Vodia PBX

Just set up a MoH that uses the "ringback.wav" as MoH file. Then you can assign that "music" to the agent group. The PBX will think it is music, but the caller will think it is ringback tone.

It also works on the Polycom - however you need to provision then through the PBX. Otherwise the setup of the configuration files will be really difficult.

The only thing that comes to my mind is that the server is "local", because the PBX does not want that the SIP phone tries to get a routable IP address (through STUN or other more-or-less buggy methods).

Are you using outbound proxy? I recommend to always set the outbound proxy, unless you really want to call SIP URI in peer to peer mode.

That is a problem with one of the web interface files. Not very serious. It means that the web server tried to load something from a table that does not exist (is was acds). We need to update the file and fix that in 2.1.12.

IMHO the packet scheduler makes only sense if the traffic that leaves the computer can exceed the speed limit of the NIC. For example if the NIC is connected to a T1 and you run both voice and data on it, well then a packet scheduler is really useful. If you have a 100 MBit NIC to the LAN and the computer is not a busy file server well then there it does not make sense to me.

You can put the files into the tftp directory. Then you can access them with http://<ip-adr>/tftp/filename. No need to run IIS for that.

Remember that you can also use two (or more) tel:alias in one account. The first one will be the one that is used as ANI, but the second one is also used for inbound matching. Maybe that helps to solve the problem.

Ehh.... Okay, that version does not have this. 2.1.12 will have it . Then you can add the parameter "connect=true" and it will start dialling when the handset it lifted up.

Nope. Does it make a difference if the contact type is white or black or just transparent?

Hmm. If the registration is fine, then we can exclude wrong password or ISP blocking SIP ports. Are you sure that the account has enough money (if prepaid) or credit?

The alternative to the AA is to use the calling card account, where you can log in (actually not only from the cell phone) and place outbound calls from your account.

We heared about Firefox issues, but we were also not able to reproduce them. Can you get us a Wireshark trace (send me a private message to discuss the location)? Then we maybe can solve that riddle.

In 2.1.10 you can do that by using the tel:alias or a per-account basis, this has higher priority than the DID number setting in the trunk. This solution is okay unless you have two extensions that should both show the same DID which is not the same as the DID for the trunk (that will be fixed in 3.0).

What version are you on? I remember we took the "1" out...

Well, then that's the problem: REGISTER sip:192.168.41.223:5060 SIP/2.0. "domain.local" != "192.168.41.223" and "server.domain.local" != "192.168.41.223". Try "localhost" as alias. The string "localhost" is a magic name that matches any domain name: "localhost" == "192.168.41.223"!

Maybe you can just use the speed dial mode for a button. Not sure if you then press hold on the same button then.

In the 3.0 release we (will) have a ANI settings that should solve many of these problems.

Very strange. Do you have an alias name of "localhost" for the domain?

Well, then it will be very difficult - the PBX knows only at the end of the DTMF sequence that the code is supposed to be intercepted. I would prefer to have a option on the PAC that turns recording on and off from a user.

Apart from the usual resource limitations (number of calls etc), the number of pending transactions is limited to "1". The PBX will send out one SOAP request and wait for the answer. That means the processing of the key input will depend on the speed of the database. But I believe unless you have a absolutely crazy database the response times should be still well below one second, which should be okay for a caller. http://wiki.pbxnsip.com/index.php/Linking_..._to_an_IVR_Node shows how to do this with PHP, I guess you have already seen it. The mySQL part is missing, though. But there is plenty of resources available on this topic.

I would solve this problem by a email-to-SMS mechanism. Most cell phones even support email, which would make things even easier - just attach the voicemail and then those support guys can listen to the VM in alomost real-time in a very convenient way. The problem with calling the cellphones at the same time would be that practically most installations would run out of PSTN channels when the PBX starts "blasting" out calls. In order to solve this, those calls would have to serialized, and that is not so easy.

I would solve that problem by setting a very long timeout (like 999 seconds) on stage 3. There you can list all extensions that should ring "forever". Looping is not a good idea (though it should be possible!), the PBX has a hard time detecting and fighting such loops. Imagine that all extensions are not registered, and then the stage duration is very short (like 0 seconds), and then such a loop will be an "endless" loop. After a reboot, typically all extensions are not registered for a few seconds, and then if a call comes in you have an endless loop. The night mode should be straight-forward. Don't forget to put a 8 in front of the mailbox number, so that it does not ring the extension first.

We solve that problem by changing the the index. Instead of "1-25 26-50 51-75" and so on we'll put the actual account there, so that you can immediately click on the right link, for example "401-425 426-456 457-co9".

Will also be fixed in 2.1.11... It was indeed pretty "open".