Vodia PBX

No, loop detection is something else (loop outside of the PBX). You can as well just edit the pbx.xml file, which is in the working directory of the PBX where you can find the max_loop parameter. This change will require a restart of the PBX (not as convenient as changing it from the web interface).

Not sure if this will help, bt we have an old link for Exchange: http://kiwi.pbxnsip.com/index.php/Microsoft_Exchange

If you use "*" as the MAC address in the extension (which is the default), then the PBX will replace the * with the real MAC address. So after a reboot it should grab the same extension. If you go the account list, you might see the MAC addresses associated with the extension. Make sure that there are no "surprises"...

Yes, there is a difference. Actually *87 means that it should pick up a call in ringing state, and a IVR node cannot be in ringing state. *87 has the problem that when the call connects, it should not "steal" the call any more from the connected party. The other code used for buttons has more the meaning of "connect to that call", and it also works for connected calls. This is more like a core B2BUA command, and it should not be used directlry from the user.

The first thing that most probably causes problems is that the number in the From/To headers contain a "-". This is a valid character, as SIP URI can contain practically any character. In a SIP URI 978-746-2777 ir not equal to 9787462777. The gateway should by all means send the number without the dashes in the middle. (I guess the XXX is not in the real trace and just there to hide your real number). And the other thing is that how can the To header have "anonymous"? The caller does not want to reveal information about who he wants to call?! There is something strange.

I think in general it is a good idea to try something new in a friendly environment. It is also a good idea to use stuff that you already own, e.g. that Mac. IMHO using the Windows installer is easier, Mac is not a mainstream platform for server-based products. If you want to reduce the power bill, later you can use the SheevaPlug, which is not only cheap (99 USD), but also uses only very little power so that a 24/7 operation is okay, especially at home. VoIP is a complex topic. There are many pitfalls, and it might be a rought ride to get into the details of the technology, especially when the network might not be 100 % rock solid. But you have to get started somehow, better in a friendly environment than in a customer environment. I would first finish the home project before going to a residential setup with 20 phones.

If you use "buttons" it should be all automatic. Apart from the *87 feature code (e.g. *8771 to pick up from accoúnt 71) there is another "hidden" feature code that connects to a call, but for that one you need to know the call number; which is provided by the PBX within the button protocol.

Did you set a country code in your domain? That would explain why you see the strange formatting.

Well obviously Skype turned on TLS transport layer (great thing) and use a certificate signed by VeriSign (also a great thing). What you need to do is import the VeriSign Root in the certificates section of the PBX (see http://www.verisign.com/repository/roots/root-certificates/PCA-3G5.pem). I hope it is the right one. -----BEGIN CERTIFICATE----- MIIE0zCCA7ugAwIBAgIQGNrRniZ96LtKIVjNzGs7SjANBgkqhkiG9w0BAQUFADCB yjELMAkGA1UEBhMCVVMxFzAVBgNVBAoTDlZlcmlTaWduLCBJbmMuMR8wHQYDVQQL ExZWZXJpU2lnbiBUcnVzdCBOZXR3b3JrMTowOAYDVQQLEzEoYykgMjAwNiBWZXJp U2lnbiwgSW5jLiAtIEZvciBhdXRob3JpemVkIHVzZSBvbmx5MUUwQwYDVQQDEzxW ZXJpU2lnbiBDbGFzcyAzIFB1YmxpYyBQcmltYXJ5IENlcnRpZmljYXRpb24gQXV0 aG9yaXR5IC0gRzUwHhcNMDYxMTA4MDAwMDAwWhcNMzYwNzE2MjM1OTU5WjCByjEL MAkGA1UEBhMCVVMxFzAVBgNVBAoTDlZlcmlTaWduLCBJbmMuMR8wHQYDVQQLExZW ZXJpU2lnbiBUcnVzdCBOZXR3b3JrMTowOAYDVQQLEzEoYykgMjAwNiBWZXJpU2ln biwgSW5jLiAtIEZvciBhdXRob3JpemVkIHVzZSBvbmx5MUUwQwYDVQQDEzxWZXJp U2lnbiBDbGFzcyAzIFB1YmxpYyBQcmltYXJ5IENlcnRpZmljYXRpb24gQXV0aG9y aXR5IC0gRzUwggEiMA0GCSqGSIb3DQEBAQUAA4IBDwAwggEKAoIBAQCvJAgIKXo1 nmAMqudLO07cfLw8RRy7K+D+KQL5VwijZIUVJ/XxrcgxiV0i6CqqpkKzj/i5Vbex t0uz/o9+B1fs70PbZmIVYc9gDaTY3vjgw2IIPVQT60nKWVSFJuUrjxuf6/WhkcIz SdhDY2pSS9KP6HBRTdGJaXvHcPaz3BJ023tdS1bTlr8Vd6Gw9KIl8q8ckmcY5fQG BO+QueQA5N06tRn/Arr0PO7gi+s3i+z016zy9vA9r911kTMZHRxAy3QkGSGT2RT+ rCpSx4/VBEnkjWNHiDxpg8v+R70rfk/Fla4OndTRQ8Bnc+MUCH7lP59zuDMKz10/ NIeWiu5T6CUVAgMBAAGjgbIwga8wDwYDVR0TAQH/BAUwAwEB/zAOBgNVHQ8BAf8E BAMCAQYwbQYIKwYBBQUHAQwEYTBfoV2gWzBZMFcwVRYJaW1hZ2UvZ2lmMCEwHzAH BgUrDgMCGgQUj+XTGoasjY5rw8+AatRIGCx7GS4wJRYjaHR0cDovL2xvZ28udmVy aXNpZ24uY29tL3ZzbG9nby5naWYwHQYDVR0OBBYEFH/TZafC3ey78DAJ80M5+gKv MzEzMA0GCSqGSIb3DQEBBQUAA4IBAQCTJEowX2LP2BqYLz3q3JktvXf2pXkiOOzE p6B4Eq1iDkVwZMXnl2YtmAl+X6/WzChl8gGqCBpH3vn5fJJaCGkgDdk+bW48DW7Y 5gaRQBi5+MHt39tBquCWIMnNZBU4gcmU7qKEKQsTb47bDN0lAtukixlE0kF6BWlK WE9gyn6CagsCqiUXObXbf+eEZSqVir2G3l6BFoMtEMze/aiCKm0oHw0LxOXnGiYZ 4fQRbxC1lfznQgUy286dUV4otp6F01vvpX1FQHKOtw5rDgb7MzVIcbidJ4vEZV8N hnacRHr2lVz2XTIIM6RUthg/aFzyQkqFOFSDX9HoLPKsEdao7WNq -----END CERTIFICATE-----

I would turn the logging on. Then you should see what patterns the PBX runs through the dialplan. Maybe you can even copy & paste then here. But this will help finding out what the dial plan does.

I opened a trouble ticket for the phones.

I guess this only happens with CSTA?

Essentially yes. Here is a link that you can study over the weekend: http://wiki.snomone....e=Service_Flags

The core problem is that the PBX does the DNS resolving every time that it sends a REGISTER. That was quite a stress test for the whole subsystem, especially when DNS records are not present (especially when DNS NAPTR and SRV records are not available). In other words, the problem was that when there was a problem with the DNS problem, the trunk would easily loose the registration as well. We made a change in this area, and if you like, please send a private message to pbx_support and indicate what OS you need. We already wanted to release version, however there are some non-critical fixed holding us back from a new release right now.

What exact message do you get? 403 Forbidden?

Yes, you can list the service flags for redirection, seperated by space. Then you should also list the redirection number, also seperated by space. For example: Service flags: 123 124 Rediration: 04131234 041311235 This means "if 123 is active, redirect the call to 04131234; else if 124 is active, redirect to 041311235; else process the call here". The number runs through the dial plan for the agent group, which is supposed to "pay" for the call. This is essentially like an extension dialling the number. If you want to use a voicemail box, you can use the settings "after ... redirect to ...". Typically, the voicemailbox 123 can be called with 8123 (8 in front). Regarding the shared mailbox, there were recently some discussions about this in the forum. In a nutshell, IMHO the best is to have the shared mailbox send the VM out by email, attach it and then process them by the user by email.

Maybe the solution is to change the translation to "Vid" or something like that very short!

You can redirect the calls in the agent group after timeout to any destination, including a mailbox. I would suggest to use "After hearing ringback for (s) ... redirect the call to the destination" and use as destination a mailbox number. For example, to use the mailbox 45, you can put "845" there, the "8" tells the PBX to call the mailbox.

I dont know... I think it is important to know where the limits are. If you really want to give customers access to the recordings, you have to keep in mind these may be thousands and thousands of files; and you can not just display them all on one web page. Potentially encrypted, with complex rules who may access the files and who may achive them, running scripts on them at midnight. Searching for file names, speech patterns, dates. All those problems are solved in a efficient way in Windows and in Linux and BSD and MacOS, there is no need we have to reinvent that wheel. Windows even might do speech recognition on the WAV files, so that you can search for content (okay, that might be Windows 8 then).

Well there are plenty user interfaces for SFTP, you dont have to rely on command line tools.

Working on it... The biggest problem seems to be the graphical artwork.

I would recomment to use SFTP. It is not only safe, it is also much easier to use than FTP (probably because of the age)

If they provide great QoS, they would love to have it documented! The PBX can help :-)

It should be there... you need to be in admin mode (not domain mode), then go to "web page control" and you should fine it there. Maybe yo ujust log out again, and then select "system administrator" on the login screen.

Generally speaking, there is a difference between the call object and the call leg. Usually, a call consists of two call legs and one call object. But calls like paging can easily end up with ten call legs and just one call leg. The user perception is about the call object, while the PBX actually deals primarily with call legs and they cunsume the CPU resources. When you want to limit calls, you want this either to protect the CPU resources or you want this simply because your hosted clients payed only for so and so many calls. That might lead to a lot of confusion.