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Beginner

SRST call-manager-fallback on SIP trunk for ISR 4K failing

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

 

 

4 REPLIES 4
Highlighted
Beginner

Re: SRST call-manager-fallback on SIP trunk for ISR 4K failing

please , Also want to understand , steps  of procedure the cisco IP phone does when call-manager connection lost and router become call manager fall back .

 

 

Highlighted
Hall of Fame Cisco Employee

Re: SRST call-manager-fallback on SIP trunk for ISR 4K failing

Are your phones SIP or SCCP?

Did you read the whole SRST configuration guide?

HTH

java

if this helps, please rate
Highlighted
Beginner

Re: SRST call-manager-fallback on SIP trunk for ISR 4K failing

the phones use both SIP and SCCP

 

Highlighted
Hall of Fame Cisco Employee

Re: SRST call-manager-fallback on SIP trunk for ISR 4K failing

There's no SIP SRST configuration, so I recommend you fully review the SRST configuration guide as your first step.

HTH

java

if this helps, please rate
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