Vodia PBX
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Well, well, it turned out there is really a bug in the licensing part.
When deleting a account (except Extension and CO-Line), the license count was not decreased.
Workaround: Manually delete the corresponding file from the file system and restart the service. Well, that is not really an option.
Therefore we made a new version 2.0.1.1624 what should fix this problem.
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AFAIK many ATA implement hold by using the hook flash. Transfer can be done using the transfer star code.
If that does not help, we must have a more flexible way of controlling a line with DTMF keys. We saw that problem already with the recording.
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We have put version 2.0.1 on the download page (http://www.pbxnsip.com/downloads.php). The release notes can be found at http://wiki.pbxnsip.com/index.php/Release_Notes_2.0.1.
Upgrades: We believe that the 2.0.1 fixes several important issues that we found in the 2.0.0 release. If you don't experience problems with the 2.0.0 release, there is no need update. However, if there should be open issues with the 2.0.0 installation, we recommend to move to 2.0.1. We recommend to backup the working directory and the pbxctrl.exe executable before performing the upgrade.
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accounts: 10
Well, that might be the problem. I guess this should be "extensions". Otherwise the total amount of all accounts will be 10, but you shoould be ablt to have more. I guess you will receive a new license key shortly...
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First let me state that we are still running version 1.5.1.6
I would definitevely upgrade to 1.5.2.7 (see http://www.pbxnsip.com/downloads.php). This is a relatively painless upgrade, because all you need is to replace the executable. You can still keep the old executable, and also move back without pain.
The whole redundancy thing depends on the error code that is returned. There is a setting for this in the trunk. Did you try that? What error code is being returned?
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Do you have a server on public IP where we can try this out? Maybe sent the address privately to support@pbxnsip. Being able to repruduce this problem will speed this up.
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Maybe there is a problem with your license. For example, if you created the attendant with the demo license, and then later got a "smaller" license, the PBX will probably allow the deleting (sic!), but not creating a new one.
There is a little trick that shows you what the license actually contains. For example, you can use the following link: http://www.opinionatedgeek.com/dotnet/tools/Base64Decode, enter the license code there. Then you can see how many of each type this license contains.
If the problem persists, please send the license back to sales and ask them to give you a new one.
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The administrator is password is stored in the global configuration file (see http://wiki.pbxnsip.com/index.php/Global_Configuration_File). Typically that's pbx.xml. You can edit this file with a regular text editor. There is a setting called "pw_pass" which contains a hash value. If you delete that hash value, save the file and restart the service the password will be empty again.
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Let me try to understand this 100 %.
So lets say the PBX receives an incoming request from 2121231234 to 9781231234 - which belongs to "Company A". The receptioning is registered at extension 123. The packet going to the phone would look like this:
INVITE sip:123@192.168.1.2;line=123 SIP/2.0
From: "2121231234" <sip:2121231234@localhost>
To: "Company A" <sip:9781231234@localhost>
P-Asserted-Identity: "2121231234" <sip:2121231234@localhost>
The phone's job is then to display the "Company A", so that the receiptionist is able to say "welcome to company A - how can I help you"?
The way to do this is to use the address book. The PBX changes the display name according to what is in the address book, also for the To-header.
I just saw that this is a bit difficult if you are using the 2.0.0 version (thought not impossible!). In 2.0.1 this will be the default behavior.
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Ok, lets clarify this:
For outgoing requests coming from a trunk, the PBX is able to answer BYE challenges.
For incoming requests that are sent to a trunk, the PBX does not challenge BYE requests.
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I don't get it... If the gateway wants to hang up, of course it should send a BYE. Where is the problem?
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Trunks can answer challenges, also for BYE request. You need to set up the username and password in the trunk for that.
However, challenging BYE has the big problem that is might not be possible to hang up a call. That is very dangerous.
The PBX does not challenge BYE. The Call-ID is the "token" for the call. If you want to protect others to send a BYE (DoS), use TLS transport layer.
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I understand the use of the tel: alias is required to properly route incoming calls to the appropriate auto-attendant.
Not neccessarily. If your provider properly handles the line parameter in the registration, the PBX can tell which domain the incoming request is using.
Both trunks have been set with no extension setting. I use the tel: alias in the autoattendant for each domain, and use the ANI assigned by our provider (example: tel:4164443333). This does not work. I'm used to having our trunks register with a username that consists of our telephone number, but our new provider does not require a user or pw to register on their side.You probably have to use the "To" header for routing purposes. If you use "!(.*)!\1!t!" (which says: use the To-header to identify the extension) in the Extensions of the trunk, the PBX should route the call correctly by the DID.
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i didnt find any option related to 100 Trying in IIS... would u plz tell me how i can turn it off?
Sorry, we did not do that ourselfs, but the one who was using somehow was able to do it.
what do u mean? what should i do to make it work?At the moment the only option seems to be to turn provisonal responses (100 Trying) off. An Ethereal/Wireshark trace would still be helpful to solve this problem.
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oh no... I guess the "100 Trying" is the problem. We already had problems with that and they should be solved, but it is something that raises eyebrows. There was an option in IIS to turn those messages off, maybe that solves the problem. But we also have to take a look into those provisional responses again.
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Well, the first priority to report problems in the respective topic folder. It is not always obvious is there is a bug, a configuration issue or just a misunderstanding. When other users search for the same problem they will likely look into the topic folder than in the gener?c "bug" folder.
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Well the phone sends the registration from a private IP address to a public IP address. So it needs to refresh the binding then. This is achieved by keeping the registration time short.
See http://wiki.pbxnsip.com/index.php/Overall_...ngs#Performance.
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Well, the PBX tries to find out if the Aastra would notify the PBX when it goes off hook. Obviously it does not do that, as many other devices don't. It is not a real reason for concern, but if you go offhook with the phone, don't be surprised you don't see that event on the LED keys of other devices.
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Well, you do not register a gateway to the PBX. Usually you just send the traffic between them without a registration. This makes Caller-ID issues simple. I am not familiar with that device, but all other gateways that I have seen so far did it that way.
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Did you see http://wiki.pbxnsip.com/index.php/Inbound_...unk#Simple_Mode on this topic?
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We about an installation where a lot of PAP2 were used over the Internet. If you keep them in the LAN, things should be even easier.
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Well, try RFC 3325 or Remote-Party-ID.
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Well in principle that looks good. Is there any log from the PBX? There could be a problem with the http transportaion of the SOAP messages.
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Well well. That feature is very problematic in many ways.
- If the call lasts for one hour, we are talking about 6 MB attachments.
- Only snom phones have a record button yet. DTMF has the problem that the other side hears this.
- If the call is already being recorded, we will end up with two (or more!) files. That is actually the biggest problem right now.
- Not talking about legal problems here.
I guess we have to revisit this topic in 2.1...
Registred Users
in Gateways
Posted
You can use SNMP to see how many registrations the PBX has. See http://wiki.pbxnsip.com/index.php/SNMP.
If you want to find out if a specific user is registered, the only thing that comes to my mind would be using curl to pull the information off the web interface of the PBX.