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Wim van Ommen

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Everything posted by Wim van Ommen

  1. Could it be that this: http://forum.pbxnsip.com/index.php?showtopic=2826 Behaviour is changed in version 4, because I think that is now displays Group name (Caller ID Name) instead of: Group Name (Caller ID Number) If there is no Caller ID Name, I don't see anything which I did when I used version 3, could this behaviour be made switchable? I would really like to see the number.
  2. When I set a hunt group to Group name (Calling Party) in version 4, I only get Group name except when we get an anonymous call, then I get Group name (Anonymous), any idear what might be causing this and how we can fix it? Everything worked fine in version 3.
  3. Dialing my cellphone only works when I set the trunk I'm using to dial out to remote party ID or no indication, the problem is that this still does not set any ANI, it just uses the main ANI of my trunk. Any other ideas?
  4. I tried these settings as the are described on the www.pbxnsipsupport.com site (below the ------) I am using (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.
  5. But I don't even see the query being sent to the pbx, which does happen when I do normal matching.
  6. 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.
  7. 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.
  8. 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"
  9. Yes it seems the PBX disconnects the call There is no NAT between the PC that runs the TSP and the PBX I have attach a trace my customer made. The pc that sends the TAPI call should be or (they where not completely sure). The calls that failed where to 0641311695, 0638679470 could you check if you see the possible cause? The trace is at http://download.topit.nl/Download/18-11-2008.pcap.zip
  10. 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 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 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.
  11. I try to monitor extensions in a different domain as the default and tried to put the username in as username@domain but the PAC translates this as username@domain@ip address of server. Any way to do this? I downloaded the latest.zip so that should not be the issue.
  12. Could you please check the webinterface of this version with other browsers then Internetexplorer, the new stuff for codecs does not work in Safari and Firefox.
  13. 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
  14. This worked for me for call pickup blf and speed dial on the SPA932 board: fnc=blf+sd+cp;sub=330@;usr=330@localhost where 330 is the number I dialed. 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?
  15. 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.
  16. 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.
  17. Does the workaround work? Any idea on when a backport would be available?
  18. 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!
  19. I'm assuming this is Intel only but: The installer does run in ppc, maybe it is possible to check if the platform is ok before installing? Or make it a UB ;-)
  20. I've finally managed to make a trace, I've attache it. It's a Wireshark file, the problem start when 331 calls (Which you can see because there is a sip message from the agent group to my snom phone). pbnsip_teles_snom_360.zip
  21. 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. 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.
  22. I've tested a little further, the issues is not really a beep, but something that sounds like a small drop in the sound when another call comes in. Other SIP phones like the Siemens DECT/IP don't have this, so it seems a SNOM pbxnsip thing. Any ideas on solving this issue?
  23. I am talking about connected state, what happens is that I get 1 short beep when another call comes in.
  24. I have set the Call Waiting Indicator to Off in advanced_audio.htm. I still get a beep with second incoming call, any ideas on getting this to work. The phones are connected to pbxnsip and I'm using 7.1.33, but it was the same with 7.1.30. The problem occurs on both the 360 and 320.
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