sip srst confiig

[iptuser55]

I don't really touch SIP SRST but i have a unique situation which forces me to use it. 

Do I just add the following

voice service voip
 
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
 sip
  registrar server
!
!
voice register global
 mode srst
 system message SRST mode
 max-dn 30
 max-pool 30
!
voice register pool  10
 id network 10.1.1.0 mask 255.255.255.0
 codec g711ulaw
!

Will SIP SRST work as above 

or do i need to ccm-manager fallback etc  as below and add the SRST max-dn, max-ephone even though Im not using SCCP?

call-manager-fallback
 max-conferences 8 gain -6
 transfer-system full-consult
 ip source-address 10.1.1.1 port 2000
 max-ephones 30
 max-dn 30 dual-line
 system message primary SRST Backup Mode
 time-zone 40
 time-format 24
 date-format dd-mm-yy
!

Also my PSTN will be H323 FXO ports so I don't need any MGCP fall back so again would i need MGCP part of the configuration or just the SIP Registrar part

Does anyone have a pure sip srst config. they can post

thanks

[islam.kamal]

HI

1-Kindly check the below for your information:-

http://www.cisco.com/en/US/products/sw/voicesw/ps2169/products_configuration_guide_chapter09186a0080557e81.html

2-Kindly find the below config.

voice service voip

no ip address trusted authenticate

ip address trusted list

allow-connections h323 to h323

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
 redirect ip2ip

sip

  bind control source-interface GigabitEthernet0/1

  bind media source-interface GigabitEthernet0/1

  registrar server

voice register global

 mode srst
system message SRST ACTIVE

max-dn 10

max-pool 5

voice register pool  1

id network 14.160.120.0 mask 255.255.255.0   ----->Subnet of sip  phones you want to register to this router

application sip.app

dtmf-relay rtp-nte

codec g711ulaw

3-Yes , if you are not use MGCP , no need . You can only use call fallback active to monitor the next preferred dial-peer.

Finally , make sure  SRST reference with the  IP of the GW

Thanks

Please rate all useful information