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Vodia PBX

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  3. Multiple phones

    in snom Phones

    Posted

    I have a user with 2 phones 1 Aastra 57i CT and 1 snom 360, when the user is talking on the Aastra and another user intercoms to him *90xx, the snom picks up the call the way it should, however the call talking on the Aastra gets terminated.

     

    Hmm. Seems there is a problem with the Alert-Info header on the Aastra. I guess the PBX puts a header into the INVITE?

     

    Also any way to enter 2 mac addresses in the registration of a user for config of 2 phones? do I just seperate the macs with spaces?

     

    You can register multiple phones, but you can automatically provision only one phone per extension.

  4. License problem

    in General Setup

    Posted

    Where can I ask for 1 new license or is it possible to buy a dongle and what are the costs.

     

    In most cases a dongle really make sense. But at the moment we only support Windows dongles. Ask sales@pbxnsip.com for help...

     

    Is it possible to run de pbxnsip server in VMware?

     

    Not possible. One reason is obviously the license problem (create one virtual machine and you are "all set"). The other reason is that VM are not very good in keeping real time requirements. Maybe one day it will be better, but at the moment we are careful about this problem.

  7. mid stream XFER to VM

    in General Setup

    Posted

    Well, the transfer happens when the other call is not connected yet. We have a patch for that, but it is not part of a released version yet.

     

    In Wireshark, enter as filter: sip.Call-ID == "998aeed0-22a00782-71608e9b@192.168.31.111" to see the unconnected call, and sip.Call-ID == "9d5c5862@pbx" to see the call that sends the REFER.

  10. Incorrect number of concurrent calls shown on Status->General page

    in General Setup

    Posted

    For the license check things are a little bit more complicated. The PBX counts the call lets that are primary call legs - so that the example with the forking the call would count only as one. So that means if your license says you can have 10 calls, then it might actually have 20 call objects.

     

    I know what you are saying... I think for the next version we should show the number of primary calls and the number of call objects. Agreed.

  13. Drop Call on Remote Extension on Aastra 480iCT

    in Extension Setup

    Posted

    Usually those kind of problems come if people are not using an outbound proxy. Other problems are when phones place calls from received call lists where the contact is not clear. But it should be possible to locate the problem looking at the INVITE with the Request-URI.

     

    In your setting the line 1 looks a little bit suspicious because the domain name is a IP address. In multiple-domain environments it would be a little bit strange to have a alias name that is the IP address.

     

    Maybe you should try to temporarily remove the alias name "localhost" to really point out where the domain does not match in your setup.

  20. DSCP/ToS setting

    in General Setup

    Posted

    Does PBXnSIP use old skool ToS instead of DiffServ then?

     

    Well, it is just a couple of bits in the IP header. There is a setting called "tos_rtp" in the pbx.xml file that defines the value. Default is 184, which is the default code point that most vendors are using for RTP.

  21. Hunt Group

    in Hunt Group Setup

    Posted

    ... then the suppervisors phone will also ring.

     

    I guess the key here is the "also"? Then I would put them both on stage 2 and set the duration for stage to to something long. The final stage can have only one extension or destination number, so you need to move agent and boss to stage 2 or 3.

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