Version Issues - Repost Of Sit Issue

[Guest u4910]

In trying to figure out my issue with SIT being detected (when dialing from handset works fine), I have found that even though the phone lines are clean sounding now, I have these issues with the following versions. It does seem that the 7.0.9 version on the web site works the best, meaning it will dial, but that version stays connected and will never play wav. The later versions just report SIT after dialing.

Please advise on what is different in these versions:

 

 

 

7.0.8 copy xml wont even try to dial

7.0.9 dials, and stays connected, but never plays wav

7.0.9.090103c SIT

7.0.9.090104 SIT

7.0.9.090106 SIT

[Guest u4910]

Log is attached.

0207_ktTel.txt

[Guest u4910]

Whatever you have in 7.0.9 that makes it dial OK (except that it never plays wav file), but now gets SIT in the later versions, can you fix back in a next version and post that download? I really want to get this working.

[SupportTeam]

Attached trace shows that SIT tone was detected on all outgoing calls, but in older versions it was just reported differentely.

 

There are no differences in how the tone detecton of outgoing calls is setup in the various versions. All versions just leave the Dialogics defaults as they are.

 

You can modify the tri-tone (SIT) frequences that Dialogic looks out for by setting the DX_CAP parameters yourself. VoiceGuide gives you access to set these. But you really should concentrate on getting rid of the hum on your phone lines.

 

If you are interested in changing Dialogic's tri-done detection params then you should read the Voice Programming PDF file (in Dialogc's \doc\ subdirectory) section: 7.16.1 Tri-Tone SIT Sequences. The below extract shows how to change DX_CAP structure values (on a per-call basis):

 

With current v7 you are able to access all entries in the Dialogic DX_CAP structure, which includes Dialogic's low level settings.

eg settings which affect the Answering Machine / Live Answer detection timeouts:

 

ca_cnosig

ca_noanswer

ca_pamd_failtime

 

There are about 30-40 settings in the DX_CAP structure, all of which are accessible to you, so you will have total control (on a per-call basis) over how the Dialogic card can do the detection.

 

For more information on the fields in the DX_CAP structure you will need to read Dialogic's documentation - see the "Voice API" file.

 

To change setting to your value (instead of using the default) you will be able to use an expression like this in the "Call Options" field (same field where CallerID to be used on the outgoing call is specified):

 

<DX_CAP><ca_cnosig>500</ca_cnosig><ca_pamd_failtime>200</ca_pamd_failtime></DX_CAP>

[Guest u4910]

Thank you very much for the information. I will certainly take a look!

Also I was trying the PBXpert to try and learn the current phone lines. I noticed there is a way to set the ca_lower allowable frequencies for frequency 1. And since the SIT tones seem to be as below; I was thinking to set the start ca_lower to be 913 (since that is abot 906), but when I run PBXpert for my two lines it works until it gets to the "connect" and then it failes on "failed waiting for rings".

So, so much for trying to learn my lines.... I tried changing numbers and everything. I think I'll try your suggestion above.

 

 

DX_OPTDIS disables frequency detection and therefor disables SIT detection.

There are four SIT sequences that can be detected:

1st Tone 2nd Tone 3rd Tone

SITName SITDescription Freq. Len. Freq. Len. Freq. Len.

NC No Circuit Found 985 38 1429 38 1777 38

IC Operator Intercept 914 27 1371 27 1777 38

VC Vacant Circuit 985 38 1370 27 1777 38

RO Reorder (system busy) 9148 27 1429 38 1777 38

SIT sequence definitions differ on Springware boards versus DM3 boards. For more information on SIT sequences and call progress analysis, see the Voice API Programming Guide.

 

 

[Guest u4910]

Tried this to keep the min above the 906hz issue I'm having, but it is still detecting 906hz.

Please advise if this doesn't look right. I added the last line.

 

<OutDialEntry>

<PhoneNumber>5551111</PhoneNumber>

<PhoneNumberPrefix>*67</PhoneNumberPrefix>

<ActivateTime></ActivateTime>

<DayTimeStart></DayTimeStart>

<DayTimeStop></DayTimeStop>

<DaysCallAllowed>MoTuWeThFrSaSu</DaysCallAllowed>

<LineSelection></LineSelection>

<CampaignName></CampaignName>

<Priority></Priority>

<OnAnswerLive>e:\ivr\outbound1\1111.wav</OnAnswerLive>

<OnAnswerMachine></OnAnswerMachine>

<OnNotAnswered></OnNotAnswered>

<OnRetriesExhausted></OnRetriesExhausted>

<AnswerTimeout>40</AnswerTimeout>

<RetriesLeft></RetriesLeft>

<RetriesDelay></RetriesDelay>

<RV></RV>

<CallOptions></CallOptions>

<Escalation></Escalation>

<DX_CAP><ca_lowerfrq>999</ca_lowerfrq></DX_CAP>

</OutDialEntry>

 

[SupportTeam]

The DX_CAP changes need to be specified in the CallOptions section. See below example:

 

<OutDialEntry>

<PhoneNumber>5551111</PhoneNumber>

<PhoneNumberPrefix>*67</PhoneNumberPrefix>

<ActivateTime></ActivateTime>

<DayTimeStart></DayTimeStart>

<DayTimeStop></DayTimeStop>

<DaysCallAllowed>MoTuWeThFrSaSu</DaysCallAllowed>

<LineSelection></LineSelection>

<CampaignName></CampaignName>

<Priority></Priority>

<OnAnswerLive>e:\ivr\outbound1\1111.wav</OnAnswerLive>

<OnAnswerMachine></OnAnswerMachine>

<OnNotAnswered></OnNotAnswered>

<OnRetriesExhausted></OnRetriesExhausted>

<AnswerTimeout>40</AnswerTimeout>

<RetriesLeft></RetriesLeft>

<RetriesDelay></RetriesDelay>

<RV></RV>

<CallOptions><DX_CAP><ca_lowerfrq>999</ca_lowerfrq></DX_CAP></CallOptions>

<Escalation></Escalation>

</OutDialEntry>

[Guest u4910]

That seems to be helping for sure now. I am able to "place" every call made now. Out of about 40 calls made, only about 4 got a SIT after I answered the phone. So, it was like I answered the phone and the application hung up on me at that point. I can still see other SIT's in the log, so I assume they still get reported even though VG is trying to ignore them? The calls sound really clear on the receiving end (as they always have). I do notice that it does falsely detect answering machine a good bit of the time.

0210_ktTel.txt

0210_0752_vgEngine.txt

[SupportTeam]

999 is not the highest frequency value that you can use in Tone definitions. The values can go up to about 2000.