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  1. Yesterday
  2. Okay solved. I don't understand why locally I have one-way audio. Windows firewall disabled, I call locally from webrtc to a local phone and I have one-way audio. n.b.: the phone has two-way audio if I call other extensions or the trunk, it has one-way audio only if I call the extension in webrtc. Damiano updating: I have now tried locally between Android app and SIP phone and audio is unidirectional. So the problem with way forward audio is only if you use pure webrtc.
  3. absolutely not, "deleting" the file noise.wav the noise disappears completely, I replaced that file with another mute with the same name and now the audio is perfect. Can I also completely delete the noise.wav file or better if there is (even if with other audio)? Damiano
  4. other vendors give the possibility to block the part of WebRTC for security. tnx Damiano
  5. ok tnx Damiano
  6. In essence it is not possible to manually set the different authentication. tnx Damiano
  7. You need to make sure that the app can access the PBX over HTTPS - so the rules for operating the PBX behind NAT also apply (https://doc.vodia.com/server_behind_nat). The App does not need port the SIP ports (it uses HTTP/HTTPS instead), so you must make sure that you also port-forward the HTTP/HTTPS ports.
  8. We do that for VoIP phones. The auth-ID is then the MAC address and the password is a random string generated by the PBX - but there is no way to set that password from the web interface. You could use curl to pretend that you are a VoIP phone that pull's its configuration and then filter out the MAC and the password from that provisioning file.
  9. If you don't want to use TFTP or FTP just clear the port setting -then no port will be opened. This is perfectly fine, especially if you are not operating the PBX in the LAN. On WAN these ports are not needed. SSH is available only for Vodia IOP and IO - on a regular server installation that setting is not there and it does not make sense because the PBX has no control over SSH anyway.
  10. There is no "service" inside the PBX for WebRTC - but what you could do is disable the user login (not giving users their web access password). What security reason would there be? WebRTC is essentially based on TLS...
  11. What you can also do is generate a PCAP on the trunk level - then listen to it from the Wireshark RTP analysis tool. Maybe the noise comes from the trunk!
  12. hi, I wanted to know if it is possible to stop the WebRTC service, in some cases it may be preferable to disable it, especially for security reasons. Is it possible to stop the WebRTC service? tnx, Damiano
  13. hi, I wanted to know how to enable/disable SSH - FTP - TFTP services, in reference to SSH the manual reports the following: SSH login: When SSH gets enabled, the PBX writes the SSH configuration files. Depending on the SSH version, this could work or not. Hopefully this works now for all versions in 63.0. as for FTP and TFTP I see how to change the port, but not how to disable services, also for SSH - FTP - TFTP access credentials are those of the administrator? tnx, Damiano
  14. If parameters occur twice in the config file they should be overwritten. For snom phones the last appearance is the one that "wins" for Yealink the first appearance wins. What is nice about those parameters is that the next upgrade of the template will not break anything. The problem is if the parameter contains a "syntax error" (e.g. forgotten / in XML) well then your whole template breaks. With power comes danger. Parameters are also available on domain level. There is actually nothing speaking against making them available on extension level, something we probably have to look into in one of the next versions. The templates are declared in the various pnp_xxx.xml files. For example the snom parameters are declared in the pnp_snom.xml file. You can if you want declare more variables there and use them in your changed template. However when you change the template you will miss the updates on those templates in the next version, kind of defeating the purpose of having templates in the first place. IMHO It is a much better solution to use the general parameter. If there are useful parameters for a vendors, better let us know what makes sense and we'll add that to the next version.
  15. really we're talking about a noise problem, it's not about the history of ISDN or any other thing, the same problem I repeat there is not if I use another server, so it's a problem that creates Vodia, maybe a wrong setting. tnx Damiano
  16. example scenario, I want to activate a new extension 209: - user -> 209 - auth id -> kjh209 - pswd -> uauasyasd6R - recording server -> I can enter all parameters but I don't know where to enter auth id. For security reasons I normally impose a different auth id on each extension. But on Vodia I don't see where I can set it. tnx, Damiano
  17. Well, this is called "comfort noise" (see e.g. https://en.wikipedia.org/wiki/Comfort_noise). The story goes back to the introduction of ISDN where users were confused by the digital silence they thought the system was dead and hung up while the call establishment was still in progress. This can be something like 10 seconds for example when calling a mobile phone or even longer if your SIP trunk provider has to first hack a system to route this call to Cuba. Remember, ringback means that the other end really rings - on other words there is a human that is getting alerted. So there are two phased when calling a person - the network is trying to locate the device and the person is being alerted. How long the comfort noise should be is of course debatable. In loud environments you want it louder , in quiet environments you want it less loud. If you want to experiment with it, the file is in audio_moh/noise.wav - if you like you can make it digital silence like in the beginning of the ISDN age or tone it down, or even make it an advertisement for your stellar hosted PBX service. If it is a very loud ear deafening noise that does not change after connecting the call you might have a old firmware version on your phone that does not device SRTP properly. In that case the phone would receive SRTP packets but play them back as regular RTP packets.
  18. well, I used to get there, the strange thing is that I close the browser, maybe I open it after 3600 seconds and I enter in Vodia without being asked again the credentials, I update you. Damiano
  19. You mean you want to use a name for the authentication that is different from the account name? For example if you account is 40@domain.com, your user name would be 40, but you authentication name would be "something-else"?
  20. Every time you do something the session gets refreshed. If you are in a admin page, the PBX will refresh the registration and call grapho every six minutes - which will also refresh the session.
  21. The pbxctrl.dat contains the templates for the web server, emails, image and configuration files like the ringback.xml. Essentially this is everything not generated by the compiler. We started compressing it some versions ago because the Vodia IO does not have too much hard drive space. It should be the same version like the pbx executable; just treat it like a DLL. We had that button for downloading everything in older versions - however it turned out to be a support problem because systems grow and when you download a Gigabyte, it crashes the browser and it also crashed the PBX web server. Especially restoring backups was a major problem. That is where file system is a lot easier and faster.
  22. hi, I noticed that although I set the parameter "Web Session Timeout(s)" to 3600 seconds the session is not interrupted after this time. If for example I access the system after 4 hours the system does not ask me to insert user/pswd again, Where am I wrong? Thank you, Damiano
  23. hi, wanting to manually register a SIP extension you can enter user, password and registration server, but I do not see if it is possible and how to enter the authentication (very important for security reasons). Is it possible to do this? thank you Damiano
  24. Thanks that was it! I had to temporarily firewall one IP with a lot of TCP connections in order to let me connect to the webUI.
  25. This is the same for me. WIth Snom phone when you dial a call during invite session or 100 or 180 answers, before the UDP session start, there's a loud white noise. This noise is introduced, in some way by PBX. It's not a confort noise function because this should be activated only after the session RTP start. Snom, in any case send only mute RTP packet in order to maintain up the UDP session. I Check in the phone settings of the provisioned phone and there are no option that ca be cause this behavior. I ask because the noise is very high and leads the user to think that there is a system malfunction. Obviously this noise completely disappears when the RTP audio session is started: at this point the audio is perfect. This is absolutely independent from the phone or the SIP trunk. Tx
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