Vodia PBX

We are holding are third UK based training session on the 13th and 14th August. The location for training will be EMEA Polycom Headquarters at Heathrow, London, UK. As a VIP partner of Polycom we will demonstrate the Polycom range of phones along with the 2 day training. For more details please email eusales at pbxnsip.com

The PBX does not collect the digits, this is already done on the FXS gateway. The FXS gateway has usually a dial plan where you can define when it sends the INVITE to the PBX. What FXS are you using?

Great! If you like, you may contribute text for the Wiki (wiki.pbxnsip.com), so that customers know how to set the phone up.

Of course IPv6 or at least a public IPv4 address. It all comes down to the question if the PBX can present a routable IP address and port. If you don't have a routable address, you can not seriously run a PBX there... Those port forwarding and DMZ games are really extremly support unfriendly. Sooner or later, you'll get the next issue because some idiot in the network blocks the RTP ports on the DMZ or something else. Finding those kind of problems takes extremly long time, and in the eyes of the customers it makes you look goofy. It is simply a fix, not a solution. In principle the device has two Ethernet interfaces. They are called "LAN" and "WAN" just because marketing read somewhere that devices must have two ports (us technical guys, we could of course also configure two IP addresses on the same port or just use VLAN, but that is hard to imaging for people trying to sell a tangible product). Is the LAN connected? There is a problem if you have two IP gateways on the system, then the Linux gets into trouble. The 2933 image has a JavaScript trying to protect the admin from entering such dangerous combinations.

Does the PBX advertize a useful address and port in the SDP? Maybe some of the RTP ports are not "visible" from the outside? Is this an issue with a UA coming from the wild Internet?

3.0 will include an option --pidfile <fn> where the PBX will write the PID of the PBX process (not in Windows).

Well, obviously there is not much coming back from 192.168.38.20 on port 5061 (TLS). So I would try outbound proxy sip:192.168.38.20:5060;transport=tcp. You don't need a SBC, that job is already done by the PBX.

Okay. Because you have only one IP address, the "Separation by IP Address" does not work. So you must use the "Separation by Route" method. So you should use the setting "192.168.0.0/255.255.0.0/192.168.1.1 0.0.0.0/0.0.0.0/123.124.125.126". Fill in you public IP address here. Also, you must make sure that your router does not change the ports for outbound traffic. That may be a problem. For example, if there is another device in the LAN (e.g. a regular SIP phone) sending traffic to the public Internet from port 5060, then this device might take NAT port 5060 on the router - which is then not available any more for the DMZ. Unfortunately most SOHO routers have very limited routing capabilities.

It is a 3.0 feature...

Not yet. If you are using a SIP phone, then it is usually better to display that information on the display. If you are using it from a pure voice device (e.g. FXS), I can understand it would be good to have a IVR. Plus we already have the WAV file for that. It is a complicated topic, because as soon as we start playing that back, then I can guarantee people will come and say that their tone detector believes the call is connected...

There is no setting for that. PRACK is being negotiated between the endpoints. If the gateway does not support it, then there is no need to disable it.

Maybe you have a network diagram, then we can discuss settings.

Eh, you are right. Once that you punch in your extension number, the PBX will start the call from that extension. If that extension has caller-ID blocked, it will be blocked. But it would be better if the caller-ID can be surpressed by the calling-card account, makes sense to me.

You can do that if you are the domain administrator. You can just delete the participant from the call list. With a star code you can kick everyone (except the moderator) out. It would be no problem to have the functionality to kick someone specific out - but how would you know which one it is? The only thing that I could imaging would be a code to mute participants, and once you muted the right one, then press the kick out button.

Generally, http://wiki.pbxnsip.com/index.php/One-way_Audio should be a interesting checklist. This setup is described in http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses. The problem here should be that the PBX does not know what the public IP address is. So you will have to work with the IP replacement list. Generally, this is very complicated, tricky, error-prone and everything. It is recommended to use a transparent router (no NAT) and configure the eth2 interface with a routable IP address. Also, make sure that the CS410 does not run DHCP on both ports. This screws the routing table up in Linux (that is a general Linux problem), at least if both DHCP servers give a IP gateway.

In general, you can install the PBX anywhere you like. Typically would be /usr/pbx or /usr/local/pbx. You need to reflect the path in the startup script. This script is called during the boot phase, it should run the PBX process. Apart from that, it is just a regular shell script.

Yea, the PBX does not wait until the user enters more digits. It is generally a good idea to make sure that all extensions have the same length (at least if they are numbers). Otherwise there are a lot of challenging problems when users try to punch in an extension number. The idea is to make better use of the # sign to terminate input. But that is not such a small change, it is in so many places...

The email from 341 does not neccessarily have to come from an extension. If there is a number configured on the PSTN gateway, the PBX would take that number. Check the date stamp, maybe those emails were sitting in the spool directory for a long time and suddenly they are set free. Restarting the server will not help. Check the spool directory if there is some nonsense. Maybe the PBX has the permission to write into the spool directory, but has no right to delete there? That would definitevely explain a lot of bogus emails...

We are looking into the possbility to send all recordings out by email. There were some technical concerns, but after receiving a 15 MB email today (unrelated topic) it seems that email attachment size plays no role today any more. And the nice thing is that once the email has left the PBX, it is not a PBX problem any more plus there are so many ways out there to sort emails into the right folders, that we don't have to worry about it.

Seems like "Voip Phone 1.0" has a small problem with the registration. The phone coming from the IP address "10.255.109.71" never reregisters, the phone coming from 10.255.109.195 has the same problem. Maybe this is because the re-registration interval is shorter than 32 seconds. Just an idea; it should be easy to fix that in the Voip phone software.

Also as outbound proxy? If the IP config of the host changing during operations?

It is part of the address book. The contact type describes how the PBX should handle incoming calls.

Overall, it should work fine. There were a few minor hickups that have been addressed in 3.0 (e.g. TCP keep-alive, no buddy-lists was confusing the display).

You can use the setting "Watch the presence of the following extensions" in the extension to provision the extension that you want to watch with presence.

The folder name must be audio_en (all lowercase) and it must be readable. You might have to restart the service so that the system gets the files.