09-10-2015 06:14 AM - edited 03-17-2019 04:16 AM
I'm trying to configure SRST on our gateway when CUCM goes down.
Our auto attendant is on DN 2000 in CUCM (goes to UCCX)
Our sample phones are 2001, 2339, 2340, 2810, 2820
2001 is a SIP 7821 while the others are SCCP phones
When the WAN is up, all phones ring instead of being answered by the auto attendant and are registered with CUCM.
When the WAN goes down, all phones ring as expected and are registered in SRST with the gateway.
In addition, when a call is answered on the SIP phone through SRST, no audio is heard. When a call is answered on the SCCP phone, there is audio.
I have tried using the alias command in call-manager-fallback; I set alias 1 2000 to 1999 and changed my hunt pilot to 1999 but that didn't work at all. If I set alias 1 2000 to 2339, my phone rings but not in the manner that I wish
I've pasted my config below. Any advice would be incredibly helpful
ROUTER#show run
Building configuration...
Current configuration : 5602 bytes
!
! Last configuration change at 08:54:37 EDT Thu Sep 10 2015 by USERNAME
! NVRAM config last updated at 08:54:38 EDT Thu Sep 10 2015 by USERNAME
! NVRAM config last updated at 08:54:38 EDT Thu Sep 10 2015 by USERNAME
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname ROUTER
!
boot-start-marker
boot system flash:c2800nm-adventerprisek9-mz.151-4.M10.bin
boot-end-marker
!
!
logging buffered 4096
!
no aaa new-model
!
clock timezone EST -5 0
clock summer-time EDT recurring
!
dot11 syslog
ip source-route
!
!
ip cef
!
ip dhcp excluded-address 172.16.16.1 172.16.16.20
!
ip dhcp pool voice
import all
network 172.16.16.0 255.255.255.0
default-router 172.16.16.1
option 150 ip 172.16.16.4
dns-server 10.10.0.5
lease 0 2
!
!
ip domain name lab.DOMAIN.com
ip name-server 172.16.16.1
ip name-server 10.10.0.5
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
!
!
!
voice service voip
ip address trusted list
ipv4 XX.XX.XX.XX
ipv4 172.10.10.0 255.255.255.0
ipv4 172.16.16.0 255.255.255.0
ip address trusted call-block cause not-in-cug
gcid
clid substitute name
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
e911
registrar server expires max 600 min 60
transport switch udp tcp
asserted-id ppi
midcall-signaling passthru
no call service stop
!
voice class sip-profiles 1
request INVITE sip-header From modify "From: (.*<)(.*>)" "From: \"Cisco Lab\" <\2"
!
!
voice register global
timeouts interdigit 5
max-dn 20
max-pool 10
!
voice register pool 1
id network 172.16.16.0 mask 255.255.255.0
dtmf-relay rtp-nte cisco-rtp
codec g711ulaw
!
voice hunt-group 20 parallel
list 2001,2339,2340
timeout 20
pilot 2000
!
!
!
!
voice translation-rule 1
rule 1 /.*/ /2000/
!
voice translation-rule 2
rule 1 /.*/ /5555555555/
!
voice translation-rule 3
rule 1 /^9\(.*\)/ /\1/
!
!
voice translation-profile INCOMING
translate called 1
!
voice translation-profile OUTGOING
translate calling 2
translate called 3
!
!
voice-card 0
!
crypto pki token default removal timeout 0
!
crypto pki trustpoint TP-self-signed-910207751
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-910207751
revocation-check none
rsakeypair TP-self-signed-910207751
!
!
!
!
license udi pid CISCO2811 sn FTX1126A20Z
archive
log config
hidekeys
username USERNAME privilege 15 secret 5 XXXXXXXXXXXXXXXXXXXXXXXX
!
redundancy
!
!
!
!
!
!
!
!
!
!
interface Loopback0
no ip address
!
interface FastEthernet0/0
description DHCPVoIP
ip address 172.16.16.2 255.255.255.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip forward-protocol nd
no ip http server
no ip http secure-server
!
!
ip route 0.0.0.0 0.0.0.0 172.16.16.1
!
access-list 23 permit 172.16.16.0 0.0.0.255
access-list 23 permit 172.10.10.0 0.0.0.255
access-list 23 permit 10.10.0.0 0.0.0.255
!
!
!
!
!
tftp-server flash:apps75.9-4-2ES9.sbn
tftp-server flash:cnu75.9-4-2ES9.sbn
tftp-server flash:cvm75sccp.9-4-2ES9.sbn
tftp-server flash:dsp75.9-4-2ES9.sbn
tftp-server flash:jar75sccp.9-4-2ES9.sbn
tftp-server flash:SCCP75.9-4-2SR1-1S.loads
tftp-server flash:term75.default.loads
tftp-server flash:Desktops/320x216x16/List.xml
tftp-server flash:Desktops/320x216x16/DOMAIN.png
tftp-server flash:Desktops/320x216x16/TN-DOMAIN.png
tftp-server flash:Desktops/320x196x4/List.xml
tftp-server flash:Desktops/320x196x4/DOMAINBW.png
tftp-server flash:Desktops/320x196x4/TN-DOMAINBW.png
!
control-plane
!
!
!
!
mgcp profile default
!
!
dial-peer voice 1 voip
translation-profile incoming INCOMING
session protocol sipv2
incoming called-number 189199
dtmf-relay cisco-rtp rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 2 voip
translation-profile outgoing OUTGOING
destination-pattern 9[2-9]..[2-9]......
session protocol sipv2
session target ipv4:XX.XX.XX.XX
no voice-class sip early-offer forced
voice-class sip profiles 1
dtmf-relay rtp-nte cisco-rtp sip-kpml sip-notify
codec g711ulaw
no vad
!
dial-peer voice 3 voip
destination-pattern 2...
session protocol sipv2
session target ipv4:172.16.16.4
dtmf-relay cisco-rtp rtp-nte
codec g711ulaw
!
dial-peer voice 13 voip
preference 1
destination-pattern 2...
session protocol sipv2
session target ipv4:172.16.16.2
dtmf-relay cisco-rtp rtp-nte
codec g711ulaw
!
dial-peer voice 14 voip
preference 1
session protocol sipv2
session target ipv4:172.16.16.2
dtmf-relay cisco-rtp rtp-nte
codec g711ulaw
!
!
sip-ua
credentials username 189199 password 7 XXXXXXXXXXXXXXXXXXXXXXXX realm SIPPROVIDER.com
authentication username 189199 password 7 XXXXXXXXXXXXXXXXXXXXXXXX realm SIPPROVIDER.com
registrar 1 dns:SIPPROVIDER.com expires 300
!
!
!
call-manager-fallback
secondary-dialtone 9
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 172.16.16.2 port 2000
max-ephones 10
max-dn 20
keepalive 20
time-zone 12
time-format 24
date-format dd-mm-yy
!
!
banner exec ^C
Authorized access only. This system is the property of DOMAIN. Disconnect IMMEDIATELY if you are not an authorized user!
^C
banner login ^C
Authorized access only. This system is the property of DOMAIN. Disconnect IMMEDIATELY if you are not an authorized user!
^C
!
line con 0
login local
line aux 0
line vty 0 4
access-class 23 in
privilege level 15
login local
transport input ssh
!
scheduler allocate 20000 1000
ntp update-calendar
ntp server 192.67.222.4
end
ROUTER#
09-10-2015 07:00 AM
If you want enhanced routing capabilities, you probably want to look into CME-as-SRST instead of plain SRST.
09-10-2015 07:38 AM
Im looking at this
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmesrst.html
Everything is for SCCP phones it seems. Is there an equivalent for SIP?
09-10-2015 08:43 AM
Hi,
I am not sure if I understand your requirement. If your question whether SIP supports SRST, the answer is yes.
Also, SIP SRST support hunt groups but as Jamie mentioned, you need CME-SRST instead of traditional SRST
11-04-2015 01:24 PM
Bringing this back from the dead
Under normal circumstances, when a call comes in on our gateway it is sent to 2000 which is a CTI port for UCCX
When we're in SRST mode, I would like all calls to go to extension 2339 on a SIP phone
If I modify my translation profile in my GW to point to 2339, it works. How would I go about having it automatically change to 2339 from 2000 when we fall to SRST mode?
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