zktech

Try doing turning on SIP Message Logging on the Advanced Tab of the Line Properties menu. Once you have the log menus check to see what is coming from the Call Centric when you get the key. It is likely an issue with Call Centric sending bad DTMF.

The servers are dedicated to just dial dictate. We changed the recording format to uncompressed PCM and the quality jumped through the roof. I am now doing a audio recompression pass on another computer. This gives us much higher quality than recording to the more compressed formats on the dial dictate servers. It appears that there are some major quality issues that are occurring when recording to some of the more compressed formats using dial dictate. For reference our Dial Dictate servers are as follows. Server 1 XP Pro SP 3 1U HP Dual Xeon 3.4GHZ 3GB or RAM Mirrored 80GB 15K Scsi drives Server 2 XP Pro SP 3 Mid Tower Intel Server 3.0 Dual Core 3GB or RAM Mirrored 120GB 10K Sata Drives

Sherry We provide a call in service for doctors that allow them to record their dictations and allow their transcriptionist to pickup their sound files for transcription.

At this point we have had to write our own code to do this kind of thing.

We are trying to print or export a list of all the users in the Dial Dictate users list with their respective information. We can not find any way of doing this. How can we do this?

We are running two dial dictate servers. Both are running on version. 4.20. We are having an issue where we are getting calls randomly bleeding into each other. This is very unusual as we are running SIP and the provider is recording the calls on their gateway and the calls are fine there. We are housing our servers in the same datacenter as the provider so our servers are directly connected into the providers network. Both of the systems are running windows XP SP3 and have more than 2 GB or ram. The CPU are never going above 20% so there should be no performance issues. Both boxes are dedicated to the dial dictate. This problem is occurring with two or more calls on a sip trunk. We need to find a way to fix this issue ASAP. Please help

We are now running Dial Dictate and have all of our configurations set correctly. The issue we are having is some of the audio files have distortion and echo in them. We are running from a PRI gateway to our Astrisk PBX and the DD system registers with the Asterisk server. All calls are being recorded on the Asterisk server as well. When we find a file that has an issue on the DD system we go listen to the one on the Asterisk server and it does not have the flaws in it. Does any one have any idea why DD would distort files durring the recording process. We are recording to GSM 11.000 format? The system is a Dual Processor Xeon box with mirrored SCSI drives and 3GB Ram running XP Pro SP3 no other applications are on the box Any advise is apperciated. Thanks zktech

Record a new value for the record tone. Just record the word "Recording" and see if you hear that. I have found their tone to conflict with most PSTN equipment. If you hear the new recording then you know their tone frequency will not work with your TAPI equipment.