Vodia PBX

What kind of transfer? A redirect? Or a transfer initiated from another phone after the call was connected?

So far it seems to be a problem. Actually the problem is that once the call is on the phone, it is very difficult to keep the call and manipulate it from the cell phone. I could imagine that we introduce a new "pickup" that allows a secretary to take over an existing call which is not on hold or parked. Permissions would be similar like call barge in.

Okay, updated http://wiki.pbxnsip.com/index.php/Agent_Group.

Ouch. Very good catch!

Sorry, try again.

The parameters are used in the respective file. For example, there is somewhere in the snom_3xx_phone.xml file a place where the parameter vlan is being used. We started doing that because there are some settings that you just don't want to hard code.

This time we want to try out the Linux versions also out a little bit before we go out with the 2.1.4 version. There were a couple of changes in the Linux threading subsystem - hopefully for good. The other changes are: - Polycom 2.2.2 support - Domain deletion and some other deletions did not make it to the file system - Support for "simple" CDR (e.g. metropolis) http://www.pbxnsip.com/download/pbxctrl-suse10-2.1.4.2344 http://www.pbxnsip.com/download/pbxctrl-rhes4-2.1.4.2344 http://www.pbxnsip.com/download/pbxctrl-debian3.1-2.1.4.2344 http://www.pbxnsip.com/download/pbxctrl-2.1.4.2344.exe Feedback appreciated - we want to make sure the release can make it until Dec 24!

Maybe you can take an Wireshark trace, put it somewhere on a web server and post a private message with the link so that we can take a look at this mystery.

What is very strange is that the PBX sends a SRTP key in the offer via an insecure connection. I guess we need to test the cases again where the initial INVITE does not contain SDP...

There is a TAPI service provider available (see http://wiki.pbxnsip.com/index.php/TAPI_Service_Provider), that should solve his problem. The latest version is 1.1.

Maybe a static registration like "sip:123456@192.168.0.123" might solve the problem.

There is a list on http://en.wikipedia.org/wiki/Comparison_of...g_media_systems. I agree most of it focus on radio broadcast, but as far as I can see they also have a simple RTP streaming mode. The whole RTP story started in the broadcast area, so many servers still support RTP. And of course in Windows you can use the loopback device to feed audio content into the WAV input device.

The PBX natively supports simultaneous calling. If you register two phones to an extension, they will ring at the same time.

Usually those problems can be solved using the tel:-alias feature (if you give an account an alias name starting with "tel:" then is has a system-global scope). The PBX will then send the call into the right domain automatically, no need for an external proxy doing that job. See http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk.

Those calls hang in the call list for a long time (I think more than 8 hours). The reason is hard to describe, if it would be clear than that call would not hang around.

Try your favorite search engine on "Streaming Audio Server"... Just make sure that the server support G.711 RTP format. I used one a long time ago, unfortunately forgot about the name.

Well, it can really happen that there are "junk" calls. That is the purpose of the garbage collector. There should just not be too much junk on the street.

Okay, now I am concerned...

Nothing in "From" or "To"???

Hmm, codec 8 is alaw, which is supported by the PBX... Looks like the initial INVITE does not contain a SDP, so the 200 Ok of the PBX must the the SDP offer. This is legal, but unusual. It would be interesting to see the whole INVITE/200/ACK sequence.

No, this is just about recordings. Either pressing the record button or matching the DTMF sequence for starting recording. You should get all recordings, either user-initiated or automatically recorded or just mailbox recordings.

The content-type must be set to "application/xml" or "text/xml", the license must support SOAP and the source IP address must be listed in soap_trusted_ip...

That way of doing the upgrade is perfectly fine. See http://wiki.pbxnsip.com/index.php/Installi...#Manual_Upgrade. That recording thing is really strange. Does your license support recording at all? Or did you previously turn on recording with the demo license and then put another license key there?

Well, this is very strange. Does the auto attendant work? Maybe just create a new extension and try with this. Is there anything in the log file that could help finding that strange problem? Do you have a personal on that computer?

No. Does he need it to understand the protocol or just for syntax checking?