We created a Directory Number and forwarded all the calls to an 855 number.
I configured a route pattern pointing to the PRI port; we are able to call the Directory Number from internal phones and from the outside, calls get forwarded to the 855 number, but the DTMF doesn’t seem to be working because the telephone’s buttons are not recognized as options when pressed. This ISR with the PRI service is configured with MGCP service in our Call Manager.
I would like help to get this ISR with the PRI service configured properly with DTMF.
We’re currently on CUCM 11.5 with one publisher and two subscribers
We have three ISRs, one of them with both SIP trunks and PRIs services.
Voice service voip:
voice service voip
ip address trusted list
ipv4 10.XXX.XXX.0 255.255.255.0
ipv4 10. XXX.XXX.0 255.255.255.0
ipv4 12. XXX.XXX.0 255.255.255.0
ipv4 12. XXX.XXX.0 255.255.255.0
ipv4 12. XXX.XXX.0 255.255.255.0
ipv4 12. XXX.XXX.0 255.255.255.0
ipv4 12. XXX.XXX.0 255.255.255.0
address-hiding
dtmf-interworking rtp-nte
mode border-element
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
no fax-relay sg3-to-g3
modem passthrough nse codec g711ulaw
sip
header-passing
error-passthru
registrar server
asserted-id pai
early-offer forced
midcall-signaling passthru
privacy-policy passthru
g729 annexb-all
!
voice class codec 100
codec preference 1 g711ulaw
codec preference 2 g729r8
I only see two dial-peers on this ISR; see below:
dial-peer voice 100 voip
description CALL MANAGER
destination-pattern [2-9]..[2-9]......$
session protocol sipv2
session server-group 100
incoming called-number 9T
voice-class codec 100
voice-class sip asymmetric payload full
voice-class sip privacy-policy passthru
voice-class sip profiles 100
voice-class sip bind control source-interface GigabitEthernet0/0/0
voice-class sip bind media source-interface GigabitEthernet0/0/0
dtmf-relay rtp-nte
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 101 voip
description ATT SIP
translation-profile outgoing PSTN_OUT
call-block translation-profile incoming PSTN_BLOCK
call-block disconnect-cause incoming unassigned-number
destination-pattern 9T
session protocol sipv2
session server-group 101
incoming called-number 956[2-9]......$
voice-class codec 100
voice-class sip asymmetric payload full
voice-class sip privacy-policy passthru
voice-class sip profiles 100
voice-class sip bind control source-interface GigabitEthernet0/0/1
voice-class sip bind media source-interface GigabitEthernet0/0/1
dtmf-relay rtp-nte
ip qos dscp cs3 signaling
no vad
The MGCP configuration is:
mgcp
mgcp call-agent ccm3.organization.com 2427 service-type mgcp version 0.1
mgcp modem passthrough voip mode nse
mgcp package-capability mf-package
mgcp package-capability fm-package
mgcp default-package fxr-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp fax t38 nsf 000000
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
I am new to this organization, and I am still learning their setup.
Any help and guidance is greatly appreciated.
Please let me know what other information is required to help me solve this issue.
Thanks again.
Art Martinez
