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Unity stops recording message before the end

borisdecout
Level 1
Level 1

Unity 3.1.5, CCM 3.2.3, Exchange 2000

Once in a while, unity stops recording the message before the user has finished. The message delivered is truncated and is missing the ends. This is not a recording limit as the cut doesn't happen at the same time in the message. Sometimes after 10 seconds, sometimes after 40...)

There is nothing recorded in the events logs and this happens to internal call as well as external.

Any idea what could cause this and how I can troubleshoot the issue? This doesn't happen really often as only a couple of users have reported the problem.

Thanks

-Boris

15 Replies 15

evonnorm
Cisco Employee
Cisco Employee

We need some more information to help diagnose this.

1) I see that it happens for both internal and external callers. Is there any transcoding (ie, are some callers in G.729 and some in G.711 regions) in the network, and does it only happen on callers for which there is transcoding? Are there gateways to the PSTN or other phone networks, and does it only happen for calls over gateways? Is CCM the only phone integration or is there also a circuit-switched PBX?

2) Have you received any reports from the callers how got cut-off, or just from the people who've received the truncated message? If you have received reports from people who were cut-off, did they say if Unity had hung-up on them or if Unity had stopped recording and gave them a prompt? Or, similarly, have people who got cut-off called back and said, "Unity hung-up on me or interrupted me and started playing prompts"? That matters, since what likely happened is that either Unity thought there was disconnect tone (which would cause the hang-up issue) or Unity thought there was silence (which would cause Unity to stop the recording and play a prompt to the user). Sometimes, a woman's voice might get interpreted as a disconnect (since many women have voices that are at a similar pitch to telephony tones), or sometimes a poor connection over the PSTN or from a cell phone might fade out and get cut-off. Also, sometimes silence can be incorrectly detected if people speak softly or record messages over a speaker phone.

Basically, we need to try to narrow this down a bit more. There are Unity traces that you can turn on which will show why each recording stops (caller presses a digit key, silence occurs, disconnect occurs, or recording limit is reached). However, that's really only useful if you know exactly when a caller was cut-off (meaning, they call and tell you exactly when) - it's often difficult to figure out which recording it was if they only report you get is from the person who receives the message. Also, if you can figure out a pattern (ie, calls over a gateway, between codec regions, etc.), then you can try to reproduce it on demand, and at that point try to get traces.

1) There is no transcoding. We use G711 everywhere.

Only three users have reported this problem so far. It was either internal calls or coming from the PSTN on a 6608.

We only have callmanagers. No other PBXs

2) I talked to one of the caller and she said the system let her talk and she never realized she got cut off.

-Boris

So far , I don't see a pattern between so three users that I could try to setup to reproduce the problem. All three messages I reviewed were from women.

miked
Level 1
Level 1

Has this issue been solved. I see no more posts to this issue and we are experiencing the same problem at a new install. Very random but very real. I ahve checked all the basic stuff and i was wondering where else to go. The set up that you detail is exactly what we have here.

nope. Never been fixed. We have since upgraded CM to 3.3.3sr2 but it didn't help. The problem is with unity and I guess we have to upgrade to 4.0.3 to attempt to resolve this (maybe). It doesn't happen enough for us to be critical but it is still annoying.

All tac could tell me is that sometimes women voices have a frequency that unity interpret has a disconnect or something like that. Unfortunately it happens also on men's recording.

Let me know if you make any progress on this

-Boris

Try this:

http://www.cisco.com/warp/customer/788/AVVID/vmdisco30sec.html

I had the same issue with 7750. It appears to be an mgcp issue, and since you are using 6608, I bet it will fix you up too. I am sad to read that TAC did not help you, especially when the answer is in their website.

Anyway, good luck.

Joe

Thanks for the info but sadly it is not what we are seeing. In our case , the message is cut off at anytime. I've samples cut off after 8 , 43 and 94 seconds. It is also not reproducible easily as it only seems to happen once every hundreds of messages.

raul_maldonado
Level 1
Level 1

Hello,

Have you solved this problem? if you have please let me know the solution because i am having exactly the same problem.

Regards.

trevor.hanekamp
Level 1
Level 1

We had the same complaints after cutover and it turns out that it was happening when people paused for 2 seconds. We increased the Long and Short Recording settings to 6 seconds and the complaints stopped. These settings are found under System>Configuration>Recordings.

EJRamstad
Level 1
Level 1

I have read all the responses to your message through today. We have been having this same issue since our first report in early 2001. It affects an unknown number of users and TAC has been working extra time since early December on it for us. I have been retrieving the cut off messages using the Outlook web access and forwarding them with their original date/time stamp to TAC to assist in setting up traces and or establing patterns. Seems to happen primarily, possibly only to women, still trying to pin that theory down. The one group of subscribers particularly hard hit have no problem when we had them switched over to Qwest voicemail, our local CO.

We have a Centrex integration and have upgraded from our original server, dialogic boards and Unity 2.4.6 to an entirely new system with 4.0.2 . The server is setup for voicemail only. All the pause times have been extended and are looking primarily at the integration at this time.

Am anxiously awaiting some word and hope that our resolution will help others. If anyone has any additional thoughts on this that would help TAC it would be appreciated.

Regards,

Erik

We did have the same problem here, the immediate solution was to set TrimDisconnectTonesOnRecordings=0 in [Configuration] in the INI file for the PBX (in our case Avaya0002.ini)

The reason we had originally set that value to 1 was that we got periodic wave-off tone messages when callers abandoned their call upon hearing Sandy's lovely voice.

It seems that LearnTones determined that a certain cadence in some caller's voices (mostly female) was being interpreted as a PBX disconnect tone, so Unity was dutifully trimming the remainder of the message after hearing that tone.

We have an ongoing open TAC case on this issue as well -- but focused on the abandoned call issue, as the cut-off message issue is no longer a problem.

Just for the official record - despite my earlier posting, this problem is NOT resolved.

We thought we had it licked back in December with this TrimDisconnectTonesOnRecordings setting, but we STILL have an ongoing TAC case open - for about six months now with no resolution in sight. TAC has many megabytes of trace logs and examples of messages, but appears stumped.

If anybody in the user community has an idea on how to fix this we are all ears.

To regroup:

Unity 4.03(SR1), Exchange 2003 (was 2000/5.5 earlier) integrated with an Avaya S8700 (was a G3R earlier). We have upgraded Exchange, and the PBX itself since this started with no effect on the problem. We are running a PBXLink with firmware version 4.4.

Callers, mostly women, both internal and external, are randomly hung up on by the voice mail 'system' (don't know if it is the Dialogic or the Unity piece) while they are speaking. This happens mid-word, on messages with no pauses or hesitation. The length of the stubby message varies from just a few seconds to well over one minute before they are disconnected.

Certain callers are repeatedly victims, others not. These folks have both very loud and very soft voices, some speak slowly and others quickly - in short I have not identified a common characteristic between them.

Please send your ideas!

What do you have for MinLoopCurrentOff= in your current active switch file?

Also, can you post the [Switch Disconnect Tone] and

[CO Disconnect Tone] definitions from the switch file?

With the Avaya PBX you should be getting loop current drop signaling real disconnects ("Adjunct Supervision"). Then if that is missed you should get busy tone ("Station Forward Tone Disconnect" set to busy). So the odds of missing a real disconnect are pretty slim, which means you probably can safely tighten down the parameters quite a bit.

Since you indicate that the hangups primarily occur while the caller is speaking, this would suggest that the problem might be a false positive on disconnect tone detection. You can tightening down the two disconnect tone defintions by setting the ...deviation parameters as small as 10. Or you might even going as far as frequencies to make the tones nearly impossible to match (i.e. rely entirely on the Adjunct Supervision and don't do any tone based disconnect).

If you suspect false positives on the loop drop detection, then bumping up MinLoopCurrentOff= may help elimite false positives there. Knowing exactly how many ms of loop current break your PBX is expected to provide would be helpful here.

-Eric

Great info - thank you!

Currently:

[Switch Disconnect Tone]

Frequency1=350

FrequencyDeviation1=100

Frequency2=440

FrequencyDeviation2=100

TimeOn1=4000

I do not have a CO Disconnect Tone section in my switch.ini file. Should I have one? During troubleshooting my switch.ini file has gained and lost this section a couple of times.

MinLoopCurrentOff is set to the default '300' now. What value would you recommend?

I will first try tightening down the deviation parameters on the switch disconnect tone, then look at relying entirely on the Adjunct Supervision if that does not work.

Ok, I'd start with dropping the Deviation values in that tone down to 40 or even 10. You should also put something in for the CO Disconnect Tone. If nothing is explicitly defined then a generic cadence tone will be used. The generic tone definition tends to be pretty loose as it'll catch reorder and busy in most countries. If you do have "Station Forward Tone Disconnect" set to busy, then I'd recommend specifying a standard US busy tone...

[CO Disconnect Tone]

Frequency1=480

FrequencyDeviation1=40

Frequency2=620

FrequencyDeviation2=40

TimeOn1=500

TimeOnDeviation1=40

TimeOff1=500

TimeOffDeviation1=40

Cycles=4

For now, the MinLoopCurrentOff= 300 should be fine. Let's see how things go with small deviations on both disconnect tone definitions.

Also, for further reading, there's a good article about Unity and tone defintions here... http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a0080093c1e.shtml