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pbx support

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  1. Alex, Could you please PM to support@pbxnsip.com on this? thanks
  2. Check out the Admin->Status page. Copy paste the original license under Admin->Settings->License tab and it fixes the issue.
  3. PBX software will be added to the "Program Files" as a shortcut. It is installed as a service. If you open the "Services", you will see "pbxnsip" as a installed service.
  4. If you have PAC related javascript files in the PBX html directory, please delete them and restart the PAC.
  5. Check 'timer_register_expires' on the CISCO phone. Default is 3600, if not set it and see it changes the behavior. Otherwise, can you verify the 'nat_enable' settings on the CISCO? If it is set to 0 (zero) try setting it to 1 and reboot the phone. If you can telnet to the phone, verify the value after the reboot.
  6. Can you please explain the call scenario? Generally, the caller should hear the ringback if the "180" is set.
  7. What is the PBX version you are using? If you use the one from the download page you should not be seeing this error
  8. I did not understand completely. But, there are couple of things you can do. One set the "country code" under Domain->Settings. Then under trunk page, you can set the trunk ANI, if you want the same number to go over the trunk all the time. Also, you can play with the "Rewrite global numbers" and "Remote Party/Privacy Indication" fields to see whether you get what you want.
  9. There is an easy way to find this. Copy and paste your license into the box in this page and click on "Decode: button http://www.opinionatedgeek.com/dotnet/tools/Base64Decode/. You will know exactly what you have
  10. Is there a PBX log SIP log file for this? You can find the log by going to Admin->Status->Logfile. BTW, before you do that, go to Admin->Settings->Logging and setup the following to get the SIP log 1. Log Level - 7 2. Log Length - say 300 3. Log general events, Log SIP events, Log trunk events, Log Other Messages (e.g. INVITE) to "Yes" 4. Save. 5. Go to Admin->Status->Logfile, press 'clear'. This will clear the page. 6. Make the call. 7. Go to Admin->Status->Logfile, press 'reload' 8. Copy the content of the page to notepad file and send it.
  11. Please download http://pbxnsip.com/protect/pbxctrl-3.3.2.3181.exe. Some basic testing shows that the problem has been fixed.
  12. Does the call gets disconnected in this case? Otherwise, you may verify the “Dial tone during hold” on the snom http://wiki.snom.com/Web_Interface/V7/Advanced#Audio
  13. Oh, OK. If you have other Polycoms working fine, I would do a factory reset of the phone and see.
  14. Some version information would be useful here!!. In any case have you seen http://forum.pbxnsip.com/index.php?showtopic=2309&hl=? If the phones are on the same LAN, then you should not have any issues configuring the Polycom phones
  15. If you are not doing the PnP with this particular phone, then this setting will have not have any impact. Maybe some phone setting is controlling the behavior.
  16. By pressing the digit 8, you should be seeing the names staring from 'T/U/V' (name search - matches both first name and last name)
  17. Member deleted. Looks like the forum patch has not taken effect yet!
  18. Glad to know the problem is resolved!! (I told you it is not the PBX.. )
  19. I have tested here and it work okay. After you press ##, it goes back to the previous menu. Try deleting the audio_en folder and then download the files again from http://www.pbxnsip.com/download/audio_en_30.zip.
  20. You need not have used the installer. You could have just stopped the service, copied the update and restarted it. That is not relevant now. For registration of the trunk, please verify the account information. As far as making the calls, take a look at the "Trunk ANI & Rewrite global numbers". Select either 10 digit or 11 digit based on the provider's requirement. You should be able to place outbound calls.
  21. IP.Board, has been notified and they are applying a patch to the forum. BTW, muzrinkl has been deleted too.
  22. Is it possible for you to verify whether a normal call works fine, i.e., PSTN--E1--AC--SIP--pbxsnip AA--extension?
  23. Seem to work ok here. Can you tell me the OS and call scenario, say, whether it is coming thought the AA or Hunt/Agent group etc?
  24. Hmm, let me take a look at it by placing the same exact pattern here. BTW, you can take one of $c out ($20c and $10c) from the pattern
  25. This version lets you write CDRs to the MySQL database. Please PM to support@pbxnsip.com for the instructions
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