IWS DTMF Problems where DN Parameter "record = true"

[David Alvarez]

Hello all,
I have a problema with DTMF tones from genesys, when in the Extensión DN / Agent Login parameter: record = true

This parameter are necesary to record calls (I have Genesys Quality Management by Zoom), but the DTMF tones not worked. If i delete this parameter, the DTMF tones worked normally.

The DN annex tab:

[TServer]
cpn = 19100
default-music = http://localhost/music/101900730.wav
display-name = Kevin Gudiel
dual-dialog-enabled = true
make-call-rfc3725-flow = 1
multimedia = true
preview-interaction = false
record = true
refer-enabled = false
reinvite-requires-hold = false
ring-tone-on-make-call = true
sip-cti-control = talk,hold,dtmf
sip-enable-moh = true
use-display-name = true
voice = true

Anyone know any alternative to make work this two functions without compromising one or the other?

[Kubig]

Could you specify what is not working on DTMF level? What type of DTMF method are you using - in-bandm, out-of-band, etc.? What steps did you make within your analysis - network sniff, log check, etc.? And what were the outputs from the analysis?

[boigas]

DId you fix the issue? i am getting the same issue. Must be something with the MCP as it get conf but i can not find the solution.

[jank]

Hi David,
Were you able to resolve this DTMF issue, what was your resolution. I'm having the same issue. For some numbers I get DTMF and for others no DTMF.

Thanks
JAn

[Kubig]

Did you check the SIP communication path? What DTMF do you use?

BTW: For GQM the record option should be configured with value "true" as the recording is driven by GQM core

[jank]

Morning Kubig.

Client is using Audiocodes with Genesys GVP. Recording used is Datavoice. I had the issue where DTMF did not work when dailling some numbers. Hardphones worked fine but sip endpoint extensions had issues where the DTMF options entered was not recognized. Added the sip-cti-control = dtmf,talk,hold to the sipendpioint Annex/Tserver section for the extensions. Problem was resolved. Problem was raised again that for other numbers DTMF not working. Changed the config for the Audio code phones option: DTMF Transport Mode - Via SIP. This resolved DTMF for the Audio code phones. Sip End point extensions still to be fixed.

Please advise what you need me to check for SIP communication? Is it a specific option?

DTMF options below on Audio codes:
RFC 2833 Mode = AS IS
RFC 2833 DTMF Payload  = 0
Alternative DTMF Method = AS IS