Hello,
I am working with my partner and ATT setting up a 2921 for IP FLex. The system is working fine currently with a PRI but we are looking to migrate the site to IP FLex.
CUCM seems to be setup ok, the CUBE elements seem to be setup ok - however I cannot bind the interface on the Dial Peers.
At this time when calls come in, the end point rings and everything looks good on the debug logs, however when you try to answer nothing happens, the 2nd time you hit answer the call disconnects. Debug Below:
477859: May 10 16:09:51.867: //629155/76192ACF9D48/CCAPI/cc_api_call_alert:
Interface=0x29662448, Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
477860: May 10 16:09:51.867: //629155/76192ACF9D48/CCAPI/cc_api_call_alert:
Call Entry(Retry Count=0, Responsed=TRUE)
477861: May 10 16:09:51.867: //629154/76192ACF9D48/CCAPI/ccCallAlert:
Progress Indication=NULL(0), Signal Indication=SIGNAL RINGBACK(1)
477862: May 10 16:09:51.867: //629154/76192ACF9D48/CCAPI/ccCallAlert:
Call Entry(Responsed=TRUE, Alert Sent=TRUE)
477863: May 10 16:09:53.319: //629155/76192ACF9D48/CCAPI/cc_api_call_disconnected:
Cause Value=63, Interface=0x29662448, Call Id=629155
477864: May 10 16:09:53.319: //629155/76192ACF9D48/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=63, Retry Count=0)
477865: May 10 16:09:53.319: //629154/xxxxxxxxxxxx/CCAPI/ccCallReleaseResources:
release reserved xcoding resource.
477866: May 10 16:09:53.319: //629155/76192ACF9D48/CCAPI/ccCallSetAAA_Accounting:
Accounting=0, Call Id=629155
477867: May 10 16:09:53.319: //629155/76192ACF9D48/CCAPI/ccCallDisconnect:
Cause Value=63, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=63)
477868: May 10 16:09:53.319: //629155/76192ACF9D48/CCAPI/ccCallDisconnect:
Cause Value=63, Call Entry(Responsed=TRUE, Cause Value=63)
477869: May 10 16:09:53.323: //629155/76192ACF9D48/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x29662448, Tag=0x0, Call Id=629155,
Call Entry(Disconnect Cause=63, Voice Class Cause Code=0, Retry Count=0)
When the call comes in it hits first DP, then the incoming DP.
Incoming Dial-peer=1500, Progress Indication=NULL(0), Calling IE Present=TRUE,
477842: May 10 16:09:51.859: //629154/76192ACF9D48/CCAPI/ccCallSetupRequest:
Destination=, Calling IE Present=TRUE, Mode=0,
Outgoing Dial-peer=601, Params=0x301ADE74, Progress Indication=NULL(0)
477843: May 10 16:09:51.859: //629154/76192ACF9D48/CCAPI/ccCheckClipClir:
In: Calling Number=XXXXXXXXXX (TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
I am binded on my interface, however that option doesn't exist on the Dial Peers.. Some configs below:
voice service voip
address-hiding
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol none
h323
sip
bind control source-interface Loopback2
bind media source-interface GigabitEthernet0/1.2 ( i have tried binding both to Loop2 and 0/1.2)
header-passing error-passthru
no update-callerid
midcall-signaling passthru
privacy-policy passthru
g729 annexb-all
sip-profiles 15
voice class uri 1000 sip
host 1xx.xx.xx.181
!
voice class uri 2000 sip
host 10.xx.xx.10
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
voice class sip-profiles 1
request INVITE sip-header Diversion modify "<sip:(.*)@(.*)>" "<sip:xxxxx\1@\2>"
request REINVITE sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" ""
!
voice class sip-profiles 15
request INVITE sip-header Allow-Header modify ", UPDATE" ""
request REINVITE sip-header Allow-Header modify ", UPDATE" ""
response 180 sip-header Allow-Header modify ", UPDATE" ""
response 200 sip-header Allow-Header modify ", UPDATE" ""
voice translation-rule 1
rule 1 // /A/
voice translation-rule 3
rule 1 /^A/ //
voice translation-profile outbound-strip-A
translate called 3
voice translation-profile inbound-from-carrier
translate called 1
interface Loopback2
ip address xx.xx.xx.194 255.255.255.255
!
!
interface GigabitEthernet0/0
description ATT_Circuit_ID_MLEC.XXXXX..ATI
no ip address
duplex auto
speed 100
!
!
interface GigabitEthernet0/0.50
bandwidth 5000
encapsulation dot1Q 50
ip address xx.xx.xx.xx 255.255.255.252
service-policy output TRAFFIC_SHAPING
!
interface GigabitEthernet0/1
no ip address
duplex auto
speed 1000
!
!
interface GigabitEthernet0/1.1
encapsulation dot1Q 1 native
ip address 10.xx.xx.4 255.255.224.0
vrrp 1 ip 10.xx.xx.1
vrrp 1 preempt delay minimum 20
vrrp 1 priority 110
vrrp 1 track 1 decrement 90
mgcp bind control source-interface GigabitEthernet0/1.2
mgcp bind media source-interface GigabitEthernet0/1.2
sccp local GigabitEthernet0/1.2
sccp ccm 10.xx.xx.1 identifier 1 version 7.0
sccp ccm 10.xx.xx.10 identifier 100 version 7.0
sccp
!
sccp ccm group 1
associate ccm 100 priority 1
associate profile 2 register mtp0123456789ab
associate profile 1 register confprofile1
dspfarm profile 2 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
!
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
conference-join custom-cptone jointone
conference-leave custom-cptone leavetone
associate application SCCP
!
dial-peer voice 601 voip
description Inbound to CUCM
translation-profile outgoing outbound-strip-A
preference 1
destination-pattern A.T
session protocol sipv2
session target ipv4:10.xx.xx.10
session transport udp
voice-class sip url sip
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 1500 voip
description Inbound Call from ATT to CUBE
translation-profile incoming inbound-from-carrier
session protocol sipv2
incoming called-number .T
voice-class codec 1
voice-class sip asymmetric payload full
voice-class sip asserted-id pai
voice-class sip privacy-policy passthru
voice-class sip profiles 1
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400
no vad
!
!
dial-peer hunt 1
sip-ua
no remote-party-id
retry invite 2
telephony-service
bulk-speed-dial list 0 flash:test.txt
max-conferences 8 gain -6
transfer-system full-consult
I tried but was unable to add this to the DPs
voice-class sip bind control source interface gig0/1.2
voice-class sip bind media source interface gig0/1.2
Any assistance is appreciated, it's my first time seeing IPFlex.
Thanks to all.
-Dennis
