02-16-2013 04:17 AM - edited 03-16-2019 03:45 PM
Hello,
I try to implement SRST feature on a 2901 router. There's already a MGCP link between router and CUCM (8.6) which is OK (FXS port is mounted for analog phone on CUCM).
But SRST doesn't work.
CUCM config is OK (System => SRST => Name & IP address, and System => Device Pool => SRST=2901), and phone have the correct IP address of the SRST router in the CM List. IP address of the CUCM is 10.1.1.10.
I try to begin by a very simple config, here it is :
!
! Last configuration change at 13:30:24 UTC Fri Feb 15 2013
! NVRAM config last updated at 11:28:20 UTC Fri Feb 15 2013
! NVRAM config last updated at 11:28:20 UTC Fri Feb 15 2013
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
!
hostname r2901srst
!
boot-start-marker
boot-end-marker
!
!
!
no aaa new-model
network-clock-participate wic 0
network-clock-participate wic 1
!
no ipv6 cef
ip source-route
ip cef
!
!
application
global
service alternate Default
!
!
ip domain name democisco.com
multilink bundle-name authenticated
!
!
!
!
isdn switch-type basic-net3
!
crypto pki token default removal timeout 0
!
!
voice-card 0
!
!
!
!
!
!
!
license udi pid CISCO2901/K9 sn FCZ********
hw-module pvdm 0/0
!
!
!
!
redundancy
!
!
!
!
!
!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
ip address 10.0.3.102 255.255.255.0
duplex auto
speed auto
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface BRI0/0/0
no ip address
isdn switch-type basic-net3
isdn timer T310 6000
isdn overlap-receiving
isdn point-to-point-setup
isdn incoming-voice voice
isdn bind-l3 ccm-manager service mgcp
isdn send-alerting
isdn sending-complete
isdn outgoing-voice info-transfer-capability 3.1kHz-audio
!
interface BRI0/0/1
no ip address
!
interface BRI0/1/0
no ip address
!
interface BRI0/1/1
no ip address
!
ip forward-protocol nd
!
no ip http server
no ip http secure-server
!
ip route 0.0.0.0 0.0.0.0 10.0.3.1
!
!
!
!
control-plane
!
!
voice-port 0/0/0
compand-type a-law
cptone FR
!
voice-port 0/0/1
!
voice-port 0/1/0
!
voice-port 0/1/1
!
voice-port 0/2/0
!
voice-port 0/2/1
!
voice-port 0/2/2
!
voice-port 0/2/3
!
voice-port 0/3/0
!
voice-port 0/3/1
!
voice-port 0/3/2
!
voice-port 0/3/3
!
ccm-manager mgcp
!
mgcp
mgcp call-agent 10.1.1.10 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
!
mgcp profile default
!
!
dial-peer voice 100 pots
port 0/0/0
!
dial-peer voice 103 pots
service mgcpapp
port 0/2/0
!
!
!
!
gatekeeper
shutdown
!
!
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 10.1.1.10 port 2000
max-ephones 10
max-dn 10
!
!
!
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
login
transport input all
!
scheduler allocate 20000 1000
end
Please help if you see what can be wrong !
Thank you,
Clement
Solved! Go to Solution.
02-16-2013 04:35 AM
Dear
Based on your configuration , i found that you type the IP address of your call manager under call- manager-fallback
Solution
call-manager-fallback
ip source-address 10.0.3.102 port 2000 ( The IP address must be the IP of the router which will replace call manager).Please check also the ip of SRST router Cisco Unified CMAdministration, choose System > SRST > Add New
Thank you
Please rate if this will help
02-16-2013 05:56 AM
1) change ip source address under fallback to the ip address of this router as per above post.
There are another couple of things missing in the config came, without which SRST will not be fully functional:
1) add: "ccm-manager fallback-mgcp" - to enable the gateway failover from mgcp to h323
2) add outbound dial-peers(there is an incomplete 100 dial peer): example, at least a basic one sending all calls to PSTN, if 0 is access code:
Dial-peer voice 100 pots
Destination-patt 0T
Port 0/0/0
3) Incoming dial-peer and translation rule for incoming calls to your DID numbers during SRST, lets say your extension are 1XXX:
Voice translation-rule 1
r 1 /.*\(1...$\)/ /\1/
Voice translation-pro FR-PSTN
t calle 1
dial-peer voice 1 pots
Incoming called-num .
Translation-pro in FR-PSTN
Direct
*PS* - Excuse syntax error if any, I dont have CLI nearby, sending from phone.
Sent from Cisco Technical Support iPhone App
02-16-2013 04:35 AM
Dear
Based on your configuration , i found that you type the IP address of your call manager under call- manager-fallback
Solution
call-manager-fallback
ip source-address 10.0.3.102 port 2000 ( The IP address must be the IP of the router which will replace call manager).Please check also the ip of SRST router Cisco Unified CMAdministration, choose System > SRST > Add New
Thank you
Please rate if this will help
02-16-2013 05:56 AM
1) change ip source address under fallback to the ip address of this router as per above post.
There are another couple of things missing in the config came, without which SRST will not be fully functional:
1) add: "ccm-manager fallback-mgcp" - to enable the gateway failover from mgcp to h323
2) add outbound dial-peers(there is an incomplete 100 dial peer): example, at least a basic one sending all calls to PSTN, if 0 is access code:
Dial-peer voice 100 pots
Destination-patt 0T
Port 0/0/0
3) Incoming dial-peer and translation rule for incoming calls to your DID numbers during SRST, lets say your extension are 1XXX:
Voice translation-rule 1
r 1 /.*\(1...$\)/ /\1/
Voice translation-pro FR-PSTN
t calle 1
dial-peer voice 1 pots
Incoming called-num .
Translation-pro in FR-PSTN
Direct
*PS* - Excuse syntax error if any, I dont have CLI nearby, sending from phone.
Sent from Cisco Technical Support iPhone App
02-16-2013 06:39 AM
Thank you guys for answers.
I think you are right.
As my router is in my office, I will test on monday to change ip-source address to router's IP, add ccm-manager fallback-mgcp, and add dial-peers.
I've already think about dial-peers, but firstly I want my phones register correctly.
Give you news and rates next week, I will now think about register third party SIP phones when SRST activated.
Thank you so much,
Clement
02-16-2013 07:04 AM
No problem - let us know if you have further questions.
Terry
Sent from Cisco Technical Support iPhone App
02-16-2013 09:11 AM
Dear
1- Firstly check the registeration is ok , after that please use the (R#sh call-manager-fallback).
2- After this you will have to add all your dial-peers for example as below.
dial-peer voice 200 pots
destination-pattern 9[2-9].........
port 0/0/1:23
!
dial-peer voice 201 pots
destination-pattern 91[2-9].........
port 0/0/1:23
!
dial-peer voice 202 pots
destination-pattern 9011...........
port 0/0/1:23
!
dial-peer voice 203 pots
destination-pattern 9911
port 0/0/1:23
Thank you
please rate if this will help
02-17-2013 09:31 AM
It works perfectly guys !
Feb 17 17:23:32.143: %IPPHONE-6-REGISTER_NEW: ephone-1:SEP00179551F950 IP:10.0.3.3 Socket:1 DeviceType:Phone has registered.
Feb 17 17:23:57.003: %SYS-5-CONFIG_I: Configured from console by console
Feb 17 17:23:57.055: %LINK-5-CHANGED: Interface BRI0/0/0, changed state to administratively down
Feb 17 17:23:58.203: %SYS-5-CONFIG_I: Configured from console by console
Feb 17 17:24:01.415: %SYS-5-CONFIG_I: Configured from console by console
Feb 17 17:24:01.455: %LINK-3-UPDOWN: Interface BRI0/0/0, changed state to up
Feb 17 17:24:42.991: %CALL_CONTROL-6-APP_NOT_FOUND: Application mgcpapp in dial-peer 103 not found. Handing callid 10 to the alternate app .
But I have a problem with an analog phone wired to a FXS port (it's the dial-peer voice 103 pots in my config). IP phone have correct tone, but I've a busy one on the analog. The message which appears in the console is the last in my log above.
I know to activate fallback on SRST for FXS port, I have to use SCCP as shown in this guide : http://www.cisco.com/en/US/docs/ios/voice/fxs/configuration/guide/fxssrst.pdf
In addition, I've add destination-pattern 1105 into the dial-peer of the analog phone, which permits it to receive calls but no to initiate a call.
My config is :
ccm-manager fallback-mgcp
ccm-manager mgcp
!
mgcp
mgcp call-agent 10.1.1.10 service-type mgcp version 0.1
mgcp dtmf-relay voip codec all mode out-of-band
!
mgcp profile default
!
sccp local GigabitEthernet0/0
sccp ccm 10.0.3.102 identifier 20 version 7.0
sccp ccm 10.1.1.10 identifier 10 version 7.0
sccp
!
sccp ccm group 1
description ### Groupe de CCM pour prise en charge SCCP sur ports FXS ###
associate ccm 10 priority 1
associate ccm 20 priority 2
switchback method graceful
!
dial-peer voice 100 pots
port 0/0/0
!
dial-peer voice 103 pots
service mgcpapp
destination-pattern 1105
port 0/2/0
!
call-manager-fallback
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 10.0.3.102 port 2000
max-ephones 10
max-dn 10
system message primary /!\ Mode de secours actif /!\
system message secondary Mode degrade actif
It might be OK following instructions in the guide ?
Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: