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Disconnect Tone

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Hello,

I am currently in Kuala Lumpur, setting up IVR system. I am using VG for this system. the problem I have is, the lines won't disconnect after customers hang-upped.

 

this is a major problem for our business. I have recorded the disconnect tone. per my analysis the tone should be as follow:

 

<Tone Name="Call Progress Tone DISCONNECT_Tone8">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone8</ID>

<Freq1>425</Freq1>

<Freq1Dev>20</Freq1Dev>

<Freq2>432</Freq2>

<Freq2Dev>20</Freq2Dev>

<On>30</On>

<OnDev>5</OnDev>

<Off>35</Off>

<OffDev>5</OffDev>

<Count>3</Count>

</Tone>

 

would you please check the attached file and let me know what would be the right freq1, 2 and on and off for this tone?

I really appreciate your quick response. I need to fix this problem ASAP. our business is down due to this problem.

thanks

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Tone is a single tone, so only one frequency needs to be specified.

Also, try relaxing the tone specifications to see at which stage the disconnect tone is detected.

 

eg:

 

<Tone Name="Call Progress Tone DISCONNECT_Tone8">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone8</ID>

<Freq1>425</Freq1>

<Freq1Dev>40</Freq1Dev>

<Freq2>0</Freq2>

<Freq2Dev>0</Freq2Dev>

<On>30</On>

<OnDev>5</OnDev>

<Off>35</Off>

<OffDev>10</OffDev>

<Count>2</Count>

</Tone>

 

How many disconnect tones do you have defined in the ConfigLine.xml, and which model Dialogic card are you using?

 

Some cards cannot load many tone definitions, so you should only have the essential tones specified.

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Tone is a single tone, so only one frequency needs to be specified.

Also, try relaxing the tone specifications to see at which stage the disconnect tone is detected.

 

eg:

 

<Tone Name="Call Progress Tone DISCONNECT_Tone8">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone8</ID>

<Freq1>425</Freq1>

<Freq1Dev>40</Freq1Dev>

<Freq2>0</Freq2>

<Freq2Dev>0</Freq2Dev>

<On>30</On>

<OnDev>5</OnDev>

<Off>35</Off>

<OffDev>10</OffDev>

<Count>2</Count>

</Tone>

 

How many disconnect tones do you have defined in the ConfigLine.xml, and which model Dialogic card are you using?

 

Some cards cannot load many tone definitions, so you should only have the essential tones specified.

 

In the area of "extra tones", there is just 3 tones: "Fax", "DISCONNECT_TAPI2 tones" and "Malaysia Disconnect tone", (the one you just gave me).

 

thanks a lot. your tone translation is working very well. would you please teach me how did you figure out it is just one frequency, and Freq1Dev is 40. I was using Audacity and Cool Edit to analysis the tone. I have tried all day 100s of combinations of frequancy for freq 1 and 2. but none was give me any result.

 

thanks again for your help. I believe you should create a database file for all countries disconnect tone, this would help VG users a lot. for example my business was done for 2 days because of this problem.

 

thanks again for your help.

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Thanks for letting us know it's working now.

 

As a rule of thumb the 'deviation' should be about 10%.

 

And as the tone only has one distinct frequency then only one frequency should be defined in the tone definition.

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Dear Admin,

I used to have Caller ID problem because of the GSM gateway I had. I change the GSM Gateway and solve the caller ID but now I have again disconnect tone problem.

 

The VG won’t hang up when caller hang up. Again I tried to enter the freq into the Configline.xml. But it seems I can not finding the freq….

 

Please find attached file the new disconnect tone I have recorded.

I really appreciate if you help me to find the frequency of these new tone.

Our business is done again because of this problem.

Thanks for your help again.

 

Farzad

Disconnect Tone.zip

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Hello,

I really need your help on this. VG doesn't disconnect calls and customers are complaining day and night.

Please help me to find the disconnect tone I have sent you in my previous post.

 

thanks

Farzad

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I have entered the suggested freq into the configline but it didn't work.

here is what I have in configline. I am not sure the count and on adnoff dev is correct.

I have so many possibilities but none works.

thanks for your help again.

 

<Tone Name="Call Progress Tone DISCONNECT_Tone1">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone1</ID>

<Freq1>420</Freq1>

<Freq1Dev>40</Freq1Dev>

<Freq2>0</Freq2>

<Freq2Dev>0</Freq2Dev>

<On>40</On>

<OnDev>3</OnDev>

<Off>40</Off>

<OffDev>13</OffDev>

<Count>2</Count>

</Tone>

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Suggest you start from more relaxed settings, test to see if they work, and then gradually reduce the deviations and increase the cadence count.

 

eg. start with this and see if this works:

 

<Tone Name="Call Progress Tone DISCONNECT_Tone1">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone1</ID>

<Freq1>420</Freq1>

<Freq1Dev>60</Freq1Dev>

<Freq2>0</Freq2>

<Freq2Dev>0</Freq2Dev>

<On>40</On>

<OnDev>10</OnDev>

<Off>40</Off>

<OffDev>10</OffDev>

<Count>1</Count>

</Tone>

 

Please also post traces (ktTel and vgEngine) capturing the system startup. This will show us if the tone definition is correctly loaded.

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Thanks for your advice. I wished I knew this since 2003, (the time I start working on VG).

I discovered very interesting point on Malaysian disconnect tone.

 

Per your advice I could find the signal and enter it to Configline.xml. it worked fine but all the sudden after few hours it didn’t work any more. (call didn’t hang-upped)

 

After investigation I found out that Malaysian disconnect tone changes for day time and night time.

It is craze but it is true. The fist attached disconnect tone in this post is the night time disconnect tone and the current attached file is the day time disconnect tone.

 

The day time disconnect tone is mix of difference type of frequencies. I did my best to analysis all the frequencies, but unfortunate none of them recognized by Dialogic to hang-up the call.

 

Here are my analysis on these frequencies. I need your expertise to help me to get them right.

Thanks a lot for your help.

 

<Tone Name="Call Progress Tone DISCONNECT_Tone1">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone1</ID>

<Freq1>420</Freq1>

<Freq1Dev>60</Freq1Dev>

<Freq2>0</Freq2>

<Freq2Dev>0</Freq2Dev>

<On>40</On>

<OnDev>3</OnDev>

<Off>40</Off>

<OffDev>3</OffDev>

<Count>6</Count>

</Tone>

 

<Tone Name="Call Progress Tone DISCONNECT_Tone2">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone2</ID>

<Freq1>425</Freq1>

<Freq1Dev>40</Freq1Dev>

<Freq2>0</Freq2>

<Freq2Dev>0</Freq2Dev>

<On>30</On>

<OnDev>5</OnDev>

<Off>35</Off>

<OffDev>10</OffDev>

<Count>2</Count>

</Tone>

 

<Tone Name="Call Progress Tone DISCONNECT_Tone3">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone3</ID>

<Freq1>425</Freq1>

<Freq1Dev>40</Freq1Dev>

<Freq2>625</Freq2>

<Freq2Dev>60</Freq2Dev>

<On>25</On>

<OnDev>5</OnDev>

<Off>35</Off>

<OffDev>10</OffDev>

<Count>2</Count>

</Tone>

 

 

<Tone Name="Call Progress Tone DISCONNECT_Tone4">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone4</ID>

<Freq1>486</Freq1>

<Freq1Dev>40</Freq1Dev>

<Freq2>625</Freq2>

<Freq2Dev>60</Freq2Dev>

<On>25</On>

<OnDev>5</OnDev>

<Off>37</Off>

<OffDev>10</OffDev>

<Count>2</Count>

</Tone>

 

<Tone Name="Call Progress Tone DISCONNECT_Tone5">

<Notes>Default Setting</Notes>

<ID>DISCONNECT_Tone5</ID>

<Freq1>622</Freq1>

<Freq1Dev>40</Freq1Dev>

<Freq2>0</Freq2>

<Freq2Dev>0</Freq2Dev>

<On>13</On>

<OnDev>5</OnDev>

<Off>13</Off>

<OffDev>5</OffDev>

<Count>2</Count>

</Tone>

DisconnectTone4.zip

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The tone is 425Hz with cadence of 330ms ON / 330ms OFF cadence.

 

Would expect for DISCONNECT_Tone2 to pick that up.

 

Suggest you start from more relaxed DISCONNECT_Tone2 settings, test to see if they work, and then gradually reduce the deviations and increase the cadence count.

 

If you need reliable and immediate end of call notification you should use ISDN lines, not analog.

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Hello,

it seems that Malaysia telecommunication provide ISDN. I have study the Dialogic D/24OSC-T1 24-Channel T1 VOICE BOARD, ISDN. the problem now I have is how can I integrate this board with my ATA which have U.S.A sip account?

 

here is summary of my business and system setup. I have a calling card business in Kuala Lumpur. my system is setup the way that my 16 port dialogic board is divided to 8 port GSM gateway that has Malaysian sim card and the other 8 ports are connected to 8 ATA that has u.s.A sip account.

 

I am not sure how to use Dialogic D/24OSC-T1 24-Channel T1 VOICE BOARD, ISDN in my system? is there any terminal that I can use to switch the ISDN voice channels on ATAs?

 

would you please advice me.

Thanks

Farzad

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Sounds like you should be looking at using a HMP (VoIP) based system.

 

Have you looked at HMP?

 

You can have an E1 interface card in HMP. You would need to get a GSM gateway that has an E1 interface, and connect that to the E1 card. Or just get E! from Malaysia telecom and connect it directly to E1 card.

 

The HMP would make SIP calls directly without need for any other ATAs etc.

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Hello,

I need to make sure if I understand it correctly. you said:"You would need to get a GSM gateway that has an E1 interface, and connect that to the E1 card."

 

if I get the GSM Gateway with E1 interface, I don't need E1 line from Malaysian Telecom. the GSM Gateway would transfer digital data into the E1 card and VG can process it as I have E1 line from malaysian telecom. it make sense since GSM is digital signal.

 

please correct me if i am wrong, the GSM Gateway with E1 interface is the same as HMP? if it is not, where HMP will be interface in this system.

 

I believe I have to down laod VG rev7 and configure it on Digital lines during installation.

 

thanks for advice.

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if I get the GSM Gateway with E1 interface, I don't need E1 line from Malaysian Telecom. the GSM Gateway would transfer digital data into the E1 card and VG can process it as I have E1 line from malaysian telecom.

Correct.

 

the GSM Gateway with E1 interface is the same as HMP?

No.

HMP is the Dialogic VoIP stack. HMP will let you send out the other leg of call using SIP, without the need of any analog ATA gateway.

 

This way the end of call signaling etc. will be done directly using SIP messages, and not using tones from analog ATA gateway.

 

Of course if you can also get an E1<->SIP gateway. So you would have two E1 cards in system. One E1 for the GSM gateway and one E1 for the SIP/VoIP gateway. This can be done using normal Dialogic cards. You do not need to use "HMP+interface cards".

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