ap_6200
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Posts posted by ap_6200
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Is there any update to this?
I recently discovered that 2.1.4 has inband DTMF issues - when calling certain 800 #'s - and using the Touch Tone Response systems - The system was not able to identify all of the digits.
We tested this same number from version 2.0.3.1715 FS, and from the test system 2.1.5.2357 and it worked fine.
Are there any outstanding issues with 2.1.5.2357 that i should be aware of?
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No, 2.1.5 is no difference regarding the performance.
My opinion is that hardware has become so cheap today that it is not worth having customers complaining about dropouts. So I would definitevely power up another server.
The new domain backup feature comes in handy. It should be relatively simple to move a few domains off to a new server!
So a thought could also be that the PBXnSIP software can't handle this load, as opposed to the hardware? We didn't buy cheap hardware - and have invested heavily.
Other suggestions?
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I have a Feature Server set up with 19 domains and a total of 390 Accounts across those domains.
I am seeing a consistent day-time call volume of 15-25 concurrent calls using 2.1.4 until 2.1.5 can be further tested:
Calls: 11323/2587 (CDR: 4124) 20/38 Calls
SIP packet statistics: Tx: 2426674 Rx: 2326830
Emails: Successful sent: 438 Unsuccessful attempts: 122 (Warning: Last email could not be sent!)
Uptime: 3 03:13:36 (77MB/2047MB 20% 43010880-0) WAV cache: 3
The media CPU usage during the day is between 25% and 50%... Ususally coinciding with the Call Objects.. Meaning at this time - the utilization would be at 38% (based on the graph)
Have I reached the theorhetical maximum? Are there any other customers or forum-posters with this many domains/accounts - and concurrent calls?
Complaints are coming in referring to static, audio cut outs, etc...
Is 2.1.5 any better, or do I need to start moving customers off of this feature server?
Please advise!
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Conversely - Can the PBXnSIP Dial Plans be manipulated to strip digits?
For Example - Someone wants to dial more than 11 digits in North America - Say a 1-800 number that is spelled out.
1-800-PBX-NSIP-FORUM
Is there a way to take the pattern of more than 11 digits and replacing them with the 1st 11 digits dialed? Effectively ignoring -FORUM from my example.
Thanks
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Okay.
Anything in the log like: "Could not find call of user ... that should be parked", "The call of user ... cannot be parked", or "The call of user ... cannot be parked as there was no free orbit available in ..."?
I have fixed this issue:
The procedure with one line only is different than multlines:
For example
1 Line:
Active Call - Place Call On Hold - Select "New Call" - *85 - Send (or Dial)
Multi-Line
Active Call - Select TRANSFER - *85 - Send
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I still don't get it. Lines equal lines in the registration tab of the extension in the PBX?
Yes:
Example:
Extension 900 - Registered 2 Regs
Extension 901 - Registered 1 Reg
Extension 902 - Registered 2 Regs
On ONE Polycom Phone - 2.2 firmware and PBXnSIP 2.1.5 -
I have Line 1 - 901
Attempting to place any active call on CALL PARK using the *85 feature code results in a successful "CALL PARK"
On ONE Polycom Phone - 2.2 firmware and PBXnSIP 2.1.5 -
I have Line 1 - 900
I have Line 2 - 902
Attempting to place any active call on CALL PARK using the *85 feature code results in a "CALL PARK FAILED"
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You mean, the phone sends a star code to the PBX? And that results in the "call park has failed"? Is it a phone config problem or a problem of the PBX handling the star code?
I mean that when we dial *85 - on any phone with only 1 line configured - Call Park works perfectly.
On phones with multiple lines (2 or more) - The Call Park fails using *85 ( or even *85xxx to park it to a specific extension)
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Call Park – Multiple Lines
I’ve found an issue where Call Park fails when you have multiple lines registered on a phone (happened to be a Polycom 501) on SIP version 2.2.0.0047. Going through the normal process to park a call produces the "call park has failed" message.
Is this a known bug with this version or not? Does anyone know of any workarounds? This could cause a lot of problems down the road. Thanks!
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So if I understand this correctly:
I have a Hunt Group - Setup to ring 2 phones concurrently (SIP PHONES) - Which so happen to be CFA'd to external numbers.
The call comes in and then is disconnected immediately.
I take it this is the expected response?
Is there any way (through HG, AA, or Extension Redirects) - to get a single call to call 2 external numbers concurrently?
Thanks
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I experienced this as well when I upgraded to 2.1.3
Evidently there were some file deletion bugs that this was attributed to... So more likely than not - these were old voicemails that were deleted, but showed up again after the upgrade.
That's my guess.
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The 2nd of 3 production servers just stopped the service today (2nd day in a row) -
There are no daily CDR's configured on this system - and it's not midnight - there is something else going wrong.
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OK - On to way more important things - 1 of 3 production systems restarted during the middle of the day yesterday - and my test system stopped at 12:02 - last entry in the log - "CDR Email Sent"
Please advise - I can't keep this in production if it's going to be unstable.
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The ad-hoc conferencing options are not working:
Authentication:
Conference Name:
Mode: Ad-hoc conference Scheduled conference
Before entering the conference: Just play a short tone or Ask participants to leave the name
Moderator Access Code:
Participant Access Code:
Number of digits for access code (4-9): 4 5 6 7 8 9
Dialog Permissions:
Ask participants to leave the name does not work - it only plays the short tone option.
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Not sure if this is the place to request it - but customers with multiple Agent Groups and Auto Attendants have repeatedly asked me for the ability to put a description (in parantheses) - on the Accounts main page - to be able to differentiate between them.
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Can you put a posting on the forum for ALL known issues/bugs with 2.1.4.2345 - and what is going to be rolled into 2.1.5 (i.e. Call Teach Mode fix)
Thank you
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In the 2.1.5 (upcoming) we fixed it so that it will automatically move to rtp_decode mode...
Is there an expected release date of 2.1.5?
Thank you
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These features are still in there, and they now can even contain DTMF sequences to make the recording start less accidential. Maybe the key that you have does not include the record feature.
You can for example use www.opinionatedgeek.com/dotnet/tools/Base64Decode to decode the key - check if the record flag is there an set to true.
Thanks - that was it.
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Can you try to set the setting "allow_pass_through" to false?
You can do this by appending "?save=save&allow_pass_through=false" to the web URL when you are logged in as admin (e.g. http://192.168.1.2/reg_settings.htm?save=s..._through=false).
Yes, that worked -
Will that be rolled into 2.1.4.2345 or the lastest release?
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Record On Key (e.g. *):
Record Off Key (e.g. #):
Record Location:
These settings were available in 2.0.3.1715 (Win32)
They are not available in 2.1.3 - Or did the location move?
Please help - I've sold this to a new customer...
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Is there a latest version of 2.1.4 - I found the domain deletion "feature" in 2.1.3
Thanks
Nevermind - I found it on the Download page - thanks.
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Ouch. Very good catch!
Is there a latest version of 2.1.4 - I found the domain deletion "feature" in 2.1.3
Thanks
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Please let me know the status quickly - as we wanted to release 2.1.3 in January to our customer base.
Any Update?
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Seems like latency issues between the phones and the server. For example, they tell me that they'll pick up a handset on a ringing phone and it will continue to ring for a few seconds before the caller is connected.
We have two locations about 200 feet apart connected using a wireless bridge using the same PBXnSIP server. Most of the complaints are indeed coming from the folks across the wireless bridge, but the ping times are consistently great - 1 and 2 ms, even under load. We currently have a call into them to make sure that the bridge is setup optimally.
Can you directly correlate the upgraded 2.2 versions to the issues you are experiencing? Does the user experience the same issue on a phone with an older version of SW? 2.1 or 2.0 perhaps?
It is this vital field testing that will help multiple others in similar scenarios.
Please Let me know the outcome of this testing - As we are preparing to roll this version (2.2) out to the field in Q1 08
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I would vote to keep them as they are. This way there is more flexibility in the product.
If learning how to use PBXNSIP is the only requirement to benefit from its features, then learn it.
That is the difference between a tech and a good tech. The good tech will always push him/herself a little bit harder knowing that he/she is about to learn something new.
Please keep flexibilty in the product.
Actually, I would like to see the Instant Message method used in parallel with Email notification for things like stations/trunks that fall out of registration
and other Management Notification events.
Just my thoughts.
Bill H
I also like the flexibility - I'm testing functionality and intended design... I believe there should be ADMIN emails (CPU Utlization and Logging) - As well as individual Domain Emails - CDR, etc...
GUI / Operator Console
in PAC and WAC
Posted
ditto