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  2. 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
  3. 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
  4. 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!
  5. 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 = "PERMISSIONFLAG"> on </disable_blind_transfer> <transfer_on_hangup perm = "on"> VALIDVALUE </transfer_on_hangup> <transfer_on_hangup_non_pots perm = "on"> VALIDVALUE </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.
  6. 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 = "PERMISSIONFLAG"> on </disable_blind_transfer> <transfer_on_hangup perm = "on"> VALIDVALUE </transfer_on_hangup> <transfer_on_hangup_non_pots perm = "on"> VALIDVALUE </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.
  7. 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
  8. 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!
  9. Yesterday
  10. 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
  11. FIXED! removed webUI customization (pbxwebai and webpages directory) and it works.
  12. 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.
  13. Hi, Please find the link here: https://doc.vodia.com/starcode_anonymous_calls
  14. After you change the domain provisioning password, you need to re-provision the phones. You can do this through the web interface, in the extension list.
  15. nothing to do, the advice you gave me does not work, now with the pswd inserted within the phone via web but as a user and not as an admin, so you can not do anything
  16. you want to tell me that if I enter in the fields "Authentication password" and repeat a pswd I want (example 34567@#) for all extensions will be accessed via web with admin/34567@# ?
  17. If you want to log in to the web interface of the phone, you need to set and use the domain provisioning password (/dom_settings.htm). By default, this is a random password - yes it is not easy to guess
  18. Alright we'll fix this. Thanks
  19. I wanted to report a bug in the starcode link indicated in: /dom_feature_codes.htm https://doc.vodia.com/starcodes_anonymouscalls
  20. I'll correct myself, The provisioning also following the online doc works partially, you must enter via web in the Grandstream phone and press the button "Start" to do the provisioning from Vodia. So you have to launch the command from the phone, but it becomes impossible because you can not enter via the web on the phone as it seems that it is impossible to trace the pswd (so I was told by you in another post).
  21. OK but you need to intervene manually via the web, enter the phone and make the appropriate customizations, as I am for the templates are not complete, if it is not possible after provisioning to make changes via the web in the phone everything becomes impractical. So I need to understand if it is possible to somehow enter via the web in the phone or it is impossible. I am testing the server, I am an installer and the solution of an inaccessible phone I do not think it is acceptable for any installer and no situation. So my question is the same, how do I access a phone via the web after provisioning?
  22. Looks to me like the PBX could not open the sockets for HTTP/HTTPS. Maybe the firewall does not like the new executable? What OS is that? Check with netstat is the ports were opened.
  23. Every phone has a different place to enter the SIP password for an account. Generally I would recommend to use the automatic provisioning for VoIP phones. There are many other settings that should be set, it is very difficult to do that manually. And the PBX is automatically assigning a password to the VoIP phone which is different from the SIP password (every extension has many passwords).
  24. Can you try with http://localhost/rawlogin
  25. I understand that it is essential to activate the DMZ etc. ... on the firewall with the IP address of the server Vodia. I'll try and update you tnx Damiano
  26. Hi, Maybe this will help you understnad better: https://doc.vodia.com/admin_sip_and_audio Look (Ctrl + F) for the word "SIP IP Replacement List" on here.
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