mike@cmd.net

Melody, Your problem is not the sme as you are using analog phone lines. However what you need to do is to increase your "recording continous auto stip time. 450 is about 7 1/2 minutes after which DD will automatically go to pause mode and read out the messageyou indicated. Set your recording continuous auto stop time = Number of minutes you want as the limit X 60. for example setting it to 1200 will allow up to 20 minutes before going to pause mode. For ALL, I'm happy to let you know that NCH support was able to have their developers come up with a fix. I recieved pre release of fixed version (4.13) this mornning. My intial testing was to dictate 12 minutes of continous recording without any issues.

I recently switched from using dialogic analog phone lines to Dial Dicate 4.12 with VoIP channels from Callcentric. I've experienced the same problem. I have worked with Callcentric tech support and based on their review of the VoIP log they have concluded that the issue stems from "one-way audio" while dictation is in progress. The VoIP providers to insure the line is not erroneously open if they don’t receive audio or a “RTP keep alive packet” they terminate the connection after certain time period (in our case 5-6 minutes). See below for excerpt of their last response to me. I followed their recommendation to set the “locate the ‘Record voice activation level (dB)’ and set your voice activation level to -30dB” That did not work. Then I shortened the “Recording continuos auto-stop time” to 320 (just before the disconnect time limit) allowing Dial Dictate to pause and read out the recording “pause mode – press 9 to continue recording”, causing the two-way audio. This keeps the connection and the dictation can continue as long as the caller gets interrupted every 5 minutes and presses 9. BTW, during this test I have resolved that “your voice activation level” setting does not work. Meaning if in pause mode if you start dictating, Dial Dictate does not start recording automatically. My Recommendation: As seen in the VoIP log below Dial Dictate upon initial INVITE sends the “keep Alive Packet” to both servers. Notice “l:348” (this is exactly the timeout period were the server expects to receive some packet from Dial Dictate). I think Dial Dictate should transmit one of the following after 5 minutes (300 sec): 1. Send out “RTP Keep Alive packets” to both servers. 2. Send out a tone. Hope this helps to resolve this problem. ================================================ Thank you for providing your SIP Trace logs onto your account, from your logs, as an example to show the issue, we are taking one call in which abruptly ended after 6 minutes: Here is the initial INVITE packet: 11:55:01 NAT Keep Alive Packet Sent to stun01.sipphone.com 11:55:10 NAT Keep Alive Packet Sent to callcentric.com 11:55:12 UDP Packet Received from 204.11.192.23:5080 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< INVITE sip:17772979347@71.98.252.64:5070 SIP/2.0 v: SIP/2.0/UDP 204.11.192.23:5080;branch=z9hG4bK-2e15492790d6e055ec04bbe2f597f21f f: <sip:18139266770@66.193.176.35>;tag=3422966101-177815 t: <sip:18139254118@ss.callcentric.com> i: 499973-3422966101-177765@msw2.telegeny.net CSeq: 1 INVITE Max-Forwards: 13 m: <sip:41b874f94b495a5eece665f36a24f316@204.11.192.23:5080;transport=udp> Supported: timer c: application/sdp l: 348 v=0 o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.23 s=sip call c=IN IP4 204.11.192.23 t=0 0 m=audio 58952 RTP/AVP 18 0 8 101 a=fmtp:18 annexb=no a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - a=setup:actpass Here is the call being established: 11:55:13 UDP Packet Sent to 204.11.192.23:5080 >>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>> SIP/2.0 200 OK Via: SIP/2.0/UDP 204.11.192.23:5080;branch=z9hG4bK-2e15492790d6e055ec04bbe2f597f21f To: <sip:18139254118@ss.callcentric.com>;tag=7709 From: <sip:18139266770@66.193.176.35>;tag=3422966101-177815 Call-ID: 499973-3422966101-177765@msw2.telegeny.net CSeq: 1 INVITE User-Agent: NCH Swift Sound Dial Dictate 4.12 Contact: <sip:17772979347@71.98.252.64:5070> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Accept: application/sdp Supported: replaces Content-Type: application/sdp Content-Length: 263 v=0 o=NCHSoftware-DialDictate 1213973312 1213973316 IN IP4 71.98.252.64 s=Dial Dictate Call c=IN IP4 71.98.252.64 t=0 0 m=audio 8014 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv And here is when you call drops after about 6 minutes: 12:01:42 UDP Packet Received from 204.11.192.23:5080 <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< BYE sip:17772979347@71.98.252.64:5070 SIP/2.0 v: SIP/2.0/UDP 204.11.192.23:5080;branch=z9hG4bK-19a88ec108849e13ded9e0276a9eb8ee f: <sip:18139266770@66.193.176.35>;tag=3422966101-177815 t: <sip:18139254118@ss.callcentric.com>;tag=7709 i: 499973-3422966101-177765@msw2.telegeny.net CSeq: 2 BYE Max-Forwards: 13 l: 0 To address what going above, as we have mentioned previously, your calls are abruptly dropping because our system detected the call as being a call experiencing one-way audio. (this helps prevent issues of mis-billing for calls that only experience one-way audio); which is what you are experiencing. The only recommendation that we can suggest to you is the following: 1.) Within the Dial Dictate interface please click on ‘Dial Dictate’ > ‘Options’ > and click on the ‘Options’ tab 2.) From there, locate the ‘Record voice activation level (dB)’ and set your voice activation level to -30dB. However if the above mentioned doesn’t work, that this seems to be more of an issue with the software itself and how the software handles its calls; more specifically, it seems that the Dial Dictate software doesn’t send out RTP Keep Alive packets to our servers to prevent our servers from dropping the calls due to one way audio. Due to the limitation of the software itself, there is not much to suggestion except that we recommend that you contact NCH support with regards to the issue at hand as it seems to be an issue with the software.