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Do you did make a reset over ssh? I had too no luck without reset, but with reset it works good.

Hello Vytasm! Can you help me please - this instruvtions shows how to change the Language files on the VTO2000. I tryed like you to make it for the VTH15xx but i don't what a destination should set at step 8. Becaus by the VTH i don't become the Message "WARNING Autodetected config: VTO" Or maybe you can upload you ready Firmware? There are two-three ways to solve this:- you set at you Dial() command in asterisks dialplan the option r or R or together. - look at Progress(), the progressinband setting in sip.conf or Ringing() if you would like to avoid the use of 'r' but have issues with the ringback behaviour of Dial(). - make in you diaplan as first step an answer and play after that ring tones. But this is ... http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial I'm sorry, but with the SIP-Protocol you must do this in you Asterisk self. There is no way at the VTO or other dialing SIP-Device to discover this situation - it is simply impossible. The only who can send a message to the VTO back when ringing is canceled, is the Asterisk self. The Panel (SIP-Client) communicates in the Ringing phase only with the Asterisk and with no body else. In this moment the VTO doesn't know that there is a Panel or more then one Clients.... He "see" only the Asterisk, nothing else. The whole Communication is going over the Asterisk. So you can modify you dialapln in such way, that when in Asterisk occure a disconnect event, the whole call to VTO is canceled. https://de.wikipedia.org/wiki/Session_Initiation_Protocol#/media/File:SIP-B2BUA-call-flow.png Look that there is no direct connection between Alice and Boris. The only Connection is the SIP-PBX. You have (may be!) direct connection between VTO and VTH ONLY if the Client is going to answer the call. And then ONLY when the RTP-Media Data a redirected between the VTO and the VTH. That is in sip-calls not allways the Case! The option in asterisk for this is canreinvite. If it is set to NO on one or both of the sides - the stream will going only true the asterisk. This is why i wrote may be. Best Regards and excuse me for my bad english!

Hello! Wrong connection - nooo.... i don't believe this. It is so simple the connection, thet there is no place to going somenthing worng. But in the firmware is it possible, that the setting of the wait time is not fuctioning correct. I never changed the time to see if this works. For me was it enough 1x to open. And i teste that much later, more than 30 or 60sec., the second opening works too. The Porblem is, i don't use any more the nonSIP-Firmware and switsched to the SIP-Protocol. So if i can chek it eve by me. but i don't know if this will help you, because we use completely different Firmware. I test it by me and write again. Offtopic: There is a way per Web Request to open the door too. There is no delay restriction in this case.

I think there is a time after the frist opening that you have to wait, before you can open the door again. By me it is so. But by me doesn't freeze the VTO. May be is yours broken? The wait time is to find/set in the configuration webpage of the VTO.

Hi! I'm not sure for what it is good? In the API PDF stays written: I'm from germany and here exist a Term - Ich verstehe nur Bahnhof (I understand only Station) Meaning - i don't understand nothing. I don't understand the Context of the API Call.... It is may be for the nonSIP-Firmware. Were a Client must register at the VTO?! The example IP-Adress is a Mutlicast-Addres - such thing doesn't exist usualy by SIP. So i think this is all for the nonSIP-Firmware. But i'm not 100% sure. I'm sorry it is to high for me

Hello! Do you tryed the Example: http://192.168.xxx.xxx/cgi-bin/accessControl.cgi?action=openDoor&channel=1&UserID=101&Type=Remote It works fine by me. In the PDF i read: URL-Syntax: http://<ip>/cgi-bin/accessControl.cgi?action=openDoor&channel=<channelNo>[&UserID=<UserID>&Type=<Type>] Param in: channelNo: channelNo: the index of door. Start from 1; The followings are optional： UserID: remote User ID; Type: the open type; default value is “Remote”. For example： http://<ip>/cgi-bin/accessControl.cgi?action=openDoor&channel=1&UserID=101&Type=Remote OK, why 101 - i don't know. I assumed it must be a SIP-DID, or the own 8001 for example... But no. May be 101 is the Linux-UID from admin? Edit: as User ID works 102 too and 103 and ....

There is a new (for me) SIP-Firmware for the VTH indoor Station:ftp://ftp.asm.cz/Dahua/videovratni/VTH15xx/firmware/20160930-CZ-SIP.ZIP

And here some Docu to the links that Foxi found in sonia: ftp://ftp.asm.cz/Dahua/videovratni/_SDK/DAHUA_HTTP_CGI_FOR_VDP.pdf Its only a small number of Links with a little bit more information.

Hi! Yes i tryed it, but become only audio and not video. The same like you. At Asterisk i become the message unknown rtp codec 95. I think that Elcom use Mjpeg (for access as IP Camera?) and something other (VP8/9?) for SIP (Really?!). The App is created from Zoiper for Elcom and it seems, that it is specialy developed for Elcom DoorPhone and Cameras. May be i'm wrong, but i think i remember that codec 95 message occurs only if i set in the original Zoiper App the V9 Video-Codet to on. Edit: Ha.... here stay that 95 is unasigned.... Great http://www.iana.org/assignments/rtp-parameters/rtp-parameters.xhtml

Hello pkozul! Please try to set dtmfmode=info on both sides - not only at the 8001. The conversation from rfc2833 to info should work, but if the other side is set to inband (audio) then asterisk can not transfer the dtmf tone to info. It continous to came as sound at the VTO. Do you hear the tone when you press the keys? If yes, then this is the case. Or show us please you other DID, of the other side, i see here only the 8001. Thanks!

Hello Serge! I tested some with Pushbullet on my Android Xperia M4. So far i now, Pushbullet works with iPhone too. He can send not only Notes per Push - he can send Files (Pictures) too. It is a little bit complicated, than Notes, but it works perfect. And Pushbullet is for free if you don't need Support. Here is a call how to send a Note: #!/bin/bash API="API-Key kommt hier rein" MSG="$1" curl -u $API: https://api.pushbullet.com/v2/pushes -d device_iden="iden-code wohin die Nachricht gehen soll" -d type=note -d title="Alert from Pi" -d body="$MSG" More or less from here: https://jankarres.de/2014/03/raspberry-pi-push-benachrichtigung-bei-ssh-anmeldung/ Here is a Script to Send a files: #!/usr/bin/perl use JSON; use LWP::UserAgent; # credentials and device identifier my $api_key = 'XXXXXXXXXXXXXXXXXXXXXXXXXXX'; my $password = 'XXXXXXXX'; # for what?!?!? my $device = 'XXXXXXXXXXXXXX'; # iden-code come here # file to send my $file_name = 'image.png'; my $file_type = 'image/png'; my $user_agent = LWP::UserAgent->new(); $user_agent->agent('My File Pusher v1.0'); # Send a request for authorisation to upload a file .. my $request_response = $user_agent->post('https://api.pushbullet.com/v2/upload-request', 'Authorization' => 'Bearer '.$api_key, 'Content' => ['file_name' => $file_name, 'file_type' => $file_type]); if ($request_response->is_success) { # no obvious error from the server, parse the JSON .. my $json = from_json($request_response->content); if ($json->{'upload_url'}) { # we have an end point to push to and we can upload the file my @data; # the order of these fields is important, AWS will cry if you don't respect the ordering!! foreach my $item ('awsaccesskeyid', 'acl', 'key', 'signature', 'policy', 'content-type') { push(@data, $item => $json->{'data'}->{$item}) } # and add the file push(@data, 'file' => [ $file_name ]); # now attempt to upload the file my $upload_response = $user_agent->post($json->{'upload_url'}, 'Content_Type' => 'form-data', 'Content' => \@data); if ($upload_response->is_success) { # looks like we uploaded the file successfully # so now send the actual push to the device my $push_response = $user_agent->post('https://api.pushbullet.com/v2/pushes', 'Authorization' => 'Bearer '.$api_key, 'Content-Type' => 'application/json', 'Content' => to_json({'device_iden' => $device, 'type' => 'file', 'file_name' => $json->{'file_name'}, 'file_type' => $json->{'file_type'}, 'file_url' => $json->{'file_url'}})); if ($push_response->is_success) { print "Pushed OK!\n"; } else { print "Push failed!\n"; } } } } Source: May be it works for you? Greetings Edit: I'm using Pushbullet to send a Note with a Key-Word. Under Android i have the App TASKER which looks what for Notes came in per Pushbullet. When he see the Keyword he start by me IP Cam Viewer App. But it should work by you without Tasker an only with a Picture too. Simple sends of Pictures per script (from here) from my Raspberry to Sony M4 works perfect. Delay is not more then 1-2 sec., but it depends of course on the internet connection. When no connection, Push comes later, when connection is reestablished.

Yes, i can confirm that. And this 3 sec. are logical: the VTO is waiting to see if there will folow more Key input and so far there no any more, he evaluate the input - in your case 55, in my 123. Thanks!

Hello Serge! Thank you for the mksquashfs statement. So if correct understand, the only reason to install squashfs-tools is that build.py can find the prerequisite file at start (mksquashfs)? I see.... ok, then symbolic sholud be fully OK and not neccesary to install the tools. I was not 100% sure, that Version is really set to 3 in build.py... But now i can sleep quiet. Thank You!

Hello Serge! Great! Thanks!!! Edit 1: Can you explain me please, how it works wit the unsquashfs or better with the mksquashfs? The squash tools 3.4 installs only a binary mksquashfs3 in /usr/local/bin. So when i start build.py he call mksquashfs and can't find it, because there is no mksquashfs (without 3 at the end) yet. When we install the squash-tools package from debian he install mksquashfs, that can be used, but is in version 4. Is this not the wrong version? So how... and when build.py start the right one (v3)? I'm not sure that this is the case. So i'm very afraid yet to flash my VTO. Maybe im thinking worng, but if we not make a manual symbolic link to mksquashfs3, he will use the new one? And will brick the VTO? Were i'm wrong. Yes i saw, that here is variable with the Name Version in build.py and that this is added to the command String for mksquashfs at the end. But he calls further only mksquashfs and not mksquashfs3. Or is there other way to start in version 3? I don't understand something. Excuse me.

Hello Serge! Big, very Big thank you!!! Great work! Greetings, Julian P.S. By me, he don't found mkimage, so installed addtitionaly u-boot-tools. Is this correct? And in /usr/loca/bin i found only unsquashfs3, but extract.py need unsquashfs, so i make symbilc link in /usr/local/bin for unsquashfs. The same for mksquashfs3 an mksquashfs. Is it ok? Is it normal, that i become a slightly smaller file - about 718 bytes (without changes in the firmware - only repackaged)?

Hello! Foxi352, can you please, please, please write here a 1-2-3 Instruction how to creata a Firmware for the VTO with and witouth new Sound Files. What is important not to do and what is must. What Package are needed, what environment is to use (Raspberry?) ... I asked, becaus i now how extremely dangerous is by mistake and ... we have already such one case where the VTO is bricked. So a Instruction will be a extremely Hopefully. Do you know how can be recovered brocked VTO? Thank You very Much!

OK, direct IP-connection i have not tested. It's true. But it has nothing with a SIP-Connection to do. Maybe Onvif... Or maybe simple Rtsp... The same from VTH - when you make a Monitoring connection to the VTO it doesn't use the SIP-Protocol. It seems like as he speak with him like with a usual IP-Camera. There is no sound in this case. Only if you make a call (to 8001), then he make a SIP-Call. But i don't know if in this case he switch to SIP-Protocol to send the unlock command or further like a simple HTML-Device. On one side this is very good, because it will be then very easy to simulate the command. And of the other side very bad, that it is so easy, so .... every can make it if he comes in your Network. (the lan-port is not far away on VTO) Greetings P.S. Principaly it is a sick story, that the Pin's for the Unlocker are directly on the backside from VTO! Every cliche from holywood, how the hero is tearing down the outdoor station, make a shortcut and the door is then open.... come true. I think it is better Idea to develop own solution over SIP and Asterisk and Relais on Raspberry, then to use the intern Unlock function. Ok, i'm little bit paranoid. And please excuse me for my bad english.

It is nearly great! Really I followed the recomendation from Allodo and it was a good decision. It has capacitive Touchscreen, not resisitive! The viewing angle is high, not like the cheap LCD's. The relative low Resolution from the VTO (800x480, 720x576) are perfect for this. You don't need more. Only at night is the Picture from VTO a little bit blurred, but this came from VTO self, not from VTH display. At day the Picture is great. Much, much, much better then the Displays from other analog Video-DoorPhones. The keys on the right side are sensory. Principaly you buy a (SIP-)Videophone. For now i can make only in combination with the VTO videocalls. But wit SIP-Clients i can make regular SIP-Calls to the whole world in both (!) directions. There only two things that are little bit disapointing are: 1. with a SIP-Client i can not make Videocall (yet?) 2. the SD-Card hast nearly no meaning with the SIP-Firmware - you can not use the VTH as Mailbox (video and audio) like with the nonSIP-version. With SIP you can use it only to record the person who calls - manually. I don't know, not tested yet, if Asterisk-Mailbox can make record of the videostream too? 3. the loudspeaker ist not sooooo loud. But enough at max. for a normal Room. Specialy for me is the little bit too low microphone sensitivity from VTO a bigger disadvantage. The quality of assembly is very good and robust. There is a Wall-Mounting kit in the package too. I buyed it, because i know how often my Tablet or Smartphone are without Power or in some room somewhere in House. In this case it is much better to have 1-3 real indoorstation where i can accpet the door ringing. For the beginning i buy only 1. But if the project goes further like the last day (german speak files ....) i buy maybe additional 1-2, for every floor. I will see. I have the VTH1550CH which mean white with capactitve touchscreen. But there is, i think, the VTH1510CH too - the same but in black. The Firmware is absolutely the same! Both should be compatible to PoE (802.3af), but i don't tested yet. Maybe this weekend. Greetings

Thank You Pkozul fpr you Configuration. I'm using raspberry too, but without follow-me. Pure asterisk I switched to the new Thread too. My big Porblem was that i not set the realm to asterisk and .... has headatack. I tested after that how the Indoor Station VTH1550 and some Voip-Clients are working with the SIP-Clients and Asterisk. Seems very good. Greetings

Yes, this works. BUT it is not SIP-Compatible. This is the App for the P2P-Push Solution and this works only with the nonSIP-Firmware. Greetings

Hello! Here are my first working settings from my Asterisk and VTO2000A and VTH1550. With help from Allodo. sip.conf: [general] language=de bindport = 5060 bindaddr = 0.0.0.0 ;realm = meindyndnsname ;nat=yes srvlookup=yes ;transport=udp ;localnet=192.168.178.40/255.255.255.0 directmedia=yes videosupport=yes [8001] host=dynamic username=VTO2000A ;it is important that in VTO as RELAM are set asterisk! type=friend secret=admin context=ausgehend canreinvite=yes qualify=yes disallow=all allow=ulaw allow=h264 [8004] videosupport=yes type=friend username=8004 secret=qwertz context=ausgehend dtmfmode=rfc2833 host=dynamic canreinvite=yes qualify=yes nat=yes allow=h264 allow=h263p allow=h263 [9902] type=friend username=9902 secret=admin context=ausgehend dtmfmode=rfc2833 host=dynamic canreinvite=yes qualify=yes allow=ulaw allow=h264 And my extensions.conf: [general] static=yes writeprotect=no [ausgehend] exten => _9901,1,Set(CALLERID(name)=VTO2000A) exten => _9901,n,Dial(Local/alle@tfe-zuhause,50,w) exten => _8001,1,Dial(SIP/8001,50,w) [tfe-zuhause] exten => alle,1,Ringing() exten => alle,n,Dial(SIP/8004&SIP/9902,50,w) ;exten => alle,n,Dial(SIP/9901@videodoorgateway&SIP/8004&SIP/9902,50,w) [default] include => ausgehend I have no Fritz-Fon, so i have not use the Part from Allodo for the fritz.box yet. What works: - Call with Audio and Video from VTO to VTH per Ring-Button. It works bevore accept the Call at the VTH and after that. Only for a short time the Video is gone, when the VTH switch from Monitoring to SIP-Video. - Call from VTH to VTO as Monitor. Whena press the VTO Button on the VTH it connects and show the Video from outside. There are not red cross in the upper right corner. (it is important that set host=dynamic, at the VTH-Did) - Call from VTH to VTO (call on the DID 8001 - SIP-Call) - Audio and Video works. - Call from VTO to SIP-Clients - Bria, Zoiper, Voip by AntiSIP. Wit Audio and Video. But only with BRIA the Video is switsched autmotaicaly on. On the other clients i must set it manually. Not so good. I have not tested Linphone yet. - Call from SIP-Client to VTO works in the same way, when i call the Number 8001. It seems, that the Downloadrate from the Video (800x480) are about 2-2,3Mbps. This is for a usual DSL16 Internetconnection a little bit too much for the limited Upload rate of 1Mbps. There is not known for me way to switch the Video for the SIP-Client from Channel 1 to 2 (lower resolution and frame rate). Only generaly the frame rate can be lowered, but this looks at the VTH indoor Station not good. I hope it helps further. Bye

Hi pkozul! Usualy PoE means the real PoE standarts - 802.3af and 802.3at. Not something proprietary. Everything other is not real PoE. https://en.wikipedia.org/wiki/Power_over_Ethernet Best Regards!

Hello pkozul, can you please post the Asterisk Conf-Files. I and other people are trying the same. Are you using the internal SIP-Server in VTO and Asterisk as Client or the Registrar in Asterisk? I'm trying right now to use the internal SIP-Server, because it works good wit VTH indoor Station. Do you have tested it with a VTH too or only SIP-Clients. But it this case is something very strange - Asterisk registered with 9901 at the VTO, but when i call 9901 or other (pres Ring Button), then the VTO is trying to connect to the Asterisk as Client with username 8001 In other words, if i don#t make something wworng th VTO is trying to resgiter self a the Asterisk for this one call. But i don't know what Password for 8001 is set inside the VTO.... admin and 888888 doesn't work. I can make from asterisk outgoing call to VTO over Number 8001 but i don't become a Picture - only Sound. Per sip show channels i see, that my Linphone Client has activated alaw and h264, but the 9901 line to the VTO use only alaw without h264. It seems he doesn't accept self calls with activetad video?! But why works with VTH and not wih Asterisk and Linphone. Wrong Realm? This is because it would be very kindly if you can post here you settung for asterisk and VTO. Thanks

Hello Allodo and All! Can you say please, was it neccessary to flash you VTO2000 with other Firmware as you write in the first Messages. And if yes, what version do you have and from where did you download? I hava an 7490 but and an Asterisk too. And would to buy me a VTO2000 but only if the SIP conection works. And the last Question - Can you make only Phone Call or VideoCalls works too, for example with Bria or Zoiper on Smartphones? I have experimented with a Gradnstream Camera and its work good, but my big problem is, how to make a good looking frontpanel for it. So i gave up. And the already developed Dahua indoor Station is a second big advantage for me. Thanks! Best Regards!