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 ... View more