Vodia PBX

Did you set the coutry code in the domain settings? Also, if you are using tel:-alias, make sure that your trunk has the "global" flag set.

Are you using plug and play? Consider changing the TFTP password policy in the System/Settings/Port section. Maybe the phone just does not get a password provisioned. http://wiki.pbxnsip.com/index.php/Polycom was updated recently Maybe this would be a good topic to discuss in the Polycom section of this forum...

Well the tag "{http-url}" is replaced with something like "http://pbx-adr/provisioning". That should be okay...

What did you put into the domain setting "External Voicemail System"?

Scratch scratch... In that field * has a special meaning... Scratch head scratch... If you put a * there it means that you can only retrieve calls. Don't ask me why, but at the moment that is my understanding (maybe because many phones can only send a simple star code and that's the only way to get some kind of BLF working, at least for call retrieve). Can't you just list the orbits there?

What is bad about spa$MA? <flat-profile> <Profile_Rule ua="na"> {http-url}/spa$MA.cfg </Profile_Rule> <Resync_Periodic ua="na"> 5 </Resync_Periodic> </flat-profile> On the packet size I agree, 30 ms is a bad idea. We'll change that to 20 ms.

Well generally the differences are described in http://wiki.pbxnsip.com/index.php/Release_Notes. That's not a very shiny "markting" document, just facts.

For that you need to set the DID numbers on the PSTN gateway and assign those numbers as alias to extension 100, 200. This will work if you leave the "Send call to extension" field is empty in the trunk. You should then assign all DID numbers to extensions or other accounts, so that all calls get routed somewhere.

In that case "sip:2\1@\r;line=2" would be a good replacement (if you want to keep the "2" in the beginning).

Well design is always a question of taste... But the metal foot stand gives a pretty solid impression. Not bad for a device that stands in a high angle. And the display is "crystal" clear. No doubt a great phone. The integration with pbxnsip is just like the other phones in the 3xx series. Paging and intercom still runs through the PBX. How could intercom be bypassing the PBX?

Nice is not enough. The RTP thread needs to run in an different scheduler class. The PBX does that by setting the RTP thread to SCHED_RR. Threads in this scheduler list are processed before any other thread in the "nice" list is being processed. Yes, SCHED_RR is brutal. Not even the mouse moves when the PBX wants to process RTP.

You can tell the PSTN gateway what line to use for an outbound call by using the line parameter. For example, when dialling sip:9787462777@1.1.1.1:5060;line=2 the PSTN gateway will use line 2. Now, in order to use this feature, you need to put that into the replacement part of the dial plan. For example, replacement sip:\1@\r;line=2. And in order to assign that to a specific user, you need to have a dial plan for every "exception" of the standard dial plan. If you have four users and they should have their own exclusive line, then you need to have four dial plans.

yes!

Well, you can include the domain name in the recording path.

Try turning polarity reversal detection off on the PSTN gateway settings. The PBX probably believes that the call is still ringing, but it is already connected.

Okay. So there is no PSTN gateway or ISP involved. Usually after a redirection the phone should show something like "9787462777 R:100" on the display (the R: means that that call was redirected from 100). But that only works if the phone actually went to a real phone, auto attendant does not count. So I guess that is where the problem is?

Ehh. Maybe use a Flash movie?

Oh you mean having calls coming from a trunk and going out to a trunk. That is actually possible if you set "Accept Redirect" and "Assume that call comes from user".

Someone mentioned recently that cell phones have a feature that allows to put pause into the address book entries. That was a trick to get it working. The PBX itself does not support two-stage dialling. The first stage SIP is completely different from the second stage which would be DTMF. But some PSTN gateways may be able to perform this task. From the PBX perspective, all it does is dial a SIP URI!

Today the answer is "no". Ipv6 works great on Windows and PC-based Linux and FreeBSD. But for the appliance it is still work to do. I wish that the chip vendor eventually provides an upgrade that includes this. We are not very good at compiling tool chains and kernels...

Otherwise it will be quiet messy for the PBX to route the call to the right destination (and difficult for the user to understand whats going on). Call it "tradition"... CMC is "client matter code". That is a new topic coming up. We'll extend on this to group address book entries together to form a "client" and use it in the CDR and in the ACD. Stay tuned.

If the file system full? Check the status page of use the shell and enter "df /". We tried that version and it even did the hangup detection in a nice way so that we can think about getting some nice sleep at night!

Nono - that is only for multicast RTP (it is in the multicase section of the Ports). For every port, you need to specify where to hind the port. The dedault is something like "80", which translates into "0.0.0.0:80 [::]:80" (unspecified IPv4 and IPv6 address). If you want to bind to a specific IP address, use something like "192.168.3.64:80". Those examples are for HTTP, you need to do the same thing for tje other ports with different port numbers.

The next build will force the same timezones for midnight reset and reporting.

Did you put something in like "198.133.219.25:80" (for HTTP)?