Vodia PBX

Good point and actually not very difficult. We'll have it in the next release.

The problem behind that was that the web interface should not present so much information, it stresses the system unneccessarily too much. The 3.1.1 version comes with a search form, so that there is no need to display more than 50 entries at a single time and there you can have thousands of address book entries. Very-short term workaround is to change the pbx.xml setting cdr_email_size. But better don't touch the web interface for the address book then...

Hehe, maybe just register them all to the same extension! Though the number 'ten' tell me there will be hell breaking loose when a call comes in.

It should be the same in Linux and Windows. I know it works in TCP (we just had that case and fixed it). UDP may be different, especially when the SBC does not kick in (maybe try to register a device from behing a router).

I would give the extensions a little bit less ambiguous number as an alias name. That will make life a lot easier. Or you can make the number a lit less ambigous in the replacement, e.g. +3412345*.

Yes that should work. Or you can just load the template into the system, see http://pbxnsip.com/templates.

You mean using a SOAP call? Not sure. But you can send a SIP request easily using a little shell script. If you use an account with no password, but bound to a specific IP address then you can bypass the tricky SIP authentication.

Well, it is empty. It is just referenced by the master XML file, you'll see the download request in the phone's log.

UDP will be really difficult as the OS makes the decision what IP address to present for a packet. The only solution is to have one socket for each IP address. For example, in the Ports section you can use the string "10.10.10.1:5060 10.10.10.2:5060 10.10.10.3:5060 10.10.10.4:5060 127.0.0.1:5060" to open five sockets; then the PBX has a chance to select the right socket for sending responses back.

1. Reboot all devices... 2. If you can, use TLS as transport layer. Then BT has no chance to mess with the SIP packets...

There is no "direct" media path between the endpoints. SRTP transcoding, recording, call barge in, SBC functionality, RTCP-XR (CQDR) reports and possibly media transcoding dictates that the PBX must stay in the media path.

Check out http://pbxnsip.com/protect/pbxctrl-3.1.1.3096.exe. This build has a new feature called "Try Loopback" in the dial plan (see http://wiki.pbxnsip.com/index.php/Dial_Plan), so that you can try to call into another domain without the need of an external proxy.

The latest and greatest (3.1.1). Okay, it is still no released... Every ITSP has it differently. The RFC3325 solves most problems with the Caller-ID indication when redirecting calls that's why we choose this as default.

We are using the following settings: Maybe the point is the ANI and the 11-digit representation. But also check the other settings.

Whow, that's great. What Linux build are you using?

I would not register a PSTN gateway to the PBX. If you can just use the gateway mode that is usually the easiest way to go. Unfortunately, I did not configure a Quintum gateway yet; but I believe you just need to point the outbound proxy to the PBX and on the PBX have a gateway trunk with the outbound proxy to the gateway.

Here you go! polycom_sip.xml

Check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses, I think that is a good guide to set up the PBX in a firewall environment.

If your device does send keep-alive traffic while the mute button is engaged, things are simple: Just reduce the one-way audio timeout to something like a minute. You can do that by changing the setting "timeout_conference" in the global config file (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File).

The PAP2 work. The above problem sounds a little like there is something in between the PBX and the ATA, maybe a firewall that wants to be smart. Can you try to get the device working just in the LAN, no Internet involved? And also, please don't use STUN for allocating a "virtual" public IP address, this usually screws pretty much everything up.

Does 101 have call redirection turned on??? You don't need a dial plan for calling in to a phone...

Starting in 3.1 you can specify what ANI to use when going through a specific trunk. If you are using the gateway mode and set the trunk for outbound only, then you should be able to solve that problem.

I think we need to take a look at the SIP packets in detail, see http://wiki.pbxnsip.com/index.php/Log_Setup#SIP_Logging on how to turn that logging on. Maybe just use the watch list to see the traffic between the provider and the PBX (put 38.99.70.46 there).

That looks okay to me... Can you turn SIP logging on and show the INVITE packet that is sent to the provider and the response which is sent back?

On snom I would use the buttons from the PBX and let PnP do the job. If you don't specify a button, you can use the local phone configuration. And in 3.1 we will include a file called snom_custom.xml, where you may specify other (generic) parameters for all phones.