Re: Unity Hanging Up On Customers Leaving Messages

pjt8537 · ‎01-12-2005

Running Unity 4.0(4)SR1 in a dual integration environment (CM 3.3.3 and Avaya G3R V11). On the PBX side intermittently when customers are trying to leave voicemail the system is just hanging up. We have run the learn tones utility, but still seem to have this issue. I have heard that some of the parameters can be adjusted manually to help get rid of this. Any ideas on what to adjust and which way.... Here is what our avaya001.ini file looks like...... Thanks! Paul

; $COPYRIGHTSTART ********************************************************

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; This product is protected by one or more of the following US patents:

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; and foreign patents pending.

;

; Unity and ActiveAssistant are trademarks of Cisco Systems, Inc.

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; Cisco Systems, Inc.

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; $COPYRIGHTEND **********************************************************

;

; NOTE: Enter values for "DelayBeforeOpening" and "RetryInterval" in seconds.

; Enter all other timing parameters in milliseconds.

;

;***************************************************************************

[Identity]

SwitchManufacturer=AVAYA

SwitchModel=DEFINITY GX

SwitchSoftwareVersion=ALL

IntegrationType=Serial

[Configuration]

DelayBeforeOpening=0

Integration=SMDI

ConfirmReturn=

TransferInitiate=&,

BusyRecall=&,&

Recall=&,&

PauseDuration=1000

PulseBreakInterval=60

PulseMakeInterval=40

PulseDialInterDigitDelay=1000

ToneDialInterDigitDelay=50

DigitToneDuration=100

DTMFDebounce=40

MinimumRingOnInterval=100

MinimumRingOffInterval=300

WaitAnswerRings=4

MaxWaitBetweenRings=10000

MinLoopCurrentOff=300

RingbackTimeout=10000

IncomingCallRings=1

MinimumMWIRequestInterval=750

TrimDisconnectTonesOnRecordings=0

OutgoingPostDialDelayMs=0

OutgoingPreDialDelayMs=0

OutdialAccess=

HookFlashDuration=0

HangUpTone=

SerialAutoAnswer=1

WaitIfBufferedPacketPresent=1

[MWI Default]

Active=Yes

MWIType=Serial

SerialConfiguration=SMDI

UpdateDisplay=No

CodesChangeable=No

RetryCount=1

RetryInterval=5

PortMemory=No

LampOn=198

LampOff=197

[Integration]

SMDIPrefix1=

[Switch Dial Tone]

Frequency1=338

FrequencyDeviation1=40

Frequency2=451

FrequencyDeviation2=40

TimeOn1=4000

TimeOnDeviation1=0

TimeOff1=0

TimeOffDeviation1=0

TimeOn2=0

TimeOnDeviation2=0

TimeOff2=0

TimeOffDeviation2=0

Cycles=0

LearnSamples=5

LearnDelay=1500

[Switch Busy Tone]

Frequency1=479

FrequencyDeviation1=40

Frequency2=620

FrequencyDeviation2=40

TimeOn1=490

TimeOnDeviation1=40

TimeOff1=480

TimeOffDeviation1=40

TimeOn2=0

TimeOnDeviation2=0

TimeOff2=0

TimeOffDeviation2=0

Cycles=0

LearnSamples=5

LearnDelay=1500

[Switch Ringback Tone]

Frequency1=427

FrequencyDeviation1=61

Frequency2=502

FrequencyDeviation2=49

TimeOn1=1160

TimeOnDeviation1=140

TimeOff1=4090

TimeOffDeviation1=410

TimeOn2=0

TimeOnDeviation2=0

TimeOff2=0

TimeOffDeviation2=0

Cycles=0

LearnSamples=5

LearnDelay=1500

[Switch Disconnect Tone]

Frequency1=330

FrequencyDeviation1=40

Frequency2=457

FrequencyDeviation2=40

TimeOn1=4000

TimeOnDeviation1=0

TimeOff1=0

TimeOffDeviation1=0

TimeOn2=0

TimeOnDeviation2=0

TimeOff2=0

TimeOffDeviation2=0

Cycles=0

LearnSamples=5

LearnDelay=1500

[CO Disconnect Tone]

Frequency1=333

FrequencyDeviation1=40

Frequency2=459

FrequencyDeviation2=40

TimeOn1=4000

TimeOnDeviation1=0

TimeOff1=0

TimeOffDeviation1=0

TimeOn2=0

TimeOnDeviation2=0

TimeOff2=0

TimeOffDeviation2=0

Cycles=0

LearnSamples=5

LearnDelay=1500

pjt8537 · ‎01-12-2005

Just an update to the above issue. From the cutoff messages that have been forwarded to me, most seem to be about 10 seconds in length and are very low volume calls. This may be why Unity is terminating the call. Is there a way to adjust the volume on these calls without adversely affecting others?

eschulz · ‎01-12-2005

Hmm, the very low volume makes me think there may be a line voltage problem on one or more of the analog lines going to Unity. If it were a case of just low volume (being falsely detected as silence) then callers would be presented with the after recording menu options. Instead, calls are being dropped, which indicates that Unity thinks the PBX has signaled a disconnect. The disconnect tone definitions in the switch file look fine so I don't think those are causing the issue.

You might try testing each line to Unity by calling it directly and leaving a long message. See if you can isolate the problem to a subset of the ports. I've seen cases where some lines from the PBX could not maintain good voltage. With a voltmeter spanning the line we could see the voltage start to drop several minutes into the call. Eventually the voltage would drop to the point where the Dialogic board considers it an open circuit and hence puts it end on-hook.

Also, if possible, distribute the VM lines across multiple line cards in the PBX. In some cases heavy loads to VM on a single line card caused problems.

Regards,

Eric

pjt8537 · ‎01-12-2005

Thanks Eric, I am learning more the further we go here. I have since heard back from the person leaving one of the voicemail messages that was cut off. The system did prompt them for the press 1 to continue or 2 to disconnect. From talking with others, this person is very soft spoken. Is there a way to boost the lower voice levels on the dialogic side?

eschulz · ‎01-12-2005

I've never had much luck tinkering with Dialogic's AGC parameters. Instead, you might try extending the silence thresholds on the Unity SA System Settings Page. This controls the amount of silence before Cisco Unity assumes the caller is no longer on the line.

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_administration_guide_chapter09186a008022d39e.html#wp1606309

pjt8537 · ‎01-27-2005

Hi Eric, we have tried a bunch of stuff to try to resolve this. It seems that it is better now, but we are still getting some of the dropped calls. After talking with some of the callers, and it appears that some it happens to regularly, that some are getting the prompt to continue leaving a message and some the system just hangs up on after the silence detection timeout is met. I have raised the dialogic median volume to 97 and the problem seems to be a little better. We have also extended the timeouts as you recommended, but that hasn't done anything except let the caller talk longer before it throws the message away. I do not think it is a line voltage issue as it seems to be isolated to certain callers with low voices. What I am wondering is if adjusting the Audio AGC: Minimum db threshold. Currently it is set at -45. If this would help, which direction should I go. Any other suggestions. This problem is getting alot of bad publicity. Any help would be greatly appreciated.

Paul

michael.armstrong · ‎02-21-2005

We had a similar problem when we upgraded to Unity 4.0(4) with CCM 3.3(3). Problem was the TSP 7.0(4), which had to be downgraded to 7.0(3b).

We're now having the same problem with CCM 4.1(2)/Unity 4.0(4)SR1 system, both with PSTN and on-net calls. It is intermittent, but seems to be getting worse, but maybe that's just because I've asked our users to watch out for it and report problems. Unlikely anything related to softspoken callers. I think I've got every trace and logging facility there is turned on, as well as packet traces to/from Unity, so it should be only a matter of time. TAC is working on it.

justincohen · ‎02-22-2005

Have you looked at The Cisco Quiet Parameter? I have a customer with same setup, same problem, modifying the Quiet Parameter in dialogic fixed the problem.

http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_installation_guide_chapter09186a008022b8d9.html#wp1046540

pjt8537 · ‎02-25-2005

I thought that that might help too. I opened a TAC case and we discussed the quiet file and they decided against changing it. What did you change it to to help your customer. I am not against trying it to see if it helps.

Thanks again for the input.

Paul

dori.griffith · ‎02-22-2005

Paul,

I'm having the same problem. I have 2 Avaya Prologix systems daisey chained to our Unity. I haven't fixed the problem yet but I'm keeping an eye on these posts for a fix.

A question for you....how many people are on your G3? I'm running into the problem where we don't have enough analog lines between the PBX and Unity so some callers rolling to voice mail get busy signals. Wondering if other VoIP/PBX mixed environments are having the same problem.

justincohen · ‎02-22-2005

Assume the following topology.... the PSTN connects to a VoIP gateway, so does your G3... So the VoIP system is frontending your PSTN stuff.. IF you are doing it that way then this works well, and has for me in the past.

The PBX only has 12 ports but has 300 users. The VoIP has about 100 users but has 24 ports.

Why the mismatch? Because on the PBX the 12th port is set to roll-over to the first port on the VoIP side. It works if you are using a protocol like DMS-100 that maintains the FWD tag when the call is sent across. QSIG not required.

So why don't you just put them ALL on VoIP? Well, then you will be using an aweful lot of the TIE lines for Voicemail calls, and calls may trombone alot.

Understand where i'm coming from? IT works well.

michael.armstrong · ‎03-01-2005

Here's the solution to our problem, FWIW: It turns out that a Unity Registry key (RTPPortBase) somehow (sinister forces, maybe) got set to zero. As the name implies, this appears to be a base address for Unity to use to assign UDP ports to incoming RTP streams. This value is incremented by 2 for each new stream, until RTPPortBase+28 has been used, at which point the cycle starts over again at RTPPortBase. (All this is guesswork on my part, gleaned from examining Ethereal traces. I suspect this value may be different in larger installations.) Thus, in our case, every 14th message failed, since Port 0 is apparently ignored or otherwise invalid. I have no idea how RTPPortBase came to be set to 0, or how long it's been that way. TAC advised setting it to 0x5910; that fixed the problem.

tdodenhoff · ‎03-08-2005

We had a similar problem here - running an Avaya G3V12. We actually changed settings on the Dialogic card - Misc - paramterfile = quiet50.prm spandti.prm, as the config files. That seemed to clear it up for us. Don't know if this will help.