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hsmptg

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  1. 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
  2. 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!
  3. 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!
  4. 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?
  5. Any reason to use General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.1.R.20180606 and not the newest one? Well, it seems that I "need" V4 in order to disable the english audio! Even with a V3 with portuguese audio messages, it is nonSIP which would not allow the integration with RasPBX. Anyhow, where could I get the "V3 nonSIP firmware with portuguese audio messages" to test? Regards
  6. After flashing General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin firmware I can still flash v3 firmware back if it does not work properly with RasPBX? When you say "non SIP", you mean a firmware that just work with the VTH monitors and Dahua apps (gDMSS), right? The General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin firmware is the one you suggest within version 4? From what I read it still does not have Portuguese language (audio messages I mean) but at least I can disable its use, right? Regards
  7. I just found the problem! I was using copy/paste to copy the FreePBX SIP password to the VTO2000A web interface, and did not realised that it only accepts 15 characters when FreePBX automatically generates passwords with 32 characters. Since the VTO2000A web interface was truncating the password it was therefore incorrect! Using a shorter password, all started working correctly. Next fight: the language used by the audio of the VTO2000A I'm from Portugal and Portuguese is not supported. It seems that version 4 allow at least to turn off the sentences in the begin/end of a call. But since version 4 is not compatible with asterisk that is not a solution for me. Any other suggestion? Regards
  8. Hi rickwookie Thanks for your suggestions! Regarding the use of V4, I was a bit scared of flashing it since I read some posts reporting some bricking problems after doing that. But since you have this opinion I thing I will try it. It is General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.zip, right? But my initial question remains: What is the use of the other bin files besides General_VTOXXX_Eng_P_16M_SIP_V4.000.0000000.5.R.20181030.bin? About my access to the Dahua firmware site, well I just used an username/password that I found in ipcamtalk forum. Regards
  9. Hi I tried several things to solve the missing text on the webpages, and one of those things was in fact "clearing browser cache". So I did not had any certain about what had solve the problem. But since you and @rickwookie both point that issue, it seems that is really the way to manage it. It is curious that you are just using Asterisk from cli since that was my initial idea. I would prefer just to install Asterisk in a clean Raspbian Stretch Lite image instead of having to use a RasPBX image. But for these initial tests it seems to be easier to use RasPBX. My idea is to build with a Raspberry Pi plus a 7" display a SIP client equivalent to a VTH1550CH to connect to my VTO2000A. That way I could have in the same equipment a Video Intercom and my home automation (home assistant) terminal. Regards
  10. Hi riogrande75 I got it from https://share.dahuasecurity.com/ from folder: 04 Product Firmware > Intelligent Building > SIP firmware By the way, I'm trying now to setup my VTO2000A with this firmware in order to link it to my RasPBX. Some of the VTO2000A webpages are a bit different compared with some of the screenshots I find here. Here are my actual settings that still do not work. As you can see my test setup have a Windows MicroSIP client (extension 1000), an Android CSipSimple App (extension 2000) and VTO2000A (extension 8001 that I accepted as default and set as PJSIP without any certain). MicroSIP and CSipSimple already can connect with each other but not with the VTO2000A. When I hit the call button of the VTO2000A, what extension it would call? The 9901? Can I make a call from any of my other SIP clients to the VTO2000A? Calling 8001, right? When I do this I get an error (503 Service Unavailable) which is different of the one I get if I use a non existing extension, so it seems that the VTO2000A is somehow recognised. Regards and thanks for any help you could give me.
  11. Hi Inside the firmware zip file there are 3 bin files, for instance General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.zip has: - General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.bin - General_VTOXXX_Eng_P_16M_SIP_V3.300.0012001.0.T.20181229.language.bin - Upall_VTOXXX.20181229.bin After using the first one within VDPConfig app, the upgrade was successful but the VTO2000A webpage was strange (without any text as can be seen in the attached image). After many resets (including factory resets) the problem remained. I believe that was just when I also upgraded the second file (the language.bin) the webpage started to appear normal. In a future upgrade I must always upgrade both files? What about the third one (Upall_VTOXXX.20181229.bin)? What is its use? Regards
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