Vodia PBX

I would not use CO-lines for that. You can just assign the DID to the extension or hunt group (as alias) and then monitor the extension or hunt group.

At least the PBX supports T.38 pass-through. There are several devices that you can use for FAX, e.g. ATA with real paper FAX attached to it. Or you can use fax2email SW, there are some vendors out there who support that. Even Microsoft Exchange hat a feature that could convert FAX to email. Maybe in the future we will include the Fax2Email topic in the PBX and make life easier, when there is no paper required.

The one we used was the GuruPlug from Globalscale.

Good point. Right now there are some limits especially with the 3rd party registrations (e.g. 5 for snom ONE free) and transfers. There are some signs that especially the transfer discrimination will be lifted soon as it really does not make sense to keep e.g. Android phones from transferring calls.

Feature comparisons are difficult. Both products have a longlist of powerful features. Cell phone IVR support might be better with snom ONE(especially with the latest), 3CX scores with their integrated soft phone onvarious platforms, including iPhone. But I believe the main point is that whenthere is trouble, snom has to fix the problem no matter if the problem is onthe phone or one the PBX. 3CX has no desktop phones, so customers are alwaysconcerned about the proper interworking inside the office, especially withfuture software updates. snom ONE also works on other platforms like variousLinux flavors and MacOS, FreeBSD and embedded devices such as the GuruPlug (notonly Windows), which can save significant amount of money. snom ONE free has no limits on the number of calls that you can make, up to tenextensions at an unbeatable price ("free"). Even the licenses formore than 10 extensions are an unbelievable bargain. At the end of the day,snom’s business model is to make money with the phones, and the PBX's job is tomake it easy to use them. 3CX business model is different; they need to makemoney with the PBX business andthere is no cross-subsidizing.

So far one queue can have one caller at the top. What do you want to achieve? Call waiting for an agent? Or is the line so busy that you can't get the callers out quick enough?

This time the root password will be no secret . I think we are using debian this time. # cat /etc/debian_version 5.0.8 So far the devices were very stable, even the heat was no problem. Because there is no gateway built-in, you will need something external. But luckily, there is plenty of choice for excellent quality and good price PSTN gateways.

We are working on a new program that does things "right" (whatever that is). Send a private message to pbx support to get a beta, if you like.

Hard to say. There must be somewhere something where the address would require the resolution of "", for example http://:8080. It does not affect the stability of the service, but somewhere something has been configured the wrong way.

I would use a list like this: "co1 co2 co3:o". That means the like 3 is always reserved for outbound traffic.

Next version will give you a lot more flexibility, should be coming out in the next few days.

If you operate the PBX on non-routable addresses, you will always be in pain. Yea, you can start playing tricks and set up a registration from the branch office to the main office. The biggest problem is that the inbound calls into the branch office all have to go through one extension. But I believe it would be much easier to use a VPN client on the Mac and then have a virtual interface where packets can be routed without NAT problems. In your HQ office, you need to have a VPN server somewhere (does not have to be on the PBX), and then things get simple.

Just blind transfer the call? You can do this either from the phone or using the *77 transfer star code.

The 5 was related to the number of third party devices. There was a lot of confusion what exactly a third party device is and can do. The next version (4.5 I believe) will fix the problem.

It is not a easy topic. Generally speaking, I would make sure that the PBX in the company can send packets directly to each other, e.g. through a VPN. Then you can set them up independently and just use gateway trunks to send traffic between them, and the whole topic becomes relatively easy. Especially if you treat calls from one branch office to another branch office like a PSTN call (full telephone number being used), you avoid extension name clashes in different offices.

Co-lines can be qualified as inbound only or outbound only. "Reserving lines exclusively for inbound or outbound traffic may be appropriate at times. If you put a :i after the line name, the system will use that line only for inbound traffic (e.g., line1:i). If you put a after the line name, the system will use that line only for outbound traffic. If there is no attribute set after the colon, the line will be available for both inbound and outbound traffic." (see http://wiki.snomone....nks&redirect=no). That might help solving the problems.

Yea it is not really a link, it is more the path to the file which is presented as the logo. It is supposed to be in the html/img directory, relative ot the working directory of the PBX.

Looks like someone's mailbox is full...

Eh, sorry there is a space between the / and the -name... I guess we need a MAC expert here.

Nothing to be really worried about. Did you edit the templates? Or did your change the "OEM" settings? Maybe there you referenced a HTTP URL where a filename was expected.

Oh so you want to uninstall it? Everything is in the installation directory. I am not the big Mac guru, I would search for directories with the name "snom" and "pbx" in it and see if there are any files left (find / -name '*snom').

We are also not happy with the way the G.729A codec has to be licensed. The call-based license model comes from a world where large hardware boxes were using DSP to perform the codec calculations.In the SIP world, the call can re-negotiate the codec at any time; so even if you first negotiated G.711 you still might end up using G.729A. Anyway, this problem is addressed in the next major release (AKA 5.0). There we have a more dynamic look on which codecs are actually used and if they are in use, we dont offer them any more.

Did you change the "Agent recovery time" on the PBX? During that time the PBX does not call the agent. This setting is used to give the agent some time to finish up with the last call, takes notes and so on. If it does not help, try to get a log from the application. The ACD reports a lot of information about why and when it tries to call what agent. Maybe this will help to solve the problem.

Do you find the file "pbxctrl-darwin9.0" on the system, e.g. using find (in a terminal)? Maybe there is a uppercase/lowercase problem somewhere? "pbxctrl-darwin9.0" is not "pbxctrl-Darwin9.0". Also make sure that the file can be executed: "chmod a+rx pbxctrl-darwin9.0".

You mean extension names? First of all, you should make a decision how many digits you want for extensions and other accounts like conference rooms and auto attendants. It makes things easier if they all have the same length. Extension names usually should start with 4, 5 or 6. That means if you are using three digits, that would give you 300 extensions. If you are using this range, you can have the auto attendant prompt for "press 0 to ...", "press 1 to ...", "press 2 to ..." and "press 3 to ..." and press the extension number to dial directly. 8xxx is reserved for going to the mailbox directly (can be changed, but thats teh default). 9xxxxx is sometimes used to force the PBX to dial on a specific trunk. 7xx is usually used for internal resources like conference rooms, hunt groups etc. It is all just a proposal, but it has shown to be useful. Yes, every domain has a domain address book and every extension has a extension address book. You need to log in as a user in the HTTP web interface of the PBX to edit the user address book. The big question is how you get the address book. For snom phones, we are using mostly LDAP; but there are also other ways to get the address book (HTTP/XML and provisioning). It is a long topic. I would suggest to stick to the default which is automatically provisioned. There is adropdown in the login screen that lets you choose as what you want to login. Keep in mind that the SIP passwords are seperated from the HTTP/login passwords.