Vodia PBX

Okay, but then we are talking about a completely different problem than the busy tone detection. Is the call still "up" on the PBX in the web interface? Maybe there is a problem between the SIP phone and the PBX. Did you change the port setup for the FXO gateway? Or the IP address (1.1.1.1 and 1.1.1.2 by default)? Hanging up from the PBX side was never a problem...

I would say at this point that it is better to check if TLS is being used. It is a small CPU and it seems that too much TLS encryption slows the CPU down so much that you can "hear" it. We had customers that went to UDP and that did solve a problem.

Just kidding. Next version will have it. But still it would be a good idea to check out this time zone!

Okay, do you have something that can monitor if the system stays alive? Something like logging ping? I want to make sure that the OS is alive so that we know if the hardware or the software is the problem here. Are we talking about the black box?

? If you are on an extension call, the call should be cleared when you hang up on the IP phone. Otherwise we have not a busy tone detection problem here.

It does not exist. The CS410 is only on version 3, because version 2 does not have the pages for setting the system up (IP, FXO). The voice quality actually should not be different between 2 and 3.

Did you turn the busy tone detection on? In reg_ip.htm, setting is called "Detect Busy Tone".

It is our strict company rule that we first have to visit the place for quality assurance purposes...

No problem. Next version will have subject "New Voice Mail from xxx-xxx-xxxx to xxx".

Well, if it really reboots it is not a software problem (it is a Debian Linux running on the box). Maybe there is a problem with the power supply. If you have, try running it behind a UPS or use a power plug that can deal with short power spikes and outages. Or you did not set a password and some guy makes a fun out of it rebooting it from time to time...

Whow, that means the AC has a problem with the codec negotiation? The PBX answers with it's own priority (which is obviously Ulaw, then Alaw), and maybe the AC cannot deal with that... Anyway, if it works then keep it this way.

Wo-ho! That means the sipfxo process goes belly up! What version are you on?

You mean status emails? Or what kind of email notification?

Well, that timeout happens after the call is already over. The gateway initiates the hangup, and it says it does that because of RTP problems. You can see in the BYE response that the PBX sends far less packets than it receives. So it is understandable why the gateway hangs up. Maybe you have silence suppression turned on? With media related problems, it makes sense to get a Wireshark trace. Then we will be able to see why the PBX does not send RTP data. Maybe it is because of the Speech Engine, maybe there is a option to disable silence suppression. Also, check if there is a upgrade availale. 3.0.0 sounds like there is...

Could be a problem with the ACK routing (do you have more than one IP address or a firewall somewhere?). Or could be a tone detection that is too aggressive detecting a hangup. Maybe you can get the SIP packets between the AC and the PBX and we can see if the hangup comes from there. If that does not give any insight, we can do the same thing between the PBX and Exchange. Divide and conquer.

I don't have the Mitel phone here, but you should set up username, password, domain, outbound proxy as usual. If you plan to install a large number of handsets, we can look into PnP.

Is there any reason why you are using 8.3? I remember some of the 8 versions were not very good. What is the latest? 8.6? Do you have a chance to give that a shot? From the logs I must say it seems that the phone has some problem authenticating. I does not respond to challenges and that is really strange.

Well, considering that a single DID is a dollar value every month I see the registration problem relaxed. The issue is that ther eis not clear specification on how to deal with one registration and multiple identities behind it. Especially because the CS410 will probably have only a few DID per box I would say lets take it easy in the beginning and then later when the standards are more clear a trunk may have more than one DID. Actually I know carriers who seriously provide hundreds of DID to the same client each of them with a seperate registration.

I would factory-reset the phone to make sure that there is no residual configuration on the phone. Then use the BLF mode on the buttons. That should work just fine.

I remember testing Mitel phones some time ago (well, must be more than a year...). But anyway, it worked and I don't remember any significant problem. The only disadvantage I can think of is the PnP will not work out of the box.

Once you have an audio session enabled you can re-INVITE with all kinds of video and image codecs. The PBX will just be the SBC between the endpoints then, and H.263 can be transmitted between the connected phones.

What kind of support are you referring to (http://wiki.pbxnsip.com/index.php/Trouble_Ticket_Processing)? I assume this is 3.0.0.2998. A agent group does not have a voicemail. If you want that the call goes to voicemail, you have to specify a timeout value and use as destination the voicemail of an extension. Usually, you can put a "8" in front of the extension number. For example: After hearing ringback for (s) ...: 30 ... redirect the call to the destination (e.g. "73"): 8123 That means that after 30 seconds of ringing an agent the call will be redirected to the mailbox of extension 123. This setting has nothing to do with a ringing agent. This timeout is for the waiting time in the queue (where a caller would hear music on hold). 5 seconds if extremly short, a typical value would five minutes (which is 5 * 60 = 300 seconds). Call forward on no answer is a setting that affects calls directly to the extension. If the call comes in through a agent group, that setting has no meaning (otherwise if every agent sents their own little call redirection the chaos would be complete). Adding features has always the danger of making things complicated, escpecially with a relatively complex thing as a waiting queue. On the one side we get the pressure to add features here and there, and on the other side we get complaints that the software is difficult to use. We try to take the advice from both sides and have more features, but at the same time still make it possible for an average person to use them. It is not always easy.

Retrieve is the same thing.

Maybe you are right. Another was of generating kilobytes in order to turn a light on (after dialog).

Okay, but then you don't have to worry about caller-ID on the CS410. You can use ANI, or a trunk prefix or just s DID for the whole trunk. Just like any other installation. On the hosting side, you would have one "extension" for each DID number. In NAPNA, I would say one DID one registration, that keeps things very simple.