Vodia PBX
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Posts posted by Vodia PBX
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About what parameter there is wrote? What filed I have to use?
In version 2.1, you can use the "parameter2" also for ANI. It is a little dirty trick that we put in some time.
In version 3 we cleaned that area up and introduced a new parameter called "ANI" that solves the problem the right way.
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Please tell me how often and what exactly time calls statistics in agent group tab resets?
Version 2 is still a little bit basic here. Version 3 has a lot more functionality in this area.
In any case, the statistics is reset during midnight.
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If a user of domain1 calls someone of domain2 and leaves a message the voicemail arrives at the extension of domain1.
900@domain1.com calls to 902@domain2.com
He chooses to leave a message.
The message did not arrive at 902@domain2.com, instead it arrived on 902@domain1.com. Which is absolutely not preferable.
How do I change this correctly? Has it to do with dialplan or something else?
If you are using multiple domains, you better stop using "localhost" for a real domain. "localhost" should be only used for a domain with nothing useful in it, maybe some global trunks.
And you need to consider using DNS for routing the calls. This will tremendously help you ro move to a server farm later, when the domains are physically located on different servers.
Then when this kind of stuff is straight, the domain identification becomes also more understandable.
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Hi. There are some minor issues. (grammatically) But they can be used.
Please report them to support@pbxnsip.com and we'll take care about it ASAP.
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I read some things about receiving fax via auto attendant but the wiki states you have to use a pattern with f but I don't understand where I have to add it.
I want a normal trunk to be directed to an extension or auto attendant (what is necessary to make it work) so Zoiper can receive faxes.
It's NOT an option to have a PSTN fax there. They are mobile.
The perfect solution would be fax2email.
Can't I force the codec to be T.38 on an extension so Zoiper will answer the call as a fax instead of Ulaw? (which now happens whan an analog fax calls the faxnumber)
Thanks in advance.
So you want the AA detect the CNG tone and then redirect the call to a specific FAX extension? Use the DTMF tone "F" for that (F is CNG in the PBX). It is like someone pushes the "F" key instead of e.g. "1".
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Is there any plans to change this in pbxnsip ?
Yea, the whole QoS story is not over yet. DiffSrv is extremly basic, that is clear. What Microsoft did in 2000 was a good start, but in the meantime so much stuff happened. For example, MPLS might be very interesting to ensure that the user has a TDM-like QoS experience.
The other problem is that we still want to support Linux (and BSD-style OS). That Windows API does not exist in Linux. Having something that works in both OS would be the best solution.
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Just ran into a second situation I did not think about. What if you need to do it to an ITSP that requires registration? Can you some how include the username/password in the static registration, or can you tell it to use the trunk's?
Well, you can't run the call phone call with a static registration "through" a trunk - however you can still point the call to the service provider. There is a little chance it will also work, but not high.
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10-digit numbers
for e.g. 0101234567 -> 0031101234567 (leave the 0, prefix 0031*)
Pattern: 0xxxxxxxxx
Replacement: 0031*
7-digit numbersfor e.g. 1234567 -> 0031201234567 (prefix 003120*)
Pattern: xxxxxxx
Replacement: 003120*
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In the phone's config file, I am actually specifying the complete URL which is http://pbxnsip ip address/tftp/services.xml I am not sure why it does not show the tftp in the packet capture.
Last resort is wireshark...
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Any timeline on when we can expect this to be patched?
We changed something for 2.1.13 (see http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.13). There is a Windows build available for this version, check out http://www.pbxnsip.com/protect/pbxctrl-2.1.13.2494.exe).
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IP on the domain on proxy?
IP address on the outbound proxy. This makes sure you don't have a problem with DNS.
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Version: 2.1.12.2489 (Linux)
There are domain defaults (dom_settings.htm) and settings on extension level (dom_ext2.htm). Maybe there is something unexpected on extension level?
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The version of the software I am using is 2923.tgz
Upgrade to 2933.
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i think now my pbxnsip and my freepbx is can integrated ,
i see in pbxnsip trunk is regitered, and on freepbx is the same registere too,
pbxnsip can dial extention on freepbx , but from freepbx still can't dial extention in pbxnsip
I would use the "gateway" mode on the pbxnsip trunk. If both the pbxnsip and the free PBX is on a routable address than this will be much easier and you don't have the registration traffic. Just make sure that you are using the outbound proxy in the pbxnsip trunk.
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This works without the authentication problem that I was having when using the HTML folder, however some phones send additional info in the url that causes your web server to return a 404 error. IIS ignores the additional info. The following is the get command from a packet capture.
GET /services.xml?locale=English_United_States&name=SEP00070E16446E HTTP/1.1\r\n
Is there any way to allow your built in web server to allow this, and ignore everything after the ? since I am not utilizing the different localles
I guess from the MAC address this is a Cisco phone...
The filename never contains the parameters after the question mark, the PBX does that part right. But because there is no path in front of the "services.xml", the PBX thinks that this file is for the user-interface of the PBX, not the provisioning part.
Usually there is a way to tell the phone from where to fetch the information. If you find any way to provision something like "/provisioning/services.xml?..." then life will be a lot easier.
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I have upgraded snom 300 & 320 to version 7 (7.1.30 ), after upgrades I no longer see call history - missed calls, received calls and dial calls. Afer an incoming or outgoing calls when I go into the call history to review it display "no data available". I further upgrades to 7.1.33 and the results are the same - still no missed calls, received calls and dial calls data
If those calls were for a hunt group and someone else picked the call up that is supposed to be a feature...
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Sorry, I think we are talking about different things. I will discribe it in more details:
Phone "A" Internal user "A" in Button-Group 1
Phone "B" Internal user "B" in Button-Group 1
Phone "C" Internal user "C" with Number "1234"
"C" is calling "A" and "B" picks this call with a Button on the phone. The Snom display ( Phone B ) shows "*6015775" and not the number from Phone "C" as the calling party.
Oh got you.
Hmm, the best thing would be if the phone displays the label in the screen right from the beginning (maybe make a call to "label" <sip:*601xxx@domain>, then the display method on the phone can figure out what to display). If the PBX changes the display after the connection, there would be a fickering on the screen before the PBX can send the proper display input.
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That doesn't seem to be what is happening.
It is getting the message to turn on button 13, but the symbol isn't appearing. Button 13 on the pbx is a DND and I changed the function key for DND to a button type instead of key event and put in 13.
Anything else I need to set to make it appear?
Don't call the button "13", better call it "dnd".
That feature requires version 7.1.33 on snom.
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It is possible to see the number from the person who is calling and not *601<call-identifier>?
Not in the star code. But the PBX sends along with the star code a label for the button, which provides additional information.
If you want to hard code a specific number, then you can still use the *87xxx starcode for "end-user" initiated call pickup.
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I saw that with 7.1.33 the DND function key can be set up to use the buttons package with PBXNSIP. If using it in this way does it show the DND icon on the display to show the status from the pbx instead of showing the light next to a standard function key?
Exactly. The "LED" is in the screen and looks like a DND symbol.
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I have been experiencing an intermittent problem with DND not turning off using snom's with 7.1.33. It seems like the phone is dialing the *79 to deactivate it because I see the call appearance light up when I press that key and the DND icon goes away on the phone goes away, but the pbx doesn't turn it off. I can't be 100% sure that the pbx is getting the *79 invite because by the time it is noticed that DND is still enabled, there have been several calls and it has been pushed out of the logs. Has anyone else been experiencing this?
Yes, the whole DND cloud is a "pain in the neck" problem.
Therefore, we introduces a couple of things in 3.0:
- At midnight the PBX may reset DND on all extensions
- The buttons may now have a name "dnd" (snom phones) which are synchronized with the state of the PBX. This works with the defeult configuration of the PBX.
The problem remains if users turn DND on locally on the phone. If that is the case, the PBX has no chance to fix the problem.
- At midnight the PBX may reset DND on all extensions
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Version: 2.1.11.2484 (Win32)
Should be fixed in 2.1.12, see http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.12.
Recordings of calls to internal numbers
in Call Recording
Posted
Okay, next question: Are you recording internal calls? Are you recording calls from an extension? If either flag matches, the call will be recorded...