Vodia PBX

For that you need to put the party into the contacts list. There you can also see your own presence status. We tried it, and it works okay. The PBX is just a relay of the status, not looking into it.

Well, for time being call that a feature ...

If you are just using IP on the WAN side, it might be an option to use the WAN interface of the CS410 for the public IP. Then you just need a hub/switch to connect the WAN to the router and the CS410. Then the CS410 will use one public IP address (and a private IP address, make sure it is a static one to avoid problems with the default IP gateway) and the router will use another public IP address. However in this setup, QoS will remain a problem. In the ultra-low cost segment, I only know about OpenWRT project where you essentially load Linux on the router. That gives you a lot of options if you are able to setup Linux routing. We did search for low-cost router solution some time ago. In the end we gave up on it and bought a standard Cisco router on eBay.

AT the moment there is no such settings. IMHO it would not be a auto attendant setting, as the same problem also exists with all other types (e.g. mailbox, IVR node). And it also depends on who is calling, e.g. a call from an extension to the auto attendant might be very very fast while a call from a trunk might be very slow. Maybe it would be a trunk setting. OR maybe it even has to be a setting of the PSTN gateway?

Clarification: Using a service flag to include a cell phone in the list of ringing devices depending on time of day? Remember that when the service flag is active, the cell phone is called, so if you say 8:00-18:00 that means the cell phone will ring during that time and stay off outside of that time.

Well, if you have only one private address getting a phone registered from outside will be tricky. Keep in mind SIP is two-way communications, and the PBX needs to be able to tell the remote phone under what IP address it can be reached. Telling the remote phone to use 192.168.1.2 does not solve that problem. Therefore, you need to have a public IP address somewhere. Check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses for more information.

Turn call waiting on the phone... Or use an agent group!

You need the call center edition for that feature...

What device are we talking about? Call waiting is generated by the SIP phone.

I would first try to delete the CO lines in the trunk and see if there is anything left in the colines directory. If that should be the case, then just delete them in the directory and restart the service.

Whow. Check the XML files in the colines directory - or just get rid of the CO-lines if you don't really need them. CO-lines were a pain in the neck in 1.5.

Again, this depends on how the carrier sends the disconnect. If the impendance changes, the PBX should also disconnect immediately. How long is the cable to the carrier? Maybe the signal quality is a problem here.

The latest & greatest also contains a setting for the busy tone and the dial tone detection. Check if you can find the tones in Kuwait, and then fill them in and turn busy and dial tone detection on.

Check out http://wiki.pbxnsip.com/index.php/Music_on_Hold.

Eh, maybe you should try the PCAP feature on the phone. Also, Wireshark is a great tool to see what is going on on the "cable".

Well, 3.x versions are already circling around. For example, http://www.pbxnsip.com/protect/pbxctrl-3.0.0.2918.exe. Needless to say, this is not a released version...

In head (3.x) it is done.

Yes, that should do it. Do you still see that the PBX does not generate the SIP000* file?

Whow it seems that Cisco also uses the 000 MAC prefix, so far we thought it would be 001 only... Try the attached pnp.xml and see if makes a difference! pnp.xml

Well, that the snom phones do is dialling a star code... Problem is, that star code (something like *602xxx) includes the call-number - which is hard to guess if the PBX does not tell you. Why don't you just park the call? That can also be put on a button, and then the pickup can be done with a regular star code.

It is okay if the log is from the PBX, but it show the messages between teliax and the PBX...

(Trying to find a way back to what the problem was:) So that NAT problem is solved, now the problem is that the FXO lines hang? "Hanging" meaning that the call stays connected, although it should be disconnected?

Do you see any SIP messages between your PC and the PBX? There should be some SUBSCRIBE/NOTIFY traffic.

Yes, that looks good to me. Make sure that the router forwards also all RTP ports specified in the port section of the admin mode of the PBX.

Skype does not support SIP. They are a closed user group using their own secret protocol that accesses your PC through the network, with all your data on it. Make your own judgement.