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  3. 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.
  4. 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
  5. example scenario, I want to activate a new extension 209: - user -> 209 - auth id -> kjh209 - pswd -> uauasyasd6R - recording server -> 192.168.2.33 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
  6. 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.
  7. 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
  8. 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"?
  9. 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.
  10. 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.
  11. 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
  12. 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
  13. 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.
  14. 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
  15. Thanks for the clarification.. if I understood the executable file well, it is "pbxctrl.exe" for windows or "pbxctrl" for other OS. This is therefore the only file that differentiates the installations on the different OS. The "pbxctrl.dat" file is always the same for all installations. Can I ask you what it contains? I imagine the configuration or version of the new system. In case of backup and subsequent restore how should this second file be treated? Suppose we have a backup of the v62 version. Reinstalling Vodia with v63 (for example) which files should be replaced? I ask you to understand what to do in case of manual recovery with the basic installation of a possible subsequent system. I would like to avoid making trouble. Based on your answer I will do tests to verify correct operation. A question comes to my mind. Since the purpose of any system is to make the operations as simple as possible, why not provide a button in the administration section that compresses the entire system folder and allows it to be downloaded? Another step coul be (as other systems do) an automatic backup procedure to a google drive writing the .zip file (or .tar). I think's not so difficult to implement using google API. This would obviously make things easier in the event of a fail. The new function may have the option to include the executable file (which is what distinguishes installations between operating systems) or not. This would make it very easy to move a single installation and complete system restore. Obviously the relative import function should also be included in the administrator panel. In the event of a crash it would be sufficient to reinstall vodia, enter the admin panel and restore the file. I think it would really be a big step forward. Thank You.
  16. Yesterday
  17. hi, I hear an annoying noise in the background on call via Trunk Voip. I dial the external number, press send and there is this noise, then it disappears to the answer of the called. The problem is definitely caused by the Vodia server because the same IP phone and the same network infrastructure with other Voip server the problem does not arise, how can I solve it? Thank you Damiano
  18. hi, I have a problem with the Android App, locally it works well (example https://192.168.1.34) but remotely I can not connect (example https://66.77.88.99) The NAT of the RTP and SIP 5060/5061 ports has been done, in fact a remote extension in SIP works regularly with two-way audio. What other ports should I do the NAT and what measures should I take in the Vodia server for the Android App to connect from the outside? tnx, Damiano
  19. Yes it is actually very simple. Everything is in the working directory of the PBX, with a few exception and notes: If you set absolute paths, e.g. for the recording directory they are obviously not necessarily in that working directory. The configuration file for the fail over itself is usually not in the working directory. Even the executable is part of the working directory. That means if you do a snapshot, it contains the exact version that was running at the time. The file contents are independent from the operating system. You can move a installation from Windows to Linux if you want. Exceptions are the executable itself. For backup that means, you can use e.g. Dropbox or the Windows backup mechanism to automatically make backup off the physical server. Those backups may even be historically, which means if you screw something up you can go back in time and restore a certain state at a given time. The good old file system can be very powerful!
  20. Hi, thank you for your indication. About page (reg_pnpparm.htm) and the field "Snom General" What happens if these parameters are already present in the general parameters. I'll give you an example. I really don't like the call transfer method used in the vodia template. I prefer a more traditional method with simultaneous blind + attended transfer. This should also be the method most appreciated by users (judging by the requests). Using these parameters it is possible to obtain a completely automated transfer that allows you to transfer the call in blind mode (transferring and hanging up) or to transfer the call in attended mode, talk to the colleague and then complete the transfer by hanging up. There is no simpler way than this. To do this, just modify these parameters in all Snom: __________________________ <disable_blind_transfer perm = "RW"> on </disable_blind_transfer> <transfer_on_hangup perm = "RW"> on </transfer_on_hangup> <transfer_on_hangup_non_pots perm = "RW"> on </transfer_on_hangup_non_pots> ____________________________ Some of these parameters are already present in the template of course. By placing them on the page (do m_ext3.htm) in the "Snom general" field does this result in an override of any parameters present in the template? At the moment I have directly modified the template of the single phone but, since parameters are generalized for all snom models, it would be preferable to do it according to your advice. The doubt arises from the fact of not creating conflicts. Finally, I would like to ask you for clarification on the placement of template.xml files within the vodia folder structure. I am not referring to the files generated by the parser but to the original files. Where are the various .xml files related to the various templates placed? I couldn't find them in the folders. Are they present in the system database or cannot be modified? I am used to working intensively with other systems on the modification and customization of the templates. I've been dealing with it for many years and I'm very familiar with all the major brands. I was therefore wondering if it was possible to duplicate some templates and customize them or even insert the templates for the missing phones or adjust those that present problems or are not fully functional. I thank you in advance.
  21. HI, I need to know a couple of information about the backup and the possible restore in a failure case. I'am on vodia v63 on debian 9.9. I know the method to backup individual domains, export them and restore them My doubt arises in the event of a general hardware failure. How can I save the full pbx configuration and then be able to restore (manually too if the case)? By "full" I mean the entire administrator part including the various domains or the single instance (in the case of a single PBX domain). In the linux environment the position of the PBX is in / usr / local / pbx. This is clear. In the Windows environment it is even simpler and the structure is the same. Is it enough to copy the whole "pbx" folder, perform a new installation and overwrite the entire contents? I have not found other solutions. The available documentation only refers to backup of the single domain. But in the event of a total failure it is necessary to perform an integral restore quickly without the need to reconfigure the administrator part. I hope there is a solution for a total backup as it now happens for many other competing pbx solutions that allow you to restore an entire pbx in minutes. Thanks in advance
  22. Yea... another common problem was that the previous default number of connections was just 50, which is very low. You should set that to 500 or even more (/reg_ports.htm). As for the login we have a secondary login page "rawlogin.htm" - you are not the first with that problem!
  23. Last week
  24. Not quite fixed after all. It works for a little while then it renders pages with huge logos and unusual layout, then after trying ctrl-F5 (firefox's shortcut to *fully* reload) it stops responding. I'm seeing these lines at the logs: [6] 20190525003418: Last message repeated 3 times [3] 20190525003418: Current number of requests 50 has reached maximum 50, connections not accepted [6] 20190525003418: 140 more requests pending to acme-v02.api.letsencrypt.org:443 [6] 20190525003419: Last message repeated 2 times [3] 20190525003419: Current number of requests 50 has reached maximum 50, connections not accepted [6] 20190525003420: 140 more requests pending to acme-v02.api.letsencrypt.org:443 [6] 20190525003422: Last message repeated 5 times [3] 20190525003422: Current number of requests 50 has reached maximum 50, connections not accepted [6] 20190525003422: 140 more requests pending to acme-v02.api.letsencrypt.org:443
  25. FIXED! removed webUI customization (pbxwebai and webpages directory) and it works.
  26. OS is linux/CentOS. There is not firewall (wget ...localhost...). The ports _are_ open (notice that wget reports that it gets connected but I also verified with ss). rawlogin has exactly the same behavior (I get connected but get zero bytes from the PBX). I rebooted (it's deep in the night here) with no success.
  27. Hi, Please find the link here: https://doc.vodia.com/starcode_anonymous_calls
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