Can't pass digits to Auto Attendant (UC520)

omair2085 · ‎12-23-2009

I have configured my UC520 with a SIP trunk and all incoming/outgoing calls are working as desired.

The only problem I face is with the Auto Attendant. When I dial the number AA picks up but it doesn't accept any digit I press.

The debug output shows that I am pressing '0' but I am not really sure what rest means.

Dec 22 14:55:15.803: //193/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 0 to dstCallId 0xC2
Dec 22 14:55:15.803: //193/xxxxxxxxxxxx/CCAPI/cc_relay_digit_begin_for_3way_conf
erence:
   Check DTMF relay digit begin for 3way conf
Dec 22 14:55:15.803: //193/xxxxxxxxxxxx/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 0 to dstCallId 0xC2
Dec 22 14:55:15.803: //193/xxxxxxxxxxxx/CCAPI/cc_relay_digit_end_for_3way_confer
ence:
   Check DTMF relay digit end for 3way conf

My AA can also be reached through the number 100. When I dial this from another extension it works as expected.

Can someone shed some light, so that I can look in the right direction.

Marwan ALshawi · ‎12-23-2009

you said you can reach it through two diffrent numbers

is that means you have two dial-peers point to you AA

if yes make sure you have same seeting

especially the DTMF setting

good luck

if helpful Rate

omair2085 · ‎12-23-2009

I forgot to mention. The other number is a PSTN number and 100 is obviously a local extension.

dial-peer voice 1006 voip
description ** AA from SIP Trunk (Auto Attendant 1)**
translation-profile incoming AA_Profile
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number 22255555
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 100
b2bua
voice-class sip outbound-proxy ipv4:10.1.10.1
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 2003 voip
description ** cue auto attendant PSTN number **
translation-profile outgoing AA_Profile
destination-pattern 22255555$
b2bua
voice-class sip outbound-proxy ipv4:10.1.10.1
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!

Added the conf.

Marwan ALshawi · ‎12-23-2009

as long as its working wiuht your internal number VOIP dialpeer

and dosent thorugh the PSTN

then your issue is DTMF

please use this under your incoming PSTN dialpeer (POTS dial peer)

if you have only one pots dial-peer use the bellow

dial-peer voice [youportsdial peer number] pots

incoming called number .

dtmf-relay h245-alphanumeric

good luck

if helpful Rate

Message was edited by: marwanshawi sorry i was in a hurry yes not pots the incoming voip dial -peer

omair2085 · ‎12-23-2009

I don't think you can specify a dtmf-relay in a pots config.

paolo bevilacqua · ‎12-24-2009

When calling to or from a CME phone, can you hear dtmf from trunk ?

Which exact IOS are you using ?