Hi ,
SRST call-manager-fallback on SIP trunk for ISR 4K failing , when call manager made non availability , cisco phones not registering to gateway router.
am i missing something ? please . below is the Gateway router configuration.
!
isdn switch-type primary-ni
!
!
trunk group PSTN-CIRCUITS
hunt-scheme longest-idle
!
voice call send-alert
voice rtp send-recv
!
voice service pots
!
voice service voip
no ip address trusted authenticate
allow-connections sip to sip
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
bind control source-interface Loopback1
bind media source-interface Loopback1
rel1xx disable
min-se 900 session-expires 900
header-passing
error-passthru
midcall-signaling passthru
sip-profiles 1
!
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g729r8 bytes 30
!
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729r8 bytes 30
!
!
!
voice class sip-profiles 1
request REINVITE sdp-header Attribute modify "a=T38FaxFillBitRemoval:0" "a=Tool: GW"
request INVITE sdp-header Audio-Attribute add "a=ptime:30"
request REINVITE sdp-header Audio-Attribute add "a=ptime:30"
response ANY sdp-header Audio-Attribute modify "a=ptime:20" "a=ptime:30"
response ANY sdp-header Audio-Attribute add "a=ptime:30"
!
!
!
voice translation-rule 1
rule 1 /^36../ /64858336../
rule 2 /^44../ /54854244../
!
!
!
controller T1 0/2/0
framing esf
linecode b8zs
cablelength long 0db
pri-group timeslots 1-24
!
!
interface Serial0/2/0:23
description ** DIDs on PRI **
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn send-alerting
trunk-group PSTN-CIRCUITS
!
!
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 10.10.10.10 port 2000 strict-match
max-ephones 60
max-dn 100 dual-line
system message primary TELEPHONY BACKUP MODE
transfer-pattern .T
call-forward pattern .T
!
dial-peer voice 1101 voip
description *Incoming 7 digits calls via PSTN directed to Sub1 **
destination-pattern .T
translation outgoing 1
rtp payload-type nse 99
rtp payload-type nte 100
session protocol sipv2
session target ipv4:20.10.10.10
voice-class codec 2
no voice-class sip g729 annexb-all
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400 bytes 48
!
dial-peer voice 1102 voip
description *Incoming 7 digits calls via PSTN directed to Sub2 **
preference 2
destination-pattern .T
translation outgoing 1
rtp payload-type nse 99
rtp payload-type nte 100
session protocol sipv2
session target ipv4:20.10.10.11
voice-class codec 2
no voice-class sip g729 annexb-all
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400 bytes 48
!
!
!
dial-peer voice 1400 pots
trunkgroup PSTN-CIRCUITS
description ** Incoming DID calls via local PSTN service **
translation-profile incoming INCOMING
incoming called-number .
direct-inward-dial
!
!
dial-peer voice 1800 voip
preference 1
destination-pattern 0T
translate-outgoing calling 1
rtp payload-type nse 99
rtp payload-type nte 100
session protocol sipv2
voice-class codec 1
dtmf-relay rtp-nte
fax rate 14400
no vad
!
dial-peer voice 1500 voip
description **Incoming calls via CUCM**
rtp payload-type nse 99
rtp payload-type nte 100
session protocol sipv2
incoming called-number .
voice-class codec 2
no voice-class sip g729 annexb-all
dtmf-relay rtp-nte
fax-relay sg3-to-g3
fax rate 14400 bytes 48
!
!
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 10.10.10.10 port 2000 strict-match
max-ephones 60
max-dn 100 dual-line
system message primary TELEPHONY BACKUP MODE
transfer-pattern .T
call-forward pattern .T
!
interface Loopback1
description * Used for IPT signaling and SRST registration *
ip address 10.10.10.10 255.255.255.255
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.10.10.10
