Vodia PBX

That is already 4 years old - in Microsoft country a lot happened since then. Any chance to upgrade?

Hmm. I know that a few folks got the Speech Server working, however I beliebe that the 2004 version might be a little bit outdate (but I am an the big Microsoft expert here).

Did you already check out http://wiki.pbxnsip.com/index.php/Linking_..._to_an_IVR_Node? Did it help?

No. The service must (and does) run without reboot. It is a PBX service, rebooting it every day is not an option. If you need to restart the service because of one-way audio the service is still there, but cannot open media connection. This might be because there are no more ports available (check your RTP port range). The reason can be a lack of resources like ports, but it can also be a firewall that has (for example) too many TCP connections that need to be closed.

Having this feature on a hunt group is more difficult than for a single call. It simply does not exist. So far we never thought about it... Usually the hunt group is used for a central number and there (so far) it is not common to ask for the name. But you are right, as more and more automatic machines are war-dialling cheap lines these days, it is becoming a more important feature.

Oh yea this topic is a real pain. Whoever invented SRTP did not think about concersations that last longer than 22 minutes - and that calls are possibly on hold for more than 22 minutes. In 3.0 we introduce a new way of guessing the rollover counter. snom's 7.1.33 also has some improvements with SRTP. We all cross fingers that this will be the last time we talk about this topic.

I believe so...

Do you see the information that you are looking for in the To-Header? Check the SIP INVITE that goes to your phone. Maybe it includes already the "To" header. Then the next step it to think about getting that into the From-Header...

The problem with the address book matching is that in many cases when the call comes in it is not clear where it exactly goes (at least at the time when the address book query happens). For example, if the call goes to the auto attendant then at that time it is impossible to use the user address book yet. But your case sounds more easy. How does the call come in? Sounds like there is a problem finding the right domain.

Yes. To be more precise, the GSM 06.10 FR codec (there are also others). The codec number is "3" or just "gsm".

Whow. Restart of the server or just the service? If you also need to restart the server then check if there is some virus or malicious program consuming a lot of resources (memory, handles, ...).

What is a PAC extension... ? Extension is extension...

The 2.1 branch does not support the provisioning of the buttons of the 300 phones. Maybe it is time to "officiallly" start the 3.0 beta runs, because 3.0 fixes these little known problems. Alternatively, you can set up your own fkey provisioning file. Take a look at the generated directory, and then you can just move these files into the tftp directory and edit them there locally.

You can't... Workaround if you don't like the emails is to set up a email rule to send them to the trash can. I guess we need to introduce a setting that says what emails should be send which emails not.

No, not yet... You will have to live with a beta image. Send me a PM if you want a link (and what OS).

You are right... That feature exists only in 3.0. If it is really important to you, then you should consider upgrading.

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.