Vodia PBX
-
Posts
11,111 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Vodia PBX
-
-
Maybe there is a "smart" firewall that understands SIP?
-
Well, just edit the domain (don't click on the link, click on the edit icon). Then you can set your primary name (e.g. pbx.company.com) and the alias names may include the "localhost" magic string (e.g. "company.com localhost"). You can do that while the PBX is running, there is no need to restart the service.
-
Check out http://www.pbxnsip.com/download/snom_installation.ppt, maybe it helps to get PnP working.
-
The auto provisioning is working, however I don't see anything on that page regarding provisioning an intercom button.
Ehh. I think I now understand what the problem is - you want the users to press a button before entering the extension to indicate that this call should be come a intercom call? That button would be "*90"... Problem is that the Intercom feature expects the extension number in the dial string and does not accept that as DTMF later (like a auto attendant).
With the speed dial you can only program one specific destination.
How hard is it to tell users that they have to hit "*90" before entering the extension?
-
We are stepping through the Wiki and update the documentation to refrect the 2.1.1 state. See http://wiki.pbxnsip.com/index.php/Special:Recentchanges
-
-
**BUT it was 100% full when i logged in I deleted a couple files**
Did you leave the logging on high level on? It is easy to forget to turn logging off after fixing a problem - then a time bomb is ticking. -
Well, the "buttons" are nothing else than a glorified on-the-fly speed dial provisioning with the indication what light should be used. The phone has no idea if it parking a call or setting the extension on DND. Or seize a CO-line. Everything still runs on star codes! Actually, we introduced a (secret) star code for seizing a line...
You can set everything up manually, but it is painful. If you can you should just use the automatic provisioning, then you can let the PBX do the work. Plus you can tell your users just to factory reset their phone if something screwed up.
Maybe you start with a phone that uses automatic provisioning, then you can copy from that phone.
-
Well, you can do that with the auto attendant. Keep in mind that the 2.1 now supports a list of night mode redirections - you you can set up a service flag for those segments and redirect to other accounts (which then can actually be IVR nodes or other auto attendants).
-
We had a issie with a customer that was keeping a lot of CDR records (talking about 100,000 entries!), that was a major problem. Also there were some versions before 2.1 and after 2.0.3 that were eating with every registration (some internal structure getting out of control), hopefully you are not running such a build. There was also a problem when provisioning the Polycom 17 MB firmware files - the PBX was loading those files into memory using malloc(). But that should be fixed as well.
-
We updated http://wiki.pbxnsip.com/index.php/Snom... Is the description there sufficient?
-
The status screen should provide more information where to search. BTW the Wiki page has just been updated: http://wiki.pbxnsip.com/index.php/System_Status
-
Did you use the buttons (http://wiki.pbxnsip.com/index.php/Assigning_Buttons)? You must use automatic provisioning if you want to use that feature.
-
Can you move to 2.1.1.2211?
-
Had a question though, why/how is this sensitive to domain name "localhost"?
Well, that was inspired by the Unix DNS entry for the host itself. Maybe it would have been better to choose a name like "any", "anyhost" or maybe "anydomain" or "whatever". But now it's too late to change something like that...
-
You can set the number of lines to "1" in the registrations tab of the extension. Then the PBX will allow only one call for that extension, a second call will be rejected before it hits the phone. And this setting is also propagated into the Polycom, which then will also display just one line.
-
We made version 2.1.1 publically available today. There are several bug fixes, as you can see in http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.1.
There is at least one critical fix for the mixing of TCP and UDP. For those who are running 2.1.0 we strongly recommend to upgrade. As usual, make a backup so that whenever you feel the desire to undo the upgrade you can move back!
Oh, did I mention this version supports IPv6? We don't make it a big story yet, first want to find a little bit more exposure on this groundbreaking feature! Anyone who wants to test IPv6 should drop us a line, we are very interested in feedback.
-
Where can I look on the forum to find the latest version?
We are working on the login for the web page - if you register with us then you can go to the latest and the greatest head versions. 2115 is the latest released version.
-
The trick in 2.1 is that the PBX can just relay the media without really looking into it. That makes it possible to treat a G729 packet just like any other RTP packet. The 2.1 PBX also sends INVITE or UPDATE after an attended transfer so that if both sides support the same codec the PBX will happily relay it through the system.
-
There is really not much difference between moderator and participant in 2.1.0.2115. In the 2.1.1.2209 we added the possiblity to kick out all other participants (*9) and send an email to the moderators email address with the list of current participants (*1).
-
Well we kind of have the same problem - we also need to merge the latest and the greatest configuration templates. Those files get really big these days and it easy to end up in a version chaos. I guess it is time to take a look at the latest Polycom files again.
-
Are you automatically provisioing the buttons with version 7? If yes, try the speed dial mode. If not, then try the "push to talk" button mode on the phone.
-
Ehh, can you be a little bit more precise what the setup is? Where do you press the DTMF key and where are you supposed to hear it? Are you using a trunk between the two PBX?
-
Do you have a domain name called "localhost"? Does the account that you are trying to register there exist? Do you see the REGISTER request in the log when you turn the log level to 7 and enable logging of REGISTER messages?
Did you leave the logging on high level on? It is easy to forget to turn logging off after fixing a problem - then a time bomb is ticking.
Call Routing
in General Setup
Posted
Do the trunks have an outbound proxy set? The "outbound" proxy is also the "inbound" proxy for identifying where the call belongs to. Check if you see anything in the log saying "Identify trunk XXX" (log level 5 in the trunk category) and if it matches your expectations.