Vodia PBX

Try http://192.168.32.30/provisioning/spa_phone_$MA.cfg. Then the HTTP server knows that this file is for PnP.

The builds for CS410 that are younger than one month have it. Documentation is still on 2.1.

Hmm. Maybe better forget about TFTP... Check out the attached files (you can also put them into the html directory, if it does not exist yet create it). There it says you should out http://192.168.1.2//spa$MA.cfg (if 192.168.1.2 is a PBX IP address) into the Profile Rule. I assume you have put a MAC address or just a star into one of your extensions? There is some general information at http://wiki.pbxnsip.com/index.php/Prepare_...r_Plug_and_Play. spa_1st.txt spa_phone.txt

Maybe the workaround is to stick to the "conference" string for clear identification it was a conference call. In 2.1.11 we'll change the type to 'v' because 't' was supposed to give the To-header. The extension is empty because the call does not go to a registered extension.

Just put the IP address in... Then the phone will automatically choose TFTP. The setting is called "Allow TFTP Password" - just set it to "always", then later when everything works you can lock this down.

Just point the tftp server to the PBX. It should work already... Also make sure that you do provision the passwords (Admin/Ports/TFTP). The problem is here that Linksys requires their own little secret algorithm for encrypting config files, which they won't gibt to use (the secret, not the program ). Also, it helps if you are using Option 66 on DHCP.

Check out http://www.dslreports.com/forum/r20408090-...wn-Since-Sunday, maybe the free service is just not available...

So you are sure that there are enough CO-lines on the trunk? Check the account overview on the web interface, if the PBX seizes a line it will show it there (you'll see the caller-ID associated with the CO-line).

The "Assume that all calls come from user" was a workaround to get things working at all - the Caller-ID presentation had lower priority. But maybe there is a way to configure the PSTN gateway to use the From-header for rendering the Caller-ID.

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...