Vodia PBX

Totally agree. There is not right now, but next version will have it.

So far the only exceptional problem we can think of is that the logging to the file system slows down the process. E.g. when the file system is slow for whatever reason, the logging would block the RTP-specific thread. The workaround is to turn file system logging only on when you are actively debugging something.

Hmm. Maybe you can post the XML documents that the PBX sends to the phone when the state changes (the body of the NOFIFY messages). Then we can see if the phone should believe that the resource is "active".

All configuration is stored in the file system. Usually a power failure does not wipe out the file system. Check if there is still data in the working directory and copy it somewhere. Another reason to make back up back up back up. If you really have to set up everything again, and everyone is happy again, then make a back up! Next time you don't have to do this again... Maybe someone who is a Windows guru can post a trick on how to SIP the working directory every night at three o'clock and move it to some safe place in the network. That would make many people's life a lot easier!

Ehhhh.... Okay, maybe temporarily turn Javascript off. Or you can also use a handcraftet URL like this: http://yourpbx/dom_feature_codes.htm?editd...gent_logout=*75.

I believe this is because the plug and play mechanism requires a password. Either change the TFTP password generation or set the password on the phone (see http://wiki.pbxnsip.com/index.php/Snom).

Indeed, that would be tricky. What I would do is take the files polycom_phone.xml from the generated directory, move it into the tftp directory and edit it there so that you have your second registration. Possibly you also need to check the other file, polycom_sip.xml for stuff like outbound proxy. I believe editing the 2nd line in the web interface of the phone will not work. You'll probably loose the information on the next reboot.

Yes. Though ERE are not very simple (you need to escape the * as it is a special symbol, \*).

Siemens Gigaset IP DECT devices are an alternative. Also, Polycom and Aastra have DECT devices - I believe those more targeted at large enterprises that require handover from one DECT base station to another. If you use FXS ATA, you have a huge choice of cordless devices; however that audio quality may be a problem because of the additional A/D step and stupid things like hangup detection.

The code for the login and logout may be the same. If that is the case, then calling that code will toggle the state. Maybe that helps?

1. Yes. The PBX can deal with any number of NIC (and IP addresses) - as long as the routing tables are set up correctly on OS level. 2. That can be done only on extension level. There is a feature called "Call forward when not registered" that you can use. Doing that on the whole domain has only limited use, as it is difficult to say when the whole domain is "offline". For hunt and agent groups you have to work with the final stage (hunt group) and all agents logged out (agent group) or just a time redirection.

Backup & upgrade...

Good point to put then at the front, Bill! We'll do that. Pre-logout sounds too difficult to me. I guess you'd have to define a preference in a new setting for that.

Well, the list of agents is stored in the agents XML. The DND state is stored in the extension XML. There are a few scripts available at http://wiki.pbxnsip.com/index.php/Shell_Scripts; it is for Linux but you should get the picture and this can also be done from Visual Basic.

Programmatically: You mean, for example, by looking to the file system or using SOAP requests?

I would just take 0 (ulaw).

Try forcing a specific codec on the trunk. Probably the provider has a problem when the PBX answeres with more than one codec (see http://wiki.pbxnsip.com/index.php/One-way_Audio).

Well, it was more about the XCAP specification and examples. Anyway, I guess we can also get those documents.

Looking at the old messages I must admit I don't exactly get what the problem is. Is it a problem related to RTP or is it a problem related to the phone number (+47 or 0047 or 01147 and so on)? Maybe you can get a fresh LOG...

Try ths: http://pbxnsip.com/protect/pbxctrl-rhes4-3.2.0.3132.

I guess we should just investigate why the auto attendant does not send the call to the Exchange...

Good point. The PBX sends the following content to the phone: Messages-Waiting: yes Message-Account: sip:840@pbx.company.com Voice-Message: 1/0 (0/0) The "Message-Account" tells the phone where to call, and that destination includes the "8" prefix that should send the call to Exchange. However, I just verified that the Polycom phone sends the call actually to sip:40@pbx.company.com. This is a configuration problem with the phones. The msg.mwi.x.callBackMode is set to "register", we have to check if there is a way to make the phone the provided account in the MWI attachment. If we want to provide the "contact" for call back, we'll have to change the PBX code.

Alias scale very well as the table in the PBX has a index (performance). Using the "trunk sends call to ..." option in the trunk is not an option if you want to enumerate all possible destinations there (a few ERE would be fine), the PBX has to step through that list for every incoming call. I would get a good text editor to generate the list of alias names. Simple as it sounds, it can save you a lot of time editing the numbers and getting them into one line. "Emacs" is a science, but maybe there are also other editors that have some macro functionality that can automate editing.

Well, the bulk import feature does exist - check out http://wiki.pbxnsip.com/index.php/Creating_New_Accounts. You can use Excel to create the list. Before you can put that into the input form, you need to export it as CSV. Then you can easily copy and paste that into the form on the web. There is still some "manual" work required; however significantly less than punching the information in one by one. Plus you have the flexibility to import also settings like cell phone numbers, MAC addresses.

Changing from 9 to 32 entries must be done on the PBX. We did that already for you, the next software version will show the first 32 matches. Yes, to get SUPERFOSS you would press 7 (wait for the XML), then 8 (wait for the XML) and then if you still don't see SUPERFOSS press 7 (wait for the XML), then if you still don't see SUPERFOSS press 3 (wait for the XML), then if you still don't see SUPERFOSS press 7 (wait for the XML), then if you still don't see SUPERFOSS press 3 (wait for the XML), then if you still don't see SUPERFOSS press 6 (wait for the XML), and so on. At any time you can use the up/down key if you see SUPERFOSS and select the entry. Maybe your cell phone has something called "T9" entry, that is similar. When using the XML on the PBX, you must wait for the answer before you can enter another digit; that is because of the link-nature of the XML documents.