ap_6200

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.

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

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.

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?

Did this happen in one of the latest releases?

Could I get an answer to this please.

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.

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.

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

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.

You are correct - it's the status page as a user. Is this detrimental at all? I've had 2 servers out of 6 on 2.1.6.2450 that i've upgrade restart the services for no apparent reason. I'm very concerned.

I don't know when it happens... I'll try and reproduce... Do i need to log in as a user?

What is this? [1] 2008/04/09 20:00:09: Unknown SSI command dbh [1] 2008/04/09 20:44:11: Last message repeated 140 times Utilization also spiked to almost 100%

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 ?

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.

Is it possible to send the *12 (Call Record On) - without the DTMF? Currently users can hear the touch tones, and ask if something is wrong with my phone. Thank you for any work arounds that can enable Call Record seemlessly.

No, I'm referring to Dial Plan Entries, not Trunk settings.

from 2.0.3 to 2.1.6? Did an upgrade over the weekend to 100 plus customers, and had to revert to using the xxxx's instead of the REG EXP. Can anyone elaborate on the changes?

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?

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?

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.

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?

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.

Would someone be able to post how to build an IVR Node and get it implemented as a sub-auto attendant so that I can utilize the pause (#) in the direct dial in the Main Auto Attendant? Thank you

Can you elaborate? I'm contemplating rolling 2.1.6.2550 this weekend - I need a recommendation please.