Vodia PBX

*00 is used for speed dial a extension cell phone. In the old times, star codes started around *60, but as we have added more features those codes are now in the *5x area. Where is that text with the 00 to 60?

Well the problem exists for practically all snom models... A batter solution might be to use the snom-general parameter instead of editing the template and just use that general parameter to set the buttons. Whow the snom phones don't allow spaces after the key name? We'll add something that will strip the spaces from regular parameters. This is something very hard to find!

There is also a "action" drop down...

You can enter anything that starts with *. If you are overlapping with another star code, then that star code will become unavailable. Typical star code ranges are *01 .. *49 where there is no overlap with any other star code.

This will add an entry to the address book - you should be able to see that in the user's address book.

The call ring duration is a global setting for the whole system - it cannot be set on domain or system level. But there are many timeout settings e.g. per group or extension, for example when the mailbox should pick up or a hunt group should redirect a call.

Yes "Super" > "System"...

If you restart from the Windows service, that is probably the fastest way to get this done. If you have no access to the Windows service manager, you can as well just terminate the process from the web interface - then the windows manager will automatically restart it. Windows is very good dealing with services that suddenly stop . This is in reg_reboot.htm:

You will need to enter something like "123" in the PIN field and then delete it. This will tell the JavaScript that the field has changed and will save it. It is a little bit of a workaround...

After saving the FTP port, can you check if the pbx.xml file contains the empty port (something like <ip_ftp_port/>)? It works well on our test system...

The minimum recording time setting is in /dom_mailbox.htm. By default the minimum recording duration is 1 second. This time starts when the recording is turned on for the voicemail message. The ring duration is not related to that recording duration - this is just about the time how long a phone will be ringing up the user. IMHO it is okay to make it a lot longer, as long as the system will eventually stop the ringing (e.g. after 5 minutes). What you don't want is that the phones are ringing the whole weekend...

Once the user has a PIN they can change it (for example, in the mailbox) but they cannot delete it. This must be done by the administrator. Maybe it is possible for the user to delete the PIN from the user web interface - not sure about that.

There are two levels of administrators - the "super" administrator and the "system" administrators. The super administrator can set up system administrators. This is useful when you work in a team with other people, and they each are able to perform system administration with their own login and own password. For example if a administrator leaves, then the super administrator can just delete the account and all is good.

Well maybe those other vendors have a problem with their WebRTC subsystem

It is a little bit better to put a file there with digital noise, so that the PBX does not try to read the file again and again.

Can you check if the FTP port setting in the reg_ports.htm is really empty? If yes, please run a netstat -anp to check which process is opening that socket - maybe it is the standard FTP operating system service that sits on that port...

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.

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.

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.

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...

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!

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.

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.

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"?

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.