Vodia PBX
-
Posts
11,069 -
Joined
-
Last visited
Content Type
Profiles
Forums
Events
Posts posted by Vodia PBX
-
-
-
I realize that the stats are reset... what i'm saying is they're now not gathering because of the setting change i made above.
Strange. Looks like the saving confuses the list of current agents which seems to trigger the panic reaction to clear the statistics.
Can you put the "tuser" field in the web interface CDR?Probably a good idea.
-
How long is the device unplugged? If the PBX thinks there is still a registered device, things will get tricky. After the registration expires, it should not try to call the device and immediately redirect the call (not even after the timeout).
-
Just watching is not possible - if you are using SOAP then the external server needs to route the call. Which should be simple if the rule is static. If there is no response, the PBX will wait and wait wait...
-
Usually these strange effects happen when that user 260 should be called (for whatever strage reason) and either that extension is not registered any more or the number of available lines to that extension is exhausted. That strange reason could be a re-INVITE with a changed from/to-tag, so that is technically is another dialog and treated as such by the PBX.
-
The statistics are in the web interface a snapshot of the current day. They are reset on midnight, just after the email has been sent out (if enabled).
Who pickup up the call cannot be seen in the web interface. However the CDR contain that information, it is in the "tuser" field.
-
We released 2.1.7, release notes as usual to be found on http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.7. This update is recommended for users that are running 2.1.6, it only fixes problems that were found in 2.1.6 and does not have any new features. For the CS410 we recommend to use the 3.0 build, as it includes important flags for the FXO subsystem.
-
Unfortunately, that is not a simple problem. When the user presses pound the message technically exists and is new. Until the caller deletes it, the user can go into the mailbox and listen to it.
What we need to do is set the message to state "temporarily" and ignore these messages in counting messages. We fix that in head, then we can see if we port it back into the 2.1 branch.
-
2. would be a nice addition to the feature.
2. will be added in the next release.
-
What was the "native" OS on the MAC mini again? Something with BSD?
-
Looks like you have a DNS problem. Simple workaround: Use the IP address in the outbound proxy.
What OS are you on? Maybe there is something strange with the DNS configuration.
-
should i be concerned?
I would be concerned. Especially about the "voice kernel" - what is that? Is the kernel aware that there is voice flowing through the network?
Apart from that, the errors seem to happen every few seconds. This will make it impossible to receive FAX, but having a normal audio conversation is still no problem. Maybe something simple like a bad cable.
-
There is a softphone that supports T.38 (kapanga, http://www.kapanga.net), that might be a workaround.
-
I changed the conference room to extension 900, and that certainly fixed it. Is there a way to override the 8 or disable it?
You can remove it from the domain settings, but then you loose the ability to call someone's mailbox directly.
Better arrange the extensions so that you stay in the range 4xx..6xx. Then you can use 7xx for stuff like conference server, and you don't conflict with direct destinations in the auto attendant in the range 0..3.
-
I would try turning the SIP awareness off and see of that module is the problem. Then we can drill deeper from there.
If that does not help a SIP trace from both sides of the firewall. Then we can see if the packet gets changed or even rejected.
-
I added the code you suggested, and it solved my problem. Just out of curiosity, I added it as you said with 123 (70 for my AA) as the default extension, but it removes the number. It is working and sending the calls to the correct extensions though.
Removes the number? You mean from that input field in the web interface?!
Is this documented somewhere that I should have been able to figure this out myself? Also is there a place that defines how to write your own codes like the one above?Well, it should be on that Wiki page, but it is hard to write something down that covers all cases clearly..... So it is good to have a (public) forum, so that people can also search for such cases and get more hands-on examples.
-
Then the tel:-alias should be the right thing for you. Don't forget to clear that "Send call to extension" field. If the carrier put the destination into the To-header, then you need to put the destination out of the to header. You can do this with the following string in "Send call to extension" (assuming that 123 is the default, which could be your auto attendant):
!(.*)!\1!t! 123
-
2.1.7 will be available this week and fixes the # problem. Maybe that's easier.
-
Did you see http://wiki.pbxnsip.com/index.php/Inbound_Calls_on_Trunk? You can just put auto attendant number into the setting "Send call to extension".
-
There is something on the Wiki http://wiki.pbxnsip.com/index.php/SonicWall, a little bit outdated maybe. Maybe a time to update it and check what the latest versions are.
-
The mediation server is like a PSTN gateway. That means the PBX does not register there, the trust is based on IP addresses (which is not very safe compared to Digest). It is a little bit "workaround", but people were able to get it working that way. Of course, registering natively is much easier. That is why we are working on it....
-
Unfortunately, OCS does not support the standard SIP authentication scheme (which is Digest). They only support NTLM and Kerberos. Microsoft recently published the specification for NTLM, so that technically we would be able to register there. However, we still need to program it... That will defintevely not be in version 2.1 of the PBX.
The workaround today is to use the mediation server.
Maybe Microsoft can come up with a service pack which adds Digest authentication. A lot of SIP-compliant vendors would appreciate it.
-
Yea, it is a new topic. We have something on http://wiki.pbxnsip.com/index.php/Office_C...ications_Server, maybe it helps get get started.
-
Can you elaborate? I'm contemplating rolling 2.1.6.2550 this weekend - I need a recommendation please.
2.1.7 will contain only bug fixes, nothing major yet. We'll do release notes when it is ready. 2.1.6.2450 seems to be pretty stable so far.
Redirecting incomming external calls
in General Setup
Posted
Hmm. If you use the auto attendant, you can block certain extensions (typically executive). On the trunk level, you can also send calls to certain extensions to other extensions.
From a button you can turn day/night mode on, so that you can route calls at night to another auto attendant, which does allow calling those certain extensions.