riogrande75

Either check this VTO2000A and SIP video resolution or try latest firmware from here. I remember there were some changes from dahua regarding Grandstream phone support.

I created a sticky thread in another forum where I try to keep all fw files for VTO/VTH up2date: Dahua VTO/VDP Firmware Link Collection Check it out and let me know if I missed something or if there any newer files arround.

As long as the cam "speaks" rtsp and you can set limits on your NAS, yes!

Found a multilanguage pack for latest SIP FW 20190305 in another forum: General_VTOXXX-data_EngDthFrnGerItlPorSpa_P_16M_SIP_PART_V4.300.0000001.0.R.20190322.bin So dutch, french, german, italian, portugese and spanish prompts are available now. Same for almost all VTH's: vth5221d_4_3_multilanguage_file.bin

Ok. Did not find this very old one, but why not tryin another 3.1 FW "General_Multi3_VTH1510_EngRusItlFreGerDutSpaPor_P_V3.100.0000.0.R.20170401", which is just slightly newer and easy to find?

Whatfor do you need such an old firmware?

If you have ever set up any SIP device, then it should not be a big deal to configure stn-number, server ip, port, username+pw in the VTO's webif in section "Netwok Settings". If you don't know how to set up a SIP port your pabx (NS700?), then you might have more luck in a panasonic pabx forum .

I would add videosupport, allow alaw for all stations (european standard coding) and set dtmfmode to sip info (opening the door will be possible from sip pohones). But that has nothing to do with your problem. [19] type=friend host=dynamic secret=xxxx qualify=yes language=de username=19 callerid="Haustuer" <19> disallow=all canreinvite=no allow=h264 allow=alaw allow=ulaw videosupport=yes insecure=invite dtmfmode=info For diagnosis I would make it easier first, remove 9901 and different context stuff and let VTO call directly to VTH 9902. Also attach "sip show peers" to make analysis easier.

Well, at my home it works with v4.3 fw on VTH5221, v3 on VTO and asterisk 11. If you add asterisk logs (sip set debug on), then we might get a clue on what's wrong. Also add relevant sections of sip.conf.

I guess you did not get it, what a forum is for If you find someone who takes you by the hand and does all the work for you, pls. let me know too...? I suggest you do some asterisk/freePBX tut's first, read this thread from the beginning and then once again. If you still don't know what it is all about, ask your local dealer for help.

According to the install file, yes: root@vlinux3:~/Dahua-Firmware-Mod-Kit/General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin.extracted# cat Install { "Commands" : [ "burn dm365_ubl_boot_16M.bin.img bootloader", "burn custom-x.cramfs.img custom", "burn pd-x.cramfs.img pd", "burn kernel-x.cramfs.img kernel", "burn romfs-x.cramfs.img rootfs", "burn user-x.cramfs.img user", "burn web-x.cramfs.img web", "burn data-x.cramfs.img data", "burn gui-x.cramfs.img gui", "burn pcm-x.cramfs.img pcm" ], "Devices" : [ [ "VTO1210A-X", "1.00" ], [ "VTO1210B-X", "1.00" ], [ "VTO1210C-X", "1.00" ], [ "VTO1220A", "1.00" ], [ "VTO1220B", "1.00" ], [ "VTO6110B", "1.00" ], [ "VTO6000C", "1.00" ], <<<<<<<<<<<<<<<<<< [ "VTO6100C", "1.00" ], [ "VTO6100C", "1.10" ], [ "VTO6100C", "1.20" ], [ "VTO6210B", "1.00" ], [ "VTO6210B", "1.10" ], [ "VTO2000A", "1.00" ], [ "VTO2000A", "1.10" ], [ "VTO2000A", "1.20" ] ], "Vendor" : "General" }

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.

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.

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.

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.

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.

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.

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.

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.

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.

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.

No sorry - mistake. I got a SIP v3 firmware dated 20181229 english only.

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.

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.

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.