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  #1  
Old 11-21-2017, 07:12 PM
hawles hawles is offline
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Exclamation CM DTMF not working on direct SIP to CMM 7.x

Hi all,

I have a CMM on the latest 7.0 and the CM is latest 7.1 as of 11/21/17. They have direct SIP integration. Internal and external calls to the CMM get through and you can hear it say welcome to Audix. Please enter your mailbox and press #. Digit tones (DTMF signalling) are not heard by the CMM. I followed the Implementation doc here: https://downloads.avaya.com/css/P8/documents/101014318. Issue happens whether you are on IP or analog phone. All other integration seems to be working. WMI lamps work. The CMM and CM are on s8300e AVP with 3 g450s for DSP/VOIP resources. The MGs are on latest firmware 38.20.1. What am I missing?
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  #2  
Old 11-24-2017, 04:21 AM
kumar1499 kumar1499 is offline
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YOu may need to check below parameters between SIP signalling group on CM and switch link Administration on CMM

DTMF over IP

rtp-payload
if
SIP INFO for DTMF
field is set to
Ignore
on the Switch Link Admin form.

out-of-band
if
SIP INFO for DTMF
field is set to
Accept
on the Switch Link Admin form.
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  #3  
Old 12-04-2017, 07:10 PM
wellington35 wellington35 is offline
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Hawles did you get this working? or still need help?
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Wellington Paez
Senior Convergence Specialist @ Carousel Industries
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  #4  
Old 05-17-2018, 06:24 AM
filip36 filip36 is offline
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Hallo! I have the same problem and i don't know how to resolve it. I have changed DTMF Parameters and there is no result.
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  #5  
Old 05-21-2018, 10:45 AM
hurdk hurdk is offline
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SIGNALING GROUP

Group Number: 10 Group Type: sip
IMS Enabled? n Transport Method: tcp
Q-SIP? n
IP Video? n Enforce SIPS URI for SRTP? y
Peer Detection Enabled? y Peer Server: Others
Prepend '+' to Outgoing Calling/Alerting/Diverting/Connected Public Numbers? n
Remove '+' from Incoming Called/Calling/Alerting/Diverting/Connected Numbers? y
Alert Incoming SIP Crisis Calls? n
Near-end Node Name: procr Far-end Node Name: msgserver
Near-end Listen Port: 5060 Far-end Listen Port: 6060
Far-end Network Region: 1

Far-end Domain: rescue.org
Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
Session Establishment Timer(min): 3 IP Audio Hairpinning? n
Enable Layer 3 Test? y Initial IP-IP Direct Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6
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  #6  
Old 05-21-2018, 10:59 PM
filip36 filip36 is offline
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Thank You! I will compare settings of the signalling group.
And i found manual "Configuration Note 88104 – Version K (12/22/16)
Avaya S8xx0 Aura Messaging SIP Integration directly to Avaya CM", want to try it.
Signalling group and trunk is up. I hear message "Welcome to Audix" but i can't leave voice message to mailbox. There is always message "Welcome to Audix".
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  #7  
Old 05-22-2018, 02:11 AM
kumar1499 kumar1499 is offline
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Does DTMF work from Phone to any other device ?
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  #8  
Old 05-22-2018, 05:17 AM
filip36 filip36 is offline
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Sure. With calls between phones and ip phones there is no problems, only with CMM.
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  #9  
Old Yesterday, 01:33 AM
filip36 filip36 is offline
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The problem was solved. The reason was in the table "public-unknown-numbering". I have added string for my local stations and trunk to CMM and it's ok.
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cmm, communication manager, dtmf signalling, integration, sip

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