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Wim van Ommen

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Posts posted by Wim van Ommen

  1. I tried these settings as the are described on the www.pbxnsipsupport.com site (below the ------)

     

    I am using 3.4.0.3201 (Linux)

     

    I have a Huntgroup that has my mobile as the final stage.

    I set the ANI to a number that is valid on my trunk.

    If I call from an internal phone to the Huntgroup, a call comes through on my mobile but not with the ANI I set on the Huntgroup, but with the global ANI.

    If I call from an external number the pbx does try to call my phone, but instead of using the ANI I set on the Huntgroup, I see in the log that it tries to setup a call from the external number to my mobile.

    This would be brilliant if my provider would allow it, but they don't.

    My impression was that setting the ANI on the Huntgroup would resolve this.

    Is this a bug? How can I get this to work the way I want to?

    ------

    Final Stage and Dial Plans

    If you want to redirect the call at the final stage to an outside number (e.g. a contact centre answering calls on your behalf), then you should assign a dial plan to the hunt group OR use the domain default.

     

    Calls will be routed off the hunt group and out to the dial plan when the final stage is met.

     

    You can even set the ANI for each hunt group so that the contact centre can recognise the calling Hunt Group.

  2. Hmm, i'm unable to reply, let's try a copy/paste action...:

     

    I got it to work, but when the PHP script is too slow to respond... the call is just being redirected, I highly recommend a scriptable interface in which you can just say:

     

    If hourof(now) = 6 then begin playsound(sound.wav); redirectcallto(extension)

    and such things...

     

    But I don't even see the query being sent to the pbx, which does happen when I do normal matching.

  3. Try using the "From-based routing match list", just match anything "!(.*)!\1!".

     

    I'm also trying to do this, your routing match list in the from does work, but the SOAP call is not sent out, is this a bug in the current version?

    If I ask for user input everything is fine, if I use the match rule you state, my call goes to my own extention an I get my voicemail.

  4. The problem I have with the current script is that:

     

    A It does not start pbxnsip

    B If you start it with the corrected commandline, it uses /var/run/pbxnsip as it's working directory but also puts it in /Library/pbxnsip which then efectively does nothing.

     

    So in my opinion it should be either /var/run/pbxnsip or /Library/pbxnsip and not both.

  5. In the last installer for the mac the Startup Item that is created does not work.

    I changed it to the following to get it working right:

     

    #!/bin/sh

    . /etc/rc.common

     

    # The start subroutine

    StartService() {

    echo "Starting the PBX Service"

    cd /Library/pbxnsip

    echo Changing to directory /Library/pbxnsip

    ./pbxctrl-darwin9.0 --dir /Library/pbxnsip

    }

     

    # The stop subroutine

    StopService() {

    echo "Stopping the PBX Service"

    killall -TERM pbxctrl-darwin9.0

    }

     

    # The restart subroutine

    RestartService() {

    echo "Restarting the PBX Service"

    StopService

    StartService

    }

     

    RunService "$1"

  6. I was trying to log in to a server that does not have the right domain name set.

    When I want to log into a different domain but use the ip to find the server it does not work.

     

    Litte example:

     

    user is user@topit.nl

    default domain is pbxnsip.com

    server is 192.168.10.6

     

    if i put user@topit.nl into the username that does not work because the server is not known as topit.nl

    in a normal sip client I can say domain is topit.nl and outbound proxy is 192.168.10.6.

     

    This is not possible at the moment with PAC but it would be useful if you have a multi domain pbxnsip with no internal DNS servers.

  7. I have a customer who has sip trunks from a dutch ITSP called Solcon.

    They use a specific account and username so the accountname is something like 31356038282 where the username is something like pbxnsip

    In version 2 pbxnsip sent 31356038282@sip.solcon.nl in version 3 pbxnsip sends pbxnsip@sip.solcon.nl.

    How can I get the old behaviour back?

    I tried setting the ANI of the trunk (and even of the account) to 31356038282 but pbxnsip still sends pbxnsip@sip.solcon.nl

  8. This worked for me for call pickup blf and speed dial on the SPA932 board:

     

    fnc=blf+sd+cp;sub=330@192.168.10.7;usr=330@localhost

     

    where 330 is the number I dialed.

    192.168.10.7 is the ip of my pbx and

    localhost is the name of the used domain.

     

    Would be nice to autoprovision this though, does anyone have an idea how to adjust this?

  9. Just to make sure we are talking about the same kind of issues here:

    The software is running on a Teles Box, we are not using TLS and are using udp, the phone we use are Snom 320/360.

    The main issue is that when a second call comes in the processor load briefly goes to a 100% and at that time the audio drops real shortly.

    Immediately after that the load becomes normal and the audio is flowing again.

  10. I have a customer who has strange issues when using the Tapi driver.

    He posted allready to the forum but did not get any response yet so I will try to explain the issue a little more an hop somebody had an idea.

     

    They use Vista business with office 2007 on the client machines.

    They have installed the tapi client.

     

    When we calling with the tapi driver the connection is broken after between 30 sec and 2 minutes.

    When just dialing the number directly on the phone everything is fine.

     

    Anyone have an idea what might be causing this issue.

  11. The G4 Mac Mini is also a mini but it has a totaly different processor.

    I'm not saying that should work as that is not the most powerful processor.

    My main testmachine is a G5 dual which probably had enough power to run the application.

    That is why I was asking if it would work on PPC.

    It is fine if it will never run on PPC but it would be good if the installer would tell you!

  12. Can you please make a pcap trace and attach it. Please make a clean one which captures the SIP traffic and the RTP traffic.

    I will try to make this on monday or tuesday, I have some other questions about you test though, and some extra info below you remark.

     

    Its very strange because with call waiting off the phone simply declines the second call and the other side simply gets a busy tone or message, And that's what we see here. That's a pcap trace will be really helpful here.

     

    Did you try this with a hunt/agent group?

     

    Other thing is that it seems to only happen on the local network (so it looks like a broadcast thing), because the phone I have connected via a VPN does not show this behavior.

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