3 way call dtmf recognition/disabling

[ivschakravarthi]

Hi,

An incoming call came and got routed to an agent.
Agent spoke for some time and using the "Transfer-Conf" button,he transfered the call to some other IVR menu using "Dial with Customer" button.

Now,IVR-Agent-Customer were in one conference.

In the IVR,let us say "Press 1 to do some thing" .Now,the customer & Agent both will be able to key in the DTMF.[Generally,its a customer decision to go or not and agent should not press the dtmf.]
Even though agent presses ,system should be able to log the data [DTMF passed by agent | DTMF passed by customer].

or else System should not recognise the dtmf from agent or agent should not have option to pass DTMF ....etc.

We are getting complaints from the customers like "I didnt press the key.Agent has pressed it without my knowledge....etc."

I checked Blind transfer option,but that is completely forwarding the call to IVR and hanging up the agent session and moving to next call.But that is not i want.
I want all to be in a 3 way call,customer will do some stuff in the IVR while agent is in silent and once ivr is done ,agent will continue his communication with the customer.

Is that option is available?How to handle to that situation.Please help



Vicdial VERSION: 2.8-403a |BUILD: 130510-1350 |asterisk 1.4.27.1|single server |Centos6.5 |One Digium card | No cluster

[geoff3dmg]

DTMF in meetme conferences are is reliable. The only reliable way is for the agent to use the 'send dtmf' input on the agent interface.

[williamconley]

DTMF in meetme conferences are is reliable. The only reliable way is for the agent to use the 'send dtmf' input on the agent interface.

not what was asked. He would like to DISABLE agent DTMF while in 3-way so ONLY the prospect could press a button.

The challenge is that DTMF tones are audio. If the agent has a soft phone or VOIP phone, they have access to those buttons and can press, and the system will generate the tone. No way to extract that or block it without muting the agent.

So the answer is no: There is no "block the DTMF in the agent screen" option during 3-way calls (or any other time, for that matter). Nor is there a way (of which I am aware) to block DTMF tones in asterisk itself.

That being said:

http://www.voip-info.org/wiki/view/Aste ... blocksDTMF could be helpful. You could also build an app to MUTE the agent during 3-way call and require the agent to terminate the 3-way (and start over) to be able to comment or interact. This could cause the process to take longer, but greatly reduce chargebacks (especially after the agents realize they can no longer cheat and have to sell properly before initiating the 3-way by prepping the prospect for the confirmation script correctly).

You could also experiment with completely disabling the agent's phones ability to DTMF (they don't need it! it's in the agent interface!) and then just write a script to disable the one in the agent interface during 3-way calls (should be a campaign preference, actually)

[ivschakravarthi]

i have enabled asterisk debug mode and checked and i got the below responses.

When pressed from the phone,got the response as..

[Mar 27 23:09:10] DTMF[31237]: channel.c:2116 __ast_read: DTMF end '1' received on Zap/3-1, duration 0 ms
[Mar 27 23:09:10] DTMF[31237]: channel.c:2152 __ast_read: DTMF begin emulation of '1' with duration 100 queued on Zap/3-1
[Mar 27 23:09:10] DTMF[31342]: channel.c:2191 __ast_read: DTMF begin '1' received on Local/1234@default-0f79,2
[Mar 27 23:09:10] DTMF[31342]: channel.c:2195 __ast_read: DTMF begin ignored '1' on Local/1234@default-0f79,2
[Mar 27 23:09:10] DTMF[31284]: channel.c:2191 __ast_read: DTMF begin '1' received on Local/58600052@default-a556,1
[Mar 27 23:09:10] DTMF[31284]: channel.c:2201 __ast_read: DTMF begin passthrough '1' on Local/58600052@default-a556,1
[Mar 27 23:09:10] DTMF[31237]: channel.c:2261 __ast_read: DTMF end emulation of '1' queued on Zap/3-1
[Mar 27 23:09:10] DTMF[31342]: channel.c:2116 __ast_read: DTMF end '1' received on Local/1234@default-0f79,2, duration 119 ms
[Mar 27 23:09:10] DTMF[31342]: channel.c:2179 __ast_read: DTMF end passthrough '1' on Local/1234@default-0f79,2
[Mar 27 23:09:10] DTMF[31284]: channel.c:2116 __ast_read: DTMF end '1' received on Local/58600052@default-a556,1, duration 119 ms
[Mar 27 23:09:10] DTMF[31284]: channel.c:2163 __ast_read: DTMF end accepted with begin '1' on Local/58600052@default-a556,1
[Mar 27 23:09:10] DTMF[31284]: channel.c:2179 __ast_read: DTMF end passthrough '1' on Local/58600052@default-a556,1
-- User entered '1'


When pressed from Agent SIP phone,got the response as

[Mar 27 23:11:03] DTMF[31721]: channel.c:2116 __ast_read: DTMF end '1' received on SIP/cc101-08fb8488, duration 250 ms
[Mar 27 23:11:03] DTMF[31721]: channel.c:2152 __ast_read: DTMF begin emulation of '1' with duration 250 queued on SIP/cc101-08fb8488
[Mar 27 23:11:03] DEBUG[31721]: chan_zap.c:1049 zt_digit_begin: Started VLDTMF digit '1'
[Mar 27 23:11:03] DTMF[31785]: channel.c:2191 __ast_read: DTMF begin '1' received on Local/58600052@default-20b5,1
[Mar 27 23:11:03] DTMF[31785]: channel.c:2201 __ast_read: DTMF begin passthrough '1' on Local/58600052@default-20b5,1
[Mar 27 23:11:03] DTMF[31823]: channel.c:2191 __ast_read: DTMF begin '1' received on Local/1234@default-fc8e,2
[Mar 27 23:11:03] DTMF[31823]: channel.c:2195 __ast_read: DTMF begin ignored '1' on Local/1234@default-fc8e,2
[Mar 27 23:11:03] DTMF[31721]: channel.c:2261 __ast_read: DTMF end emulation of '1' queued on SIP/cc101-08fb8488
[Mar 27 23:11:03] DEBUG[31721]: chan_zap.c:1084 zt_digit_end: Ending VLDTMF digit '1'
[Mar 27 23:11:03] DTMF[31823]: channel.c:2116 __ast_read: DTMF end '1' received on Local/1234@default-fc8e,2, duration 263 ms
[Mar 27 23:11:03] DTMF[31785]: channel.c:2116 __ast_read: DTMF end '1' received on Local/58600052@default-20b5,1, duration 263 ms
[Mar 27 23:11:03] DTMF[31785]: channel.c:2163 __ast_read: DTMF end accepted with begin '1' on Local/58600052@default-20b5,1
[Mar 27 23:11:03] DTMF[31785]: channel.c:2179 __ast_read: DTMF end passthrough '1' on Local/58600052@default-20b5,1
[Mar 27 23:11:03] DTMF[31823]: channel.c:2179 __ast_read: DTMF end passthrough '1' on Local/1234@default-fc8e,2
-- User entered '1'

In this setup,both PRI/SIP are in same asterisk server.
I need to separate both and test it .
May be after some more testing,i can conclude.

[williamconley]

Keep hunting. If you get it ... it will be possible for a new feature, even if only in limited circumstances.