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CUE Autoattendant can't transfer calls after PRI to SIP circuit chang

CUE  - Auto attendant can't transfer calls to phones  after PRI to SIP circuit changes.  Call is hitting on the auto attendant, but when press the options, calls are not transferring.

Configuration for autoattendant

 

Building configuration...

 

boot-start-marker
boot system flash flash0:c2900-universalk9-mz.SPA.157-3.M.bin
boot system flash flash0:
boot-end-marker
!

 


voice service voip
ip address trusted list

no ip address trusted authenticate
mode border-element license capacity 25
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711alaw

sip
bind control source-interface GigabitEthernet0/0 >>>>>>>>> LAN Interface
bind media source-interface GigabitEthernet0/0
header-passing
registrar server
early-offer forced
midcall-signaling passthru
privacy-policy passthru
g729 annexb-all
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8

!
voice hunt-group 1 parallel
final 2001
list 7500,7528
timeout 15
pilot 2000
!
!
voice hunt-group 2 parallel
list 7500,7427,7428,7429,7430,7407,7412,7514,7511,7510,7516,7519,7508,7509
timeout 3600
pilot 2001
!
!
voice hunt-group 3 parallel
final 2001
list 7514,7516,7511,7510
timeout 10
pilot 2004
!
!
voice hunt-group 4 parallel
final 2001
list 7427,7429,7407,7412
timeout 10
pilot 2005
!
!
voice hunt-group 5 parallel
final 2001
list 7428,7528
timeout 10
pilot 2006
!
!
voice hunt-group 6 parallel
final 2001
list 7508,7523
timeout 10
pilot 2007
!
!
voice hunt-group 7 parallel
final 2001
list 7509,7523
timeout 10
pilot 2008
!
!
!
!
voice translation-rule 10
rule 1 /^9\(.*$\)/ /\1/
!
voice translation-rule 20
rule 1 /^(.*)$/ /03\1/
!
voice translation-rule 30
rule 1 /^1(..).*$/ /\1/
!
voice translation-rule 111
rule 1 /^03/ //
!
!
voice translation-profile INBOUND_TRANSLATION
translate called 111
!
voice translation-profile OUTGOING_3DIGITS
translate calling 20
translate called 30
!
voice translation-profile OUTGOING_TRANSLATION
translate calling 20
translate called 10
!
!


!
!
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
!
mgcp profile default
!
!
!
!
dial-peer voice 200 voip
description Calls to AutoAttendant
destination-pattern 123456
b2bua
session protocol sipv2
session target ipv4:CME IP
dtmf-relay rtp-nte
codec g711ulaw
no vad
!


dial-peer voice 500 voip
description Inbound Dial-Peer match from carrier to CUBE
translation-profile incoming INBOUND_TRANSLATION
session protocol sipv2
session transport udp
incoming called-number .
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
fax protocol pass-through g711ulaw
no vad
!

telephony-service
authentication credential xxxxxxx
xml user admin password xxxxxx 15
max-ephones 100
max-dn 200
ip source-address cme ip port 2000
service phone webAccess 0
service phone ehookEnable 1
service directed-pickup gpickup
timeouts interdigit 12
system message Bell Helicopter

cnf-file perphone
user-locale
network-locale
time-zone
time-format
date-format yy-mm-dd
voicemail
mwi relay
max-conferences 8 gain -6
call-park system application
call-forward pattern .T
moh enable-g711 "flash0:/music-on-hold.au"
multicast moh port 2000 route 10
w
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern .T
secondary-dialtone 9
fac standard
create cnf-files version-stamp Jan 01 2002 00:00:00
!

Please advise!

9 Replies 9

If I understood correctly, you are unable to select the menus when the call reaches the IVR, specifically for calls coming to the CME through the PRI.

I couldn't find any configurations for the PRI in the shared configuration. What DTMF settings have been configured for calls coming through the PRI?

It appears to be a DTMF configuration issue.



Response Signature


Hi Nithin,

They were using T1 PRI and we moved them to SIP circuits.

here is the dialpeer for autoattendant.


dial-peer voice 200 voip
description Calls to AutoAttendant
destination-pattern 123456
b2bua
session protocol sipv2
session target ipv4:10.12.12.12
dtmf-relay rtp-nte
codec g711ulaw
no vad
 

 

configuration is , when user press menu option, it will send to a hunt pilot , not directly to the 6 digit number.

 

For us to help you you’ll need to share some output from debugs. Can you please turn on these debugs and collect the output in a text file that you attach to your reply?

  • debug ccsip message 
  • debug voip ccapi inout 


Response Signature


HI All,

Thanks a lot Nitin and Roger for your quick response.

I have captured 2 logs, 1 with dtmf-rtp-nte and other one after change the dtmf relay yo sip-notify.

both case I can able to get the greetings and nothing happens when I press the options.
From auto attendant, we are sending the calls to the hunt as there are some more phones to ring together.


Hi Nitin,
These are the dial-peers for incoming call for auto-attendant

dial-peer voice 200 voip
description Calls to AutoAttendant
destination-pattern 45107525
b2bua
session protocol sipv2
session target ipv4:10.xx.xx.xx CUE ip
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
codec g711ulaw
no vad


dial-peer voice 500 voip
description Inbound Dial-Peer match from Colt to CUBE
translation-profile incoming INBOUND_TRANSLATION
session protocol sipv2
session transport udp
incoming called-number .
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
fax protocol pass-through g711ulaw
no vad

DhaneeshKarthikeyan_0-1734382873018.png

Script Screenshot, for testing option 0 is changed to phone number, otherwise all options configured as hunt pilot and calls routing to phones from there.

So, are you saying that transferring to a DID number worked, but using the hunt pilot number did not? Our assumption was that the menu options are not working when you select the menu, which might be a DTMF issue. If it's about transferring, that would be a different issue.



Response Signature


Hi Nithin,

Thanks a lot for your prompt reply's.
I can bale to call and hear the greeting, when I  press the menu options (0-sales, 1-maintenance. 0 to operator) nothing works.
I changed the 0 (for operator was again a hunt pilot, i have changed it to a actual phone number for testing,) that also not working.
Is it something related to the DTMF of time expiry .?
I have uploaded the test call logs as well. Could you please verify .?

Carrier update :
We have made test call to 0345107525 from our landline and heard the greetings but the call was not transferred.
In addition, we heard the same voice message after pressed "0" and call was disconnected automatically.
As far as we checked our CDR, the call was disconnected by the called party.(Disc reason 16: normal call clearing)
So, please kindly check your PBX setting.
we only support In-Band DTMF.