Vodia PBX

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.

Make a backup and move to 2.1.10...

The CS410 is running Debian 3.1 (Linux). It is pretty much the same like a PC, but not running an Intel processor but an embededd processor type. If you feel safe dealing with the PC Debian, then it is worth trying to copy the file onto the CS410. All you need to do it drop the file in the /pbx/audio_moh folder.

Yes. Either use the "Calling Card" account and call that account (your extension must have a PIN code set for that) or put your home number in as cell phone number of an extension and call the auto attendant. Then you will get a special IVR that asks you if you want to place an outbound call.

What about the logs between the provider and the PBX? Whers is the 404 coming from?! If it is sporadic than I would not assume it is something in the dialplan.

So that means the PBX is able to send media to the phone, which is good. Question is why the phone cannot send media to the PBX. How is the PBX configured? Are you using several interfaces (IP addresses)? Is it behiond NAT?

What I would suggest in this case is to change the primary name of the domain to something useful (maybe even the IP address) and move the "localhost" to the alias name. Then the PBX will present the primary name to the outside world. But if you put the domain name into the trunk, then it should also put that name there - however that depends on the RFC3325 mode. Maybe also worth a quick try.

Well, what you can do is assign a ACD for the extensions that should have this behavior. Then you get pretty much that behavior. A "one-person" ACD...

Are you using both Ethernet interfaces? How did you configure them?

What OS? Do you see the packet coming in? What is the IP configuration?

You can use sftp to upload the file. For example, you can use Putty (psftp). Put it into the /pbx/audio_moh directory. Default username/password is root/root123.

No. Actually, sending recordings as email scares me (even in the year 2008): One hour recording = 3600 * 13200/8 = 6 MB. Maybe you can fix the problem with a little Linux script (sorry for the meta code): while [ 1 ]; do sleep 10 for i in 'ls'; do if [ -older than 2 minutes ] # To make sure that this file is currently not being written mail -s "Recording $i" <$i user@domain.com sleep 1 rm $i fi done done

Well, that means that the PBX has only 192.168.1.x as IP address. That is not a public IP address... If you want it in the public you have to do something about the public IP address (http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses is a good start). Do you have a public IP address? Can you use your router in router mode, no NAT? Otherwise things will get complicated or instable.

The user-initiated recordings are parallel to the system-initiated recordings. Why outbound calls are not recorded - they should. The only thing that I can think about is licsense (which seems to be okay here) and the recording location. But maybe we should just re-test here.

The only thing that rings a bell is that the polarity change detection is off in the beginning (by default). This is because the gateway wants to "learn" if it is available or not. Maybe on next reboot try to turn it off and see if that makes a difference.

Sorry, try again.

The gain adjustment was a little bit buggy on the 2.1 version. There is a new version http://www.pbxnsip.com/cs410/update-2914.tgz that should be better with the gain. If you want to be able to move back to the old state, just make a backup of the /pbx directory before doing the upgrade.