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Vodia PBX

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Everything posted by Vodia PBX

  1. Well the problem is probably here that the Re-INVITE is killing it. RFC 4916 makes sense, but it is not supported by the phone. See http://wiki.pbxnsip.com/index.php/Indicati...ge_of_Caller-ID.
  2. Ehh that DTMF level feature was added in 2.1.5...
  3. We have an internal discussion on this. After a attended transfer it is not clear what duration should be shown. It could be that we change the currently policy and show only the duration after the attended transfer. It seems that this is what the good old systems were doing.
  4. I recently had a disussion with a very large carrier and a guy that really understands SIP and the whole VoIP story. DTMF is one of their biggest problems, and the only common denominator is inband. "Transcoding" DTMF is very very difficult, if not impossible. That is the sad truth and we should not hit too hard on the service providers, they really have a difficult position.
  5. At the moment the only thing I can think about is using curl or SOAP. Maybe we need a setting "use default of the domain" and have it changeable in the domain.
  6. On log level 9 (media) there is a way to see what the inband DTMF detector is doing. You can use this to see if there s a problem with the power levels for DTMF. You should see something like "DTMF: Power: 122 34 322 343 535 345".
  7. Well, this provider must support T.38 - otherwise the whole discussion is quite theoretical. This is the first thing to check.
  8. No we are still struggling with other bulls*** like callback.
  9. Yea, the IVR prompts are not there yet - but the Wiki is correct: http://wiki.pbxnsip.com/index.php/Mailbox#...opying_Messages
  10. Yea, sorry typo. I know that for example AudioCodes, Vegastream, Mediatrix and also newer versions of Grandstream work with T.38. I think also Cisco Gateways support it. If you are looking for a sofphone that supports FAX (!) look at Zoiper. Did I forget a vendor?
  11. Check out http://wiki.pbxnsip.com/index.php/Office_w...ic_IP_addresses for thus ugly topic. Can't wait until we have IPv6 rolled out, then there is no more need for this kind of address recycling!
  12. The best way to track this problem down is to use Wireshark and record a conversation. Typically, those kind of problems are because of NAT or firewalls. Also silence suppression can be a problem. A wireshark trace will show it. More information can be found in http://wiki.pbxnsip.com/index.php/One-way_Audio.
  13. Set the outbound proxy of the trunk to your service providers address. The outbound proxy on the phone should point to the PBX address. And BTW you better upgrade those phones to 7.1.30, otherwise you will experience problems with SRTP. You better include the logs from the PBX, not from the phone - then it is easier to see the PBX perspective!
  14. For this, you should take a look at the buttons (http://wiki.pbxnsip.com/index.php/Assigning_Buttons). Should be an easy thing to do.
  15. For those who can give the latest and the greatest a shot: http://www.pbxnsip.com/protect/pbxctrl-2.1.6.2437.exe
  16. That is caused by the fact that many devices cannot deal properly with multiple codecs in the SDP answer. 2.1.6 should improve this (if not fix this).
  17. I would set an outbound proxy on the trunk. Also, if you change the IP configuration, you should restart the service (just to be sure). If it does not help, turn SIP logging on and see what the PBX tries to send out.
  18. Are you sure? If should read out the two new messages (in "historical" order), then the 10 saved messages ("in historical order"). If you have a lot of messages, then the historical might be an issue because you have to go through all of the messages before you get to the latest, but IHMO that is okay because as a policy.
  19. That sounds extremly high to me. Does that number go down after a restart of the service?
  20. Error 1/03/2008 09:34:17 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/04/2008 04:15:35 PM Service Control Manager None 7031 N/A PBXNSIP Error 1/05/2008 11:00:10 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/05/2008 11:29:58 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/11/2008 10:32:54 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/11/2008 11:24:39 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/18/2008 02:36:33 PM Service Control Manager None 7031 N/A PBXNSIP Error 1/18/2008 04:32:31 PM Service Control Manager None 7031 N/A PBXNSIP Error 1/30/2008 11:06:43 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/31/2008 09:48:51 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/31/2008 10:24:44 AM Service Control Manager None 7031 N/A PBXNSIP Error 1/31/2008 04:17:22 PM Service Control Manager None 7031 N/A PBXNSIP Error 2/04/2008 10:34:08 AM Service Control Manager None 7031 N/A PBXNSIP Error 2/04/2008 10:43:59 AM Service Control Manager None 7031 N/A PBXNSIP Whow. THe first thing that comes to my mind would be memory. Is there any way that you can monitor how much memory the process took? The patterns above look to me like it is working for about a week, then it crashes (restarts) and then goes into a weekly cycle.
  21. Yea, the fast forward and rewind have to deal with the problem that the codec compresses the audio stream depending on the previous and following frames. Jumping in the stream means that the context is missing, and that leads to bad autio. Check your cell phone when you are driving in a tunnel... The fast forward and rewind are just to locate right position in the message, so I would not give this problem too high priority.
  22. Fro FAX with T.38, both the PSTN gateway and the FXS gateway needs to support it. If you are using a ITSP, then I would suggest to ask them if their switch supports T.38. As for Fax to email, there is a softphone out there (name?) what supports T.38 and fax-to-email. Maybe someone remembers what the name was.
  23. Maybe we really need to expire DND, for example every night at 12 o'clock. But I am not sure if that would have some negative side effects. It is really difficult to find out that you are on DND in SIP, there is no standard that would indicate this.
  24. If you want to receive a CDR email after each call, that feature is in 2.1.6 available.
  25. You can check the extension status - is it on DND, are there registrations? Log in as user through the web interface and see your own status.
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