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ap_6200

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Posts posted by ap_6200

  1. I'm not sure where the problem exists - Can you reproduce this problem? I believe it should be easily reproducible.

     

    Please let me know if I need to open a trouble ticket on this for faster response - This is critical to our upgrade path from Windows to Linux - and we have 7 Feature Servers with multiple domains to backup and move.

     

    Thank you.

  2. Hmm. The only thing that comes to my mind are case-sensitive file names. Anything in this direction? The other things are backslashes - anything there in the XML files?

     

    Maybe we need a short script that patches the XML files accordingly.

     

     

    I'm not sure where the problem exists - Can you reproduce this problem? I believe it should be easily reproducible.

  3. The problem is the underlying data model. If create a service flag, it will get a number. Then if you reference it from the extension web page, everything is fone. When you delete the service flag and create a new one, that one has a different number, even if the name is the same. Then the reference gets broken.

     

    That is perfectly normal to a computer programmer, but not for a end user. We think using more Ajax can address this and make end user's life easier, but we want to stay away from hardcoding the internal database relations and resolve conflicts.

     

     

    I just zipped a domain from one FS running 2.1.6.2450 (Win32) and moved it to a new FS running 2.1.8.2463 (Linux) - Most things worked correctly - however the voicemail Greetings were gone, and must be re-recorded. We experienced this on a domain zip - move from Win32 to Win32 as well. Could you please identify the problem and implement a patch fix? We need this feature, as we are rolling to Linux - we have to zip and move.

     

    Thank you for your urgency.

  4. Try hitting the save button again. Maybe the reference got broken somehow, saving it would restore it.

     

     

    This appears to be a reoccuring theme in the latest releases - Do you believe this will be rectified prior to future releases, or will be something that we will have to do for each upgrade to ensure the references are stable?

  5. 2.1.8 is available. The only thing changed was a neccessary fix in the camp on feature. Workaround is to disable camp on or upgrade to 2.1.8. See the release notes at http://wiki.pbxnsip.com/index.php/Release_Notes_2.1.8.

     

     

    I'm using 2.1.6.2450 - and I could not duplicate the issue fixed in 2.1.8.

     

    I also tried 2.1.7 - but still could not duplicate.

     

    i'm trying to determine if I need to upgrade from 2.1.6 to 2.1.8 to avoid this situation.

     

    Please advise.

  6. Well, there are two things.

     

    First, there are two places for settings the SMTP server. One is on admin level, and there is another one on domain level.

     

    The other thing is that once that emails are being put into the spool directory, the email server is fixed. That means, you might have to manually delete the emails from the spool directory to get them out of the system.

     

     

    Off topic slightly - How often are the Admin status reports (the graph) - scheduled to send out? I seem to get them every other day or every 3rd day.

  7. This is a serious message because the NAT router is not really ready for VoIP. There are some routers on the market that change the port during re-registration, leaving a blind spot for the reachability for the registered device. In the above case, the registration changes every 90 seconds - don't be surprised if calling that phone will result in random call drops.

     

     

    Can you define what "not ready for VoIP" means? We have not seen this message in previous versions of pbxnsip.

  8. I was thinking if we could use one of the A-D DTMF codes, which are not very known. Then we could block the relay. But I have no idea on how to program a key on the Polycom so that it sends out such a code...

     

     

    I am in dire need of an alternate solution - if PBXnSIP or anyone in the community can provide an answer, I'd definitely appreciate it.

  9. Well, how would the phone interact otherwise? You could try INFO messages, but then the pass-through of DTMF to external applications will suffer. The Record button solves that problem, if available (AFAIK only on snom).

     

     

    I am using Polycom's; therefore no Record Button.

     

    Is it possible to just send the DTMF to the PBXnSIP FS, and not to the end user? Especially when it's preceded by a * code ?

  10. Oh. In theory they should be the same. In 2.1 there is more logging, but that should not have any side effects.

     

    It appears that at least upgrading from 2.0.3 to 2.1.6 - you have to (at a minimum) re-add the dial plan entries (or hit save) with the reg exp entries.

  11. The statistics are in the web interface a snapshot of the current day. They are reset on midnight, just after the email has been sent out (if enabled).

     

    Who pickup up the call cannot be seen in the web interface. However the CDR contain that information, it is in the "tuser" field.

     

    I realize that the stats are reset... what i'm saying is they're now not gathering because of the setting change i made above.

     

    Can you put the "tuser" field in the web interface CDR?

  12. Usually these strange effects happen when that user 260 should be called (for whatever strage reason) and either that extension is not registered any more or the number of available lines to that extension is exhausted. That strange reason could be a re-INVITE with a changed from/to-tag, so that is technically is another dialog and treated as such by the PBX.

     

    I'm not as concerned with the 2nd entry into the CDR as I am - how and why a voice recording of the far end caller showed up in her email as a voice message?

  13. I'm confused on something:

     

    Three days ago, I was getting Agent Queue status emails like this:

     

    Overview on the status of queue Agent Queue on 2008/04/01

     

    Agent Name Availability DND Calls Duration

    101 Office Logged Out Off 0:00:00

    102 Office Logged Out Off 16 0:06:28

    103 Office Logged Out Off 10 0:06:02

    104 Office Logged In Off 21 0:14:30

    105 Office Logged Out Off 37 0:16:31

    106 Office Logged Out Off 0:00:00

    107 Office Logged Out Off 50 0:39:39

    110 Office Logged Out Off 0:00:00

    111 Office Home Logged Out Off

    Hangup while waiting 139

    Hangup while ringing 40

     

     

    I made a change to the Agent Queue to Event for connecting the call: from "Entering the Agent Group", to "Agent Picks up Call"

     

    And now my emails are like this:

     

    Overview on the status of queue Agent Queue on 2008/04/03

     

    Agent Name Availability DND Calls Duration

    101 Office Logged Out Off 0:00:00

    102 Office Logged Out Off 0:00:00

    103 Office Logged Out Off 0:00:00

    104 Office Logged In Off 0:00:00

    105 Office Logged Out Off 0:00:00

    106 Office Logged Out Off 0:00:00

    107 Office Logged Out Off 0:00:00

    108 Office Logged Out Off 0:00:00

    110 Office Logged Out Off 0:00:00

    111 Office Home Logged Out Off 0:00:00

    Hangup while waiting

    Hangup while ringing

     

     

    So that's question 1: - Why does this affect my logs?

     

    Question 2: - I want to be able to know, by looking in the call logs - which of my Office Agents picked up a certain call. (Currently, if you are monitoring the Calls in the Active Agent Queue, it will tell you who answered and is on the call) - However on the main call logs (historical) - it does not indicate who picked up the call.

     

    Thanks for the input.

  14. I do not have logs to back me up; but I upgraded from 2.1.5.x to 2.1.6.2450 on Wednesday and today I had an occurence I haven't seen in the year I've been using PBXnSIP

     

    A caller on extension 260 called an external number... and when she got done - her voicemail indicator light came on.

     

    When she checked it, it was the external numbers voice only in a recorded session. What was strangest about this - It wasn't the Registered User's voice at all, but you can hear apparent breaks in the conversation, as if the Registered User was talking

     

    Things I've checked:

     

    Call Log:

    2008/04/04 10:09:33 Registered User (260) 5554441111 01:36

    2008/04/04 10:08:54 5554441111@x.x.x.x Registered User (260) - 260 02:17 M

     

    I confirmed with the user that they didn't receive a call from the user at the time specified above - this was conceivably the same call, but it showed up as 2 separate entries.

     

    Record Capabilities:

    Default is OFF on the domain

    All settings on the Registered User are "DEFAULT"

     

    I have no idea how to reproduce this - as it's obviously not every phone call that is placed...

     

    Anyone have any experience with this in testing the new version?

  15. Why is it that I can not access a conference room from the auto attendant?

    If I call from my mobile phone, or if I dial extension 70 (our auto attendant) and then dial 800 (our conference room) it returns "This extension does not exist".

     

    If I dial 800 from my soft phone, it dials into the conference number as expected.

     

     

    I believe this would be related to sending a call directly to voicemail from the Auto Attendant... 8 is the default.. So most likely it's trying to send a call to Extension 00's voicemail.

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