out of the box and is cheaper than a RPi with a display module (~60-80$). 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: SIPml5 Installed on Raspberry Pi 2 Asterisk Server: https://www.algissalys.com/how-to/sipml5-installed-on-raspberry-pi-2-asterisk-server WebRTC tutorial using SIPML5: https://wiki.asterisk.org/wiki/display/AST/WebRTC+tutorial+using+SIPML5 Regards Share this post Link to post Share on other sites
riogrande75 3 Posted February 11, 2019 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. Share this post Link to post Share on other sites
matlab22 0 Posted February 14, 2019 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? Share this post Link to post Share on other sites
riogrande75 3 Posted February 14, 2019 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. 1 Share this post Link to post Share on other sites
matlab22 0 Posted February 16, 2019 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? Share this post Link to post Share on other sites
riogrande75 3 Posted February 16, 2019 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. Share this post Link to post Share on other sites
matlab22 0 Posted February 16, 2019 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. Share this post Link to post Share on other sites
Soft 0 Posted February 18, 2019 On 2/16/2018 at 11:53 AM, riogrande75 said: 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. hello! How did you read the file? I get the file from vto-2111D (sip firmware) and can't read it at all. It is a little part of it: MWPZWJGSK#M)EАщOK-‰m|=hЭyJ®вЁЌ”ЉЪ„єi|еГ‡є~qзА±зrЗ¶д‰Г>Э8¬фn.„зБ-«ЮFЎ…Ыvъd¬—E`µ=іB-µcкц¶]¶»UаXЫчV#ГТБ-RµЁN‡ьД€NъPйuх‡ЉЏpYY‘<Ёз®…М“"9А°%UЂµ]Sёґ›Ее (к@~mЖь(LЈ—™•ЛпЕ1Ъg@,k§Ё©ЧШТЩ ‚¬&~ h eМ?[єеgµDx`‡РПЖ"эЋАH$·џс—WЉ!Ђ†‘їђдIFфД•вр‰$іНљNJ\±ќз&ЙMЕ¦ю–7?Ѓч‘~'¦љЧсpґikЛф‰ЯV“}웧6wоTZLE}uLц¶Ѓ;J›sµ” ЏрщИ>4&mKG€БЄ7ё¶LТд [rВ„AЭ‹UЁ;Џ†mшЖФЎHцщЕАH<©|а1’УЉ‹мчd#Мй„Ґ9ї¶дЗ6Ґ;nс<ЉЏЛ#І@·чЬхзХvL configFileExport.txt Share this post Link to post Share on other sites
riogrande75 3 Posted February 18, 2019 As mentioned (I hope), you have to use an old unsigned firmware. There even the config was not encrypted, so it could be read in clear text. Share this post Link to post Share on other sites
matlab22 0 Posted February 18, 2019 On 2/16/2019 at 8:03 PM, matlab22 said: 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. I downgraded from General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin to General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.bin, but I am still not able to register my rfid cards. The issue card button is still missing. Does anybudy has the same issue? BTW SIP is now working in all versions: General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030 General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.1.R.20180606 General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229 Share this post Link to post Share on other sites
naponet 0 Posted February 21, 2019 Hello everyone, I'm desperate, I updated a vto1210c-x with the webversion 4, I want to downgrade but I do not do it anymore, trying with configtoool and the 37777 port appears to me trying with the 3800 door appears to me I deice to change the parameters because the device is not in the same range of the computer (from the browser I manage it correctly) click on ok and it appears I tried them all, help.Thank you. Share this post Link to post Share on other sites
matlab22 0 Posted February 24, 2019 I am using the VDPconfig Tool from the ToolBox. So fare I did not had any problem down and upgrading my VTO. Share this post Link to post Share on other sites
mimk97 0 Posted February 27, 2019 On 11/24/2017 at 12:11 PM, bassbumper said: @Allodo There are 2 options. Either you take an old (non-sip) firmware which includes multilanguage or you use one of the "Config" firmwares which Dahua provides on the firmware website. In both cases you extract the firmware/Config the same way and copy the included language (txt) files to your mod firmware. In addition you have to adapt "i18n" and "ProductInfo" file in your mod firmware (data-x and custom-x partition) to accept the new files. Where is a good place to upload? Yes, you are right. I guess there is not chance as long as we cannot decompile sonia file But asterisk is also necessary if you want to include other devices and automate actions. I use asterisk in my current scenario with: - VTO - VTH - Fritzbox with Fritzphones - Asterisk - Homeautomation (Outdoor Lights, etc) - Mobile Phones - Snapshots All devices with sound and Picture/Video and door unlock functionality Do you have a Setup Guide for Install VTO with Asterisk and Fritzfon? Share this post Link to post Share on other sites
mimk97 0 Posted March 7, 2019 On 5/2/2018 at 11:04 PM, Flexstarr said: 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.. Hi, how do you solve "...preview picture snapshoted when someone pushes the doorbell, also ffmpeg mjpeg solution with low framerate works" ? Share this post Link to post Share on other sites
HDman 0 Posted March 10, 2019 I looking for a solution for a working push notification for house with 20 rooms and vth1550 monitors. So basically the people should be able to receive push to there smartphones. When someone rings his number of the room. Is this possible with out of the box dahua firmware + app (i dont find any possibility to set up room number in the app) or is a SIP solution needed for this to work with multiple rooms Share this post Link to post Share on other sites
19fabiom84 0 Posted March 11, 2019 Hello Guys, does someone knows if SIP firmware v4 can be installed in VTO6100? Now, I have v1 installed and it is working with Asterisk but I would like to deactivate sounds and I read that it is possible with v4. Thank you for your help. Fabio Share this post Link to post Share on other sites
