06-17-2003 10:59 AM - edited 03-13-2019 12:23 AM
I have a Unity 3.15 VM system with E2K and all latest patches. CCM is running 3.22c with latest patches. When a caller is passed into Unity, the persons greeting is played and the caller is given the opportunity to record a VM message. Regardless of how long they speak, the message is cut off after 5 seconds and the caller is presented with message options. Basically, Unity is recording 5 seconds of silence. I have verified that the system doesn't think it is running in demo mode, the rainbow driver is ok and licenses show up. I can navigate all menus within Unity, so DTMF tones are being passed, I can hear the users greeting, so voice passes, just the callers message is not being recorded. Any assistance will be much appreciated.
Thanks!!!
Adam
06-17-2003 11:04 AM
What is your codec configuration on Unity?
What is the codec configuration on CCM for the regions calling Unity (& compared to Unity's own region?)
06-17-2003 11:13 AM
Thanks for the quick response. The regions on CCM between the IPPhones and Unity are set for G711, and Unity is using the default G711 codec. If this were a codec issue I would think that I would get disconnected immediately whereas I do hear all voice prompts from Unity and the users greeting. Is this not true?
Thanks!!!
Adam
06-17-2003 11:18 AM
Hi Adam,
That's right - I didn't consider your comment about dtmf's getting through.
Here's a troubleshooting doc on 'one-way audio':
http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a0080093c11.shtml
Hope this helps,
-Bill
06-17-2003 12:13 PM
Bill,
Thanks for the document. I have verified that there are not any wave record errors, Tx Packet count is incrementing while trying to record a voice message, and there are not any firewalls between the phones and Unity. However, for users calling into the system from the PSTN, they go thru an Avaya PBX that has a T1 trunk to our VG200 which allows access to the Cisco voice equipment. When I had a local user test though they called from an IP Phone that would have a direct connection to Unity thu the routed network, not having to go thru the PBX to access Unity. All devices on both sides of the routed network can ping Unity, and Unity can ping them.
Thanks!!!
Adam
06-17-2003 12:34 PM
Adam,
To clarify - you've found that a caller 'local' to the Unity (not going through the PBX to access Unity) is working as expected, but callers from the PSTN are having the problem stated? Or is this failing for everyone calling Unity?
-Bill
06-17-2003 12:41 PM
Bill,
All callers, regardless of whether they are local or coming thru the PSTN/PBX, cannot leave a voice message.
Sorry for the confusion.
Adam
06-17-2003 12:51 PM
Ok, let's get back to basics for the audio then:
How many NIC's are enabled on the Unity system? What is the duplex setting on the NIC(s)?
Could you check the Unity registry for the following key's value:
HKLM\Software\Cisco Systems, Inc.\Cisco TSP\WaveDBGainRecord
Thanks,
Bill
06-18-2003 09:01 AM
Bill,
The registry value in question is: 0x00000000 (0)
There are two built in NICs in the server, one in use and one is disabled. The NIC is set for 100Mbps Full Duplex, as is the switch port it is connected to.
Thanks!!!
Adam
06-18-2003 09:33 AM
Bill,
I found a similar reference under "external calls forwarded to unity " and his issue was a codec problem. However he was using H323 and I am using MGCP. What do you think?
Thanks!!!
Adam
06-18-2003 04:52 PM
Hi Adam,
If this is still impacting the callers, let's get a case opened with TAC and go for more troubleshooting there.
Thanks,
Bill
06-17-2003 12:41 PM
Hi Adam,
That's right - I didn't consider your comment about dtmf's getting through.
Here's a troubleshooting doc on 'one-way audio':
http://www.cisco.com/en/US/products/sw/voicesw/ps2237/products_tech_note09186a0080093c11.shtml
Hope this helps,
-Bill
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