Vodia PBX

In the bad.rtf there is a call redirection turned on on the phone to 18606704381?!

I don't want to be too optimistic here. Presence has the problem that you might see the "presence" (whatever that is), but the PBX is not able to tell the phone what to do when the button is being pushed. And there are a lot of x-cisco-xxx headers in the packets. Look for "extended-refer", if you want to have some fun. When the draft was written Rohan Mahy was still with Cisco, that is a long time ago. I would not build opon that...

BLF is supposed to be really stupid. It does show the status of the monitored resource. But whenever you are pushing the button, it just calls the number. Pickup is not possible with that, that is a feature because it avoids race conditions (incoming call and at the same time pushing the button will accidentially pick up an incoming call - you don't want that). Speed dial is even more stupid. It just dials the number, no matter what. Transfer scenarios in SIP require that the phone initiates a REFER. That is a little bit tricky with the buttons. Lets see if a firmware upgrade on the phones addresses this problem in the future.

I dont 100 % get it... You mean the CS410 is only for the case of failover? Or should it run at all times?

I think the easiest way to solve that problem is to use the setting "Explicitly specify park orbit preference". If someone put a "*" in there it will mean "ask". This way we nicely stay backward compatible with that we have now and we can even specific the behavior on per-extension basis.

If you can, give 3.0.1.3016 a try: http://www.pbxnsip.com/protect/pbxctrl-3.0.1.3016.exe.

Check out http://wiki.pbxnsip.com/index.php/Snom_M3.

There is no settings for that, but you can easily fool it by putting a prefix to the cell phone and then strip that prefix in the dial plan. For example, if your cell phone number is 212-123-4567 you would put 999-212-123-4567 there and then have a dial plan entry with the pattern 999*.

Well for park/retrieve, that is not so easy. There is already a park/retrieve code that does not take an extension behind it. Plus in this case there is a useful meaning behind it - let the PBX search for you. And usually there are not so many park orbits, so that it is reasonable to put one on each key.

Okay, the next step is to get packet level trace (e.g., Wireshark) to see what is going on on the cable. Then we see if it is a jitter problem, packet loss problem, SRTP problem or whatever.

Next version will have a command line option "--no-check-ports" that will allow the PBX to start up even if FATAL errors are reported. TFTP and SNMP will not be fatal any more. The default is that the PBX will not start up any more if the SIP or HTTP ports cannot be used by the PBX.

Well, that is a policy question... If the PBX does not start because it does nto get the TFTP port, I would call that picky. HTTP is also not essential for making phone calls... So where is the line? IMHO it is reasolable to start the process anyway. If you have HTTP, then you can fix the other problems through the web interface. If you have SIP, you can run the service already.

Well, those errors are probably because the PBX does not have superturtle powers to open those protected ports.

The best way is to define hunt groups. Even if you don't really call them they form a group association. http://wiki.pbxnsip.com/index.php/Park_and_Pickup explains the priority on how calls are being picked up.

Well, there is a special feature with the cell phone. When users are calling from their cell phone into the office, they get a "special treatment" (feature). It might be annoying for testing, but for the real life it is quite useful, especially when those users want to place international calls or just use the caller-ID of the PBX for outbound calls. The feature kicks in when the cell phone is calling an auto attendant. For direct extension calls, that feature is not active.

Is that a QoS problem? If you are sharing the bandwidth with other Internet applications like Email or HTTP, then you should have a mechanism in place that makes sure that the voice packets are delivered in time. And of course you should have enough bandwidth. If you look at the BYE packet send from the PBX, you'll see some statistics that show how many packets have been received and how many have been sent. If there is a big difference, you probably have a QoS problem. There is a checklist on http://wiki.pbxnsip.com/index.php/Troubles...d_Audio_Quality that you can check. BTW is this a Microsoft-related topic? Or just accidentially the wrong forum?

Can you start the service manually? Is this a problem with the startup script or the PBX executable?

Does the FXS extension actually ring? Keep in mind this is FXO, and the gateway will DTMF the destination number out. You might have to put the real extension number in there, e.g. sip:123@127.0.0.1:5062;line=1.

What we did is allowing the intercom code to start without * at the beginning. That means you can make the code look like 99123. With an appropriate dial plan on the phone, users can just dial 9-9-1-2-3 and then can talk to their extension. But maybe we are making it too complicated. The PBX already collects extension numbers also in other places and if someone dials the intercom code without an extension number behind it, well nothing stops the PBX asking for the extension number. Bottom line, lets put it in. Next release will have it.

Unfortunately, that change did not make it in to the recordings (we found the a/this problem already, but the update did not make it on the speakers excel list). Next recording session will have it.

There is a setting called "smtp_starttls" (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File on how to set it). If it is "false" the PBX does not try TLS. That might require a newer version, at least 3.0.1. Let me now what OS you need and we'll get you a link.

Well you can add a static registration to an extension that looks like this: "sip:0@127.0.0.1:5062;line=4" (where 4 would be the FXS 4).

Good catch. Please update with the attached WAV. mb_to_move2.wav

Well, you can use "DND" for that same purpose. Call it primitive presence status...

Do you see the speed dial code in the address book? Yes, no difference between CS410 and other builds.