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R. Baltzer

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Posts posted by R. Baltzer

  1. It has been fixed already and we will put out the update. When we tested the software 8.4.32+ was not available. The Call-Info header(looks like it has been introduced in this version and it is always on by default) exposed a bug in the PBX software.

     

    What OS are you using?

     

    I am using Windows 2008 x86

     

    Kind regards,

    Rene.

  2. Can you post the snom ONE log with SIP debug or PM it to me?

     

    To get the log with maximum debug level:

     

    - go to Admin -> Settings -> Logging

    - set "Log Level" to 9 and "Log Length" to 500

    - enable all logs except for "Analyze audio levels"

    - also set the following logs to to Yes: "Log REGISTER", "Log SUBSCRIBE/NOTIFY", "Log other messages", "Log call messages"

    - press "Save"

    - go to Status -> Logfile and press 'Clear'

    - make the test, then press 'Reload', copy the log and attach it here or send it to me

     

    Hi Katerina,

     

    I have PM'ed you.

    Thanks in advance.

     

    Kind regards,

    Rene

  3. Note that if you have button profile assigned to that extension, it will try to override the default assignment(snom_300_fkeys.xml). I think the extensions, be default, get assigned with the a "12 Buttons" profile, which would be wrong for a snom 300. So, can you please check if there is a wrong profile assigned?

     

    Thanks for replying.

     

    I understand that the assigned buttonprofile should overwrite the default button assignment, but that is my problem;

    There are some Snom300 phones with assigned buttonprofiles and they still have the default Snom300 buttons.

    So my guess was that the button assignment is not working.

    Please note that the button assignment for my Snom320 phones is working fine, only the Snom300 phones are not getting the alternate defined buttons.

     

    I have reverted back to PBXnSIP version 4.2.0.3958 and now also the buttons for the Snom300 phones are provisioned correctly again.

     

    Kind regards,

     

    Rene.

  4. Hi,

     

    I have upgraded PBXnSIP version 4.2.0.3958 to the new version 4.2.0.3981 and now I have some provisioning problems:

     

    My Snom 300 phones are not getting the configured function keys anymore (the phone is not picking up the config) and also the TFTP custom xml files (snom_300_custom.xml and snom_320_custom.xml) are not picked up by the Snom 300 and Snom 320 phones anymore.

     

    I have reverted back to version 4.2.0.3958 and now PnP is working fine again.

     

    My PnP config is based on the DHCP option 66 and my Snom phones have firmware version 8.4.27.

     

    Could this be a bug in the new PBXnSIP version?

     

    Kind regards,

     

    Rene.

    PS: I have posted this same question at http://forum.pbxnsip.com/index.php?/topic/4328-v42-first-update-available/

    but there it is still unanswered.

  5. Just curious why you upgraded to 4.2 if it was working fine in 3.4. Was there a particular feature you wanted in 4.x? Can you fall back to 3.4 for now since it sounds like a bug as pointed out in the prior post with the buttons that needs to be sorted out.

     

    I know, but we are a reseller of IT related solutions and we also support the PBXnSIP software.

    For this reason I want find problems myself in our own network instead of onsite by my customers.

     

     

     

  6. The pbxnsip shared lines implicity lock the call to the seized line on the trunk, that is obviously not what you want. Park orbits may be something you want to use instead, they look similar and small groups can still transfer calls with one button push.

     

     

    Thanks for replying.

    This configuration was working fine with PBXnSIP 3.4.0.3201 (also with the Shared Line buttons) and it stopped working after the upgrade to version 4.2.0.3958.

    I guess that the working of the Shared Lines is changed a lot then...

    I will consider changing the configuration.

    Thanks again.

     

    Rene.

  7. I do not understand what you mean by this.

    I have provisioned the fkeys with PBXnSIP and programmed them to be "Shared line" buttons.

    All my phones have been reset to the factory defaults before they were provisioned by PBXnSIP so there can not be any leftovers from an old configuration.

    Do you mean by this that I should not be using the Shared Line buttons with PBXnSIP?

  8. Well, are you using shared lines on the phones? Or did you use it before (maybe the configuration is still present on the phone)? This would explain such strange behavior: First the phone seizes the line on the on-Exchange trunk and then when it dials the number the PBX actually keeps the promise and looks only at those entries of the dial plan that have this trunk as the destination.

     

    I have checked the shared line option on my Snom phones, but these are all disabled:

     

    <user_shared_line idx="1" perm="">off</user_shared_line> <user_shared_line idx="2" perm="">off</user_shared_line> <user_shared_line idx="3" perm="">off</user_shared_line> <user_shared_line idx="4" perm="">off</user_shared_line> <user_shared_line idx="5" perm="">off</user_shared_line> <user_shared_line idx="6" perm="">off</user_shared_line> <user_shared_line idx="7" perm="">off</user_shared_line> <user_shared_line idx="8" perm="">off</user_shared_line> <user_shared_line idx="9" perm="">off</user_shared_line> <user_shared_line idx="10" perm="">off</user_shared_line> <user_shared_line idx="11" perm="">off</user_shared_line> <user_shared_line idx="12" perm="">off</user_shared_line>

    All the Snom phones were flashed to firmware 8.4.22 after upgrading PBXnSIP from version 3 to 4 and the phones were reset to the factory defaults too.

    So there could be no leftovers from an old configuration I suppose. The other phone settings are all due to PnP from PBXnSIP.

     

     

     

  9. The only thing that could explain this is that the phone seizes the outbound line when dialling. Then the PBX would restrict the call to that trunk only. Could you check in this direction?

     

     

     

     

    Could you explain this a bit more because I have no idea where to look?

  10. Yes, that is really very strange. The same dial plan returns different results for the same input. So again, it makes a difference if you dial via the speed dial or directly from the phone? If that is reproducable, any clue from the SIP packets (INVITE)? Maybe there is a local dial plan on the phone that could cause this. Do you have multiple identities on that phone? Did you try the dial plan simulator (at the botton of the dial plan web page)?

     

     

    I have tried the dial plan simulator, it tells me that the Exchange trunk should be used.

    So the dial plan is looking fine.

     

    All my phones do have only one identity configured.

     

    I have also created 2 log files on my Snom 320 phone (firmware 8.4.22).

    One log while manually dialing 7900 and one log while pressing the retrieve button (programmed with speed dial 7900).

    It looks to me like the problem is caused by the Snom phone; when I start dialing 7900 the phone is immediately sending some sip commands and these are using my sip trunk.

    So now my dial plan is not functioning anymore I guess...

     

    Could this be the problem?

     

    LOG1.txt

    LOG2.txt

  11. Hi,

     

    Since I have upgraded from PBXnSIP 3.4.0.3201 to PBXnSIP 4.2.0.3958 the dial plan is doing some strange things.

     

    My dial plan:

    100;Exchange Gateway;;7*;*

    200;SIP Provider;;112;

    300;SIP Provider;;xxxxxx;"sip:0123\1@\r;user=phone"

    400;SIP Provider;;*;

     

    With PBXnSIP version 3.4.0.3201 I was able to dial 7900 to connect to my Exchange server.

    Since version 4.2.0.3958 this is not working anymore, this call is always directed to the "SIP Provider" trunk.

    But even more strange is that my Snom 320 programmed button (speeddial 7900) is directed correctly to my Exchange server!

     

    When I dial 7900 I see the following log information:

    [5] 2010/11/24 08:15:59: Dialplan "PBXnSIP": Match 7900@localhost to <sip:7900@113.156.197.104;user=phone> on trunk SIP Provider

    [5] 2010/11/24 08:15:59: INVITE Response 500 Server Internal Error: Terminate 304602b4@pbx

     

    When I press the programmed speeddial button I see the following log information:

    [5] 2010/11/24 08:15:20: Dialplan "PBXnSIP": Match 7900@localhost to <sip:900@192.168.0.10;user=phone> on trunk Exchange Gateway

    [5] 2010/11/24 08:15:20: Charge user 900 for redirecting calls

     

    As mentioned before, PBXnSIP version 3.4.0.3201 was working fine with this same dial plan!

    Does anyone know what's going on here?

     

    Kind regards,

     

    Rene.

  12. Hi,

     

    I have upgraded my working configuration PBXnSIP 3.4.0.3201 (with a configured and working Exchange 2010 UM server connection) to PBXnSIP 4.2.0.3958.

    Everything is working fine except one thing:

    I cannot play a voicemail message anymore on my PBXnSIP connected sip-phones.

     

    When I click in Outlook on "Play on phone" then I get the following error message on my Exchange server:

     

    An outbound call to sip:907@192.168.0.35;user=phone couldn't be established. The selected outbound gateway "(PBXNSIP 192.168.0.35:5060 outbound=True secured=False)" returned the error "Peer to peer endpoint does not support authentication.". The caller ID used for this call was 'sip:907@EXCH.LAN.LOCAL;user=phone'. Contact the vendors who support your VoIP gateway and IP PBX hardware for help troubleshooting the SIP response error code specified in the event description. You can also run diagnostic tests on your VoIP gateway or IP PBX hardware to make sure the devices are operational. Additional information: Peer to peer endpoint does not support authentication.

     

    At the same time I see the following error message on PBXnSIP:

     

    [8] 2010/11/22 17:17:57: Received SIP connection 48 from 192.168.0.10:30325

    [5] 2010/11/22 17:17:57: Identify trunk (IP address and DID match) 2

    [8] 2010/11/22 17:17:57: Call from an user 907

    [8] 2010/11/22 17:17:57: To is <sip:907@192.168.0.35;user=phone>, user 55, domain 1

    [8] 2010/11/22 17:17:57: To user 907

    [8] 2010/11/22 17:17:57: Set the To domain based on From user 907@localhost

    [8] 2010/11/22 17:17:57: Call state for call object 5: idle

    [8] 2010/11/22 17:17:57: Call state for call object 5: alerting

    [8] 2010/11/22 17:17:57: Play audio_moh/noise.wav

    [7] 2010/11/22 17:17:57: set_codecs: for 24f8e252-8d1e-4637-873c-49bef5ab4587 codecs "", codec_preference count 6

    [7] 2010/11/22 17:17:57: set_codecs: for 20985dd5@pbx codecs "", codec_preference count 6

    [7] 2010/11/22 17:17:57: update_codecs for 20985dd5@pbx: adding codec pcma/8000 to available list

    [7] 2010/11/22 17:17:57: update_codecs for 20985dd5@pbx: adding codec pcmu/8000 to available list

    [7] 2010/11/22 17:17:57: update_codecs for 20985dd5@pbx: adding codec g722/8000 to available list

    [7] 2010/11/22 17:17:57: update_codecs for 20985dd5@pbx: adding codec g726-32/8000 to available list

    [7] 2010/11/22 17:17:57: update_codecs for 20985dd5@pbx: adding codec gsm/8000 to available list

    [7] 2010/11/22 17:17:57: update_codecs for 20985dd5@pbx: codec_preference size 6, available codecs size 6

    [3] 2010/11/22 17:17:57: Via is empty, cannot send reply

    [3] 2010/11/22 17:17:57: Last message repeated 2 times

    [5] 2010/11/22 17:17:57: Call 20985dd5@pbx: Last request not finished

    [8] 2010/11/22 17:17:57: Call 20985dd5@pbx: Response does not correspond to open request

    [8] 2010/11/22 17:17:58: Last message repeated 2 times

    [7] 2010/11/22 17:17:58: Call 20985dd5@pbx: Clear last INVITE

    [5] 2010/11/22 17:17:58: INVITE Response 487 Request Terminated: Terminate 20985dd5@pbx

    [8] 2010/11/22 17:18:00: Packet authenticated by transport layer

    [8] 2010/11/22 17:18:05: Last message repeated 4 times

    [6] 2010/11/22 17:18:05: SIP TCP/TLS timeout on 192.168.0.10:30325, closing connection

    [8] 2010/11/22 17:18:05: Release SIP thread 48

     

    When I click on "Play on phone" my sip-phone rings one time and then tells me that the connection was ended.

     

    This "Play on phone" functionality was working fine with PBXnSIP 3.4.0.3201.

     

    Can anyone tell me what is going on?

    Nothing else was changed, only the PBXnSIP software was upgraded to the latest 4.2 release.

     

    Kind regards,

     

    Rene.

  13. Hi,

     

    I have some PnP problems with the new version 4.0.1.3499, my Snom 300 and 320 (fw 8.2.29) phones are not provisioned correctly anymore:

     

    I cannot set the VLAN ID anymore via the PBXNSIP webinterface, the field is gone on the PnP tab.

     

    Also the Pin for Administrator is not working, it doesn't matter what is configured, the phone will be provisioned with "1234".

     

    And even the web password needed for access to the Snom webinterface is not provisioned to the phone so I cannot access it anymore due to authentication failure. The password provisioned from the snom_3xx_phone.xml placed into the generated directory is not working to get webaccess to the phone...

     

    The last thing I have noticed about provisioning is that the language on my phones is set to english, the phones are not picking up the dutch language anymore? When set logging to 8 I see the following error message in PBXNSIP:

    [8] 2010/05/01 15:15:26: Received request for file snom/gui_lang_NL.xml

    [8] 2010/05/01 15:15:26: Error finding snom/gui_lang_NL.xml, Send back 404 Not Found

    And the files are still there...

     

    Another problem with the assigned buttons is that the agent login/logout button is not working anymore.

    With the 3.4 release of PBXNSIP this was a toggle button to login to or logout of the assigned agent group but now only the led is showing the status and the button is not working anymore. Using the star codes to login and logout is working fine, but the button does not.

     

    Does anyone know how to handle these problems?

     

    Kind regards,

     

    Rene.

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