Vodia PBX

Well, you can use a agent group with only the receptionist as agent to queue callers up. That could help solving the problem.

Unfortunately, we cannot change the chips with a software update. But at least it would be good to know what exactly is the problem. I know from other cases that it might help to set the link speed to 10 MBit/s, which is a lot less critical than 100 MBit/s.

Also, there was a common problem with SPA phones that they did Intercom even when you were in a conversation. The intercom put the call on hold and started the playout immediately.

Well, if you want to perform an attended transfer you need the 2nd line. One call on hold (which counts!) and the other one for the consulting call. Why do you want to turn call waiting off?

Well, at the moment it is a feature. When you use the auto attendant that AA fixes the outside's callers inability to directly call and extension and it shows the call went right to the extension. For other calls like hunt groups, it should which number was dialled. In the CDR record you have more information so you can figure out where the call actually went. The alternative is that the auto attendant does not change headers. But then the headers would usually only show that the call went to a DID number of the company; IMHO that would be worse than changing the headers. BTW there is a flag in the domain settings that tells the PBX not to change the headers, did you try that out?

Well, we had the discussion about the park accounts and in the early versions we even had park accounts. The way it is now is that every extension has a park orbit associated with the orbit. We did a little fine tuning and now the mailbox has a setting that tells the PBX it should really prefer the park functionality. On the Cisco phones we were able to program a button to act as park/pickup button, where you have to enter the park orbit via DTMF. That seems to be a usable solution, without having to use star codes. The button is essentially a speed dial and that should also work with other phone types. I agree. People coming from traditional phone systems just don't understand what we mean by "hold" and "park". On the traditional PBX, they always parked the call. Having more than one call on an extension was simply not possible (only one cable). This is coming back with more and more cell phones being used as PBX extension. Lets see where this leads us...

Set the outbound proxy on the phone to the PBX, no matter to what domain. And use the domain in the identity setting to point to the right domain. That should work. But be aware that multiple accounts also means that the customer must pay close attention which one he is using. Things may get confusing very quickly. We found that it is in such a scenarion easier to get two phones, and paint one red the other green to make sure which phone he is answering. Well, keep in mind if you don't have a stable internet address you are building your system on sand. STUN and all that dirty workarounds cause a lot of grief in support and with the customers; dyndns might help to deal with the changing IP addresses but all this is a very instable and dirty workaround. Better get a IP address that is not changing. If you can, get an IPv6 address!

Everyone does it differently. There is no standard. Some even use * to enter or terminate information. I think it is a good idea to use one key (e.g. #) consistently in all dialogues as the "terminate", "enter", "execute" key and another one (e.g. *) as the "clear", "correction", "cancel" key. That is what pbxnsip tries to do.

Ideas: The outbound trunk uses CO lines and there are only two lines available The license allows only two calls The PSTN gateway can only deal with two ports (e.g. has only two FXO lines...) The log should have something that gives more hints what is going on. Regarding the phones you might have to factory reset them and provision them again, as the dial plan "sticks" to the phone once you assigned one.

"500 Line Unavailable" means the trunk uses CO-lines (see http://wiki.pbxnsip.com/index.php/CO_Lines) and all lines are in use.

We have it somewhere on the to-do list...

Strange. I assume that the system time zone is the same as the domain time zone? Also, give the system a couple of minutes, because the flag states are not calculated every second.

Send it to support@pbxnsip.com.

Yea, agreed. We need something that detects "#" as hold and then you can start entering a new number.

I believe it is a mistake to have users enter a "9" at the beginning. It screws up the address book and if you want to call someone back you always have trouble putting the "9" at the beginning. Better put a "1" at the beginning and use the 11-digit dialling in the USA. THe only drawback of this is that you must be careful with extensions--they should not start with "1"...

There are several ways to do this. The first way is to use two servers that use rsync to replicate the data from the active to the standby and which use a "virtual" IP address. That seems to be fine for cases where the failover can take a minute or two. The other way is to run the PBX in a virtual machine and let the virtualization software take care about the redundancy. This can failover (depending on the VM) within a few ms and you would just hear a short click when it fails over. Needless to say, the first one can be done with two Linux machines relatively easy. Kevin can get you the necessary shell scripts that do the job. The second one requires some serious investments into the virtualization environment. Plus you should run only one VM on a physical CPU in order to avoid jitter.

What operating system? Did you check with netstat which ports are open?

If we use a hunt group or a paging group for that it will be no biggie. But currently we don't have it.

I would do it the following way: 1. Before the installation, use netstat to see which ports are already taken. Usual suspects are 80, 443 but maybe also other ports like 5060, 5061 or 69. 2. Then perform the installation. If the port 80 or 443 is blocked already, just manually edit the pbx.xml file in the installation directory and change the ports to something available. 3. Then restart the PBX (using the service manager), log into the web interface and change the other ports. 4. Restart with the service manager again. Then everything should be up & running. Needless to say, if you don't use standard ports, you must specify the port numbers in the HTTP and SIP URI.

It looks like the firewall closes TCP connections after 60 seconds. Try making the "Maximum Registration Time (s)" shorter, for example 50 seconds.

There is nothing special about the CS410 regarding NAT... It can deal with two IP addresses, one public and one private if you want or two public. It is just a Linux computer with two Ethernet interfaces. http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses fully applies. Oh yes, the "old" CS410 had only one Ethernet interface. Yes, the new one has two now.

You need to convert it into 8 kHz/s sampling rate, WAV file, mono, linear (16 bit/s) equal to 128 kbit/s music. Then you can use it as a file for MoH. There are a lot of good audio tools available that can do that. In the Linux world I believe sox does the job; in Windows I know about GoldWave as a nice tool for editing audio files.

You will see a SIP message (typically a INVITE) coming from the ATA into the PBX. If that message that does not appear, check the configuration of the ATA, maybe you need to turn T.38 on. There is no setting on the PBX for that, everything happens automatically. T.38 is not a codec, at least not for the PBX. It is like video, gaming or whatever you want to send through an established session on the PBX. The PBX does not even know it's FAX. BTW that's why you can also turn on a camera on - once you have an audio connection established.

No, so far there is only one flat log file per system. This is only accessible to the system admin. The idea behind is that the Joe Average domain admin should not bother about topics that require logging. Maybe once nice day we'll change that; but it will be bad for memory and there are many cases where the domain is not clear...

Once the call is initiated the PBX will think that it was the extension who initiated the call. If not, it will think this was just a call to the calling card account. This is a little confusing, yes... In the next major version we will split the CDR into trunk CDR and account CDR. The trunk CDR is about what who has to pay to the carrier; the account CDR is about what user made what call and how long a call was on hold and so on. I believe we will not be able to address all questions with one CDR type only.