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Vodia PBX

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Everything posted by Vodia PBX

  1. Yes there was someone using it with the speech server and it worked. The setup was prettys much the same as with Exchange - I guess they are using the same underlying components.
  2. Well, if you are in the LAN and there is no packet loss than that is a very doable option. Otherwise you better use T.38. More and more gateways support it, but unfortunately this standard is very much different from RTP and the different vendors are not talking well to each other.
  3. Here we go with 2.1.0.2113: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2113.exe. Changes: Some Email servers had a problem with the base64-encoded format, that was changed. Also the SOAP CC authentication (PIN) was not working. Credentials based on P-Preferred-Identity (inbound) were not working. The ACD IM messages could "stall". Linksys provisioning files updated.
  4. Are you using CO lines in the PSTN trunk? If you put co1 co2 co3 co4 there then you should limit the number of calls that the PBX allows on that trunk to 4.
  5. Well, we had it! The PBX was sending MIME/alternative, but it was causing a lot of grief, so we took it out and now send only HTML.
  6. Well, the help/help.php is really hard coded. In 2.1, we changed it at least to help/help. What you can do is putting a subdirectory in front of it, e.g. make the setting look like this: domain.com/pbxnsip. Then it should be possible to have the help page in that location.
  7. The missed call emails that I am getting are okay. Are you using your own files in the html directory?
  8. At the moment we don't support CSTA. I have flipped through some presentations and it looks doable.
  9. Ehh... Not sure if it is available in 2.0.3.1715. It would be in the System Admin/Settings page. But it will be for sure in 2.1. In not, there might be a setting called "camp_enabled" in the pbx.xml file. If it is there, change it to "false". That would require a restart of the service.
  10. Forking calls was supported since version 1.0. Just register an extension twice and they will ring at the same time. I don't know why some people have so many problems with this trivial feature.
  11. Very cool!!! If you send me a private message I can grab the files and put them into the Linksys Phone forum and also put them into the standard distribution for the PBX. The current dial plan file is attached for your reference. dialplans.xml
  12. Just go to the admin settings and turn "Offer Camp On" off.
  13. Okay, seems we have to iron the memory during the startup phase... Will be in the next latest and greatest!
  14. Permission problem in the forum... Try again.
  15. G722: Yes. Also known as codec 9. But you must list it in the codec_preference setting. Image: I attached a screenshot of something that works.
  16. 2.1.0.2111 is out: http://www.pbxnsip.com/download/pbxctrl-2.1.0.2111.exe, only Windows so far. It fixes problems with certain versions of Polycom phones putting callers on hold (using a=inactive) and another Problem with Linksys phones that believe that codec 18 is G729a.
  17. Do yo usee SUBSCRIBE with the Event x-tapi showing up on the PBX? If not there must be something with the TSP setup. And you have to reboot your PC after installing the TSP and setting it up.
  18. We will later provide a .tgz that you can put into the web interface. The above is just the "naked" image that you would have to put directly into the file system by SFTP or some other mechanism (ftp).
  19. Hmm. Probably a problem with the case. The TZ parameters must be lower case, better keep the whole parameters lower case.
  20. The web interface feature will make it into the 2.1 release.
  21. Don't use FTP. Better use TFTP or (if you have PBX 2.1 or higher) HTTP. The PBX has a built-on TFTP server than can also perform software updates and on-the-fly generation of the configuration data for Polycom phones. See http://wiki.pbxnsip.com/index.php/Polycom.
  22. The 4-digit dial plans should be also there in the latest lang_xx files, so it should be okay to move them out of the way.
  23. That is really weired. I could hear the DTMF in the audio stream, it did not sound too bad. Of course, jitter in the DTMF tones is very bad for the detection (that is why they invented out of band!), but I guess that does not seem to be the problem here. [shrugging shoulders]
  24. Okay, got the PCAP. Did you turn inband detection on on the PBX? See the admin settings at the bottom. See http://wiki.pbxnsip.com/index.php/Overall_...gs#SIP_Settings. My success in detecting inband depends on that flag.
  25. If you only have those files there should be no problem. These files have nothing to do with the appearance of the web interface. Apart from the browser's cache. Usually pressing F5 should solve that problem, maybe you need to explicitly delete the cache.
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