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.