SRST Configuration on SIP Gateway

[mesut ozcan]

Dear all,

I need to configure SRST feature kindly below topology;

• CUCM v10.5.1
• Router 2911 - 15.4(3)M2
• CUCM -(SIP)-> Voice GW -(PRI)-> ITSP
• Location Phone Types 7841,8945,8961 (SIP)

There is almost 40 phones.

Could you please share some sample configurations or guides with me about this?

Best Regards,

Mesut

[Vivek Batra]

voice service voip
allow-connections sip to sip
sip
registrar server
!
voice register global
 mode srst
 system message SRST
 max-dn 40
 max-pool 40
!
voice register pool  1
 id network 10.10.10.0 mask 255.255.255.0
 dtmf-relay rtp-nte sip-notify
 voice-class codec 1
!

- Vivek

[Ayodeji Okanlawon]

Now I have rated 5 :)

[mesut ozcan]

Hi,

I have configured my router as kindly below but i think something missing.

I do not configure any sip registrar IP adress or any calling privileges to support incoming outgoing calls. Is it normal?

Could you please check below config and let me know if something missing or wrong?

!!!!!

voice service voip
ip address trusted list
ipv4 10.210.255.11
ipv4 10.210.255.12
ipv4 10.210.255.13
ipv4 10.210.255.14
ipv4 10.210.255.15
ipv4 10.210.254.2
ipv4 10.210.254.6
ipv4 191.0.0.0 255.255.255.0
ipv4 10.210.100.0 255.255.255.0
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 expires max 600 min 60
!!!!!!

!!!!!!

voice register global
mode srst
max-dn 100
max-pool 15
!
voice register pool 11
id network 10.210.24.0 mask 255.255.255.0
preference 2
proxy 10.210.24.253 preference 1
no digit collect kpml
dtmf-relay rtp-nte sip-notify
codec g711ulaw
no vad
!!!!!!

!!!!!!!

!
call-manager-fallback
secondary-dialtone 0
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 10.210.24.253 port 2000
max-ephones 35
max-dn 100 dual-line
system message primary CM Fallback Service Operating
system message secondary CM Fallback Service
time-format 24
date-format dd-mm-yy
!
!!!!!!!

[Ayodeji Okanlawon]

Your configuration is fine..

Why do you need these two commands..Please remove them unless you know what you are doinbg with them.

preference 2
proxy 10.210.24.253 preference 1

This is SRST and I will not worry about calling privileges. Any WAN outage that leads to SRS is supposed to be short lived, so no need for calling privileges

[mesut ozcan]

Ok,thanks.

What about COR lists exc? Is this config enough to register SIP Phones with SRST and call other phonesi POTS or get called by outsite?

Regards,

Mesut

[Ayodeji Okanlawon]

I dont know what you want me to tell you. I already answered your questions. All of them. It seems you dont believe what I have said.

Why do you need or want corlists in SRST? Are your users going to be forever in SRST? Perhaps they will have lunch, dinner while in SRST :)

I dont know what your full dial plan looks like. You may need to make additional modifications for POTS line or your Phones both sip and sccp to be reachable from outside in SRST.

You need to look at the ff

1. What are your DDI range

2. what are your phone extensions

3. How many digits do your telco send in

4.Do you need to do any digit manipulation for the calls to ring on phones..

Example if your DDI is 0114445000 and your extension is 5000..and your telco sends 10 digit..When the calls come in to your gateway it will come as 0114445000. You will thenneed to use voice translation rules to match this to extension 5000

[mesut ozcan]

The main purpose of SRST is to support connection between internal and external voice network.

That is why i need to figure out these parts of SRST.

I already thank you for all helps and this question is not only for you btw :)

Any other comment from any other one?

I need SRST to connect internal IP Phones users with external POTS system during WAN outage.

Best Regards,

Mesut

[Ayodeji Okanlawon]

My friend take some time and do some study. SRST is not to connect internal and external voice network. I wish you all the very best in your endeavours

[mesut ozcan]

Thank you my friend.

[devils_advocate]

SRST is designed to be an 'all in one device' (call manager and gateway) for when your CUCM server (s) fail.

If CUCM were to be unavailable (for whatever reason), the phones would reregister with the SRST device. The SRST device would handle the registrations of the IP phones and you would configure dial peers for call routing.

The configuration you were given above by Vivek and Ayodeji is purely for the SRST function of the device.

You still need to configure the relevant PSTN voice ports, as well as the dial peers to support inbound and outbound calling.

Effectively the configuration is almost the same as being a CME device in terms of the dial peers and voice ports except you don't need to route calls towards CUCM as the phones are locally registered.

[mesut ozcan]

It is a SIP gateway, thats why there are some dial-peer exc. configurations.

I just need config steps to handle inbound/outbound calls. 

Anyway thank you for your helps in advance.

Best Regards,

Mesut

[Vivek Batra]

My friend Deji has quoted nicely in the previous thread, you need both inbound and outbound dial peers for SRST to work effectively.

I assume you are comfortable with translation rules/profiles. 

In the normal operation (when not in SRST), if you do most of the digit manipulation in gateway itself (instead of in CUCM), same digit manipulation may work in SRST too.

For outbound calls, you need outbound POTS dial peers as per your dial plan. You may need to apply calling/called translation rules in outbound direction to manipulate DNIS and ANI.

- Vivek 

[mesut ozcan]

Dear all,

My config is kindly below.

Could you please check and say if i am ready to test or not?

Users call outside with prefix 0. All incoming calls will be answer by operator during SRST. Operator number is 343001.

I just wonder how it will work unless any configuration about phones, DNs and COR lists.

!
!
voice service voip
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
sip
registrar server expires max 600 min 60
!
!
!
!
voice register global
mode srst
max-dn 100
max-pool 15
!
voice register pool 11
translation-profile outgoing IPtoPSTN
id network 10.210.X.0 mask 255.255.255.0
preference 2
incoming called-number
proxy 10.210.X.253 preference 1
alias 1 020182... to 343001 preference 2
no digit collect kpml
dtmf-relay rtp-nte sip-notify
codec g711ulaw
no vad
!
!
!
voice translation-rule 1
rule 1 /^0/ //
!
!
voice translation-profile IPtoPSTN
translate called 1
!
!

!
!
!
dial-peer voice 1 voip
incoming called-number .
voice-class codec 1
!
dial-peer voice 2 voip
destination-pattern 020182...
session protocol sipv2
session target ipv4:10.210.255.X
voice-class codec 1
dtmf-relay rtp-nte
!
dial-peer voice 10 pots
description Intercity and Mobile
destination-pattern 0[3-9]........
port 0/0/0:15
forward-digits all
!
dial-peer voice 11 pots
destination-pattern 00T
port 0/0/0:15
forward-digits all
!
dial-peer voice 12 pots
destination-pattern 02.......
port 0/0/0:15
forward-digits all
!
!
!
!
gatekeeper
shutdown
!
!
call-manager-fallback
secondary-dialtone 0
max-conferences 8 gain -6
transfer-system full-consult
ip source-address 10.210.24.X port 2000
max-ephones 35
max-dn 100 dual-line
system message primary CM Fallback Service Operating
system message secondary CM Fallback Service
time-format 24
date-format dd-mm-yy
!
!

[Ayodeji Okanlawon]

in additon to Viveks's post you need additional configuration like the registrar ip address \and cucm config,,

voice service voip
 allow-connections sip to sip

sip

  registrar server expires max 600 min 60

!

voice register pool  10
 id network 172.26.10.0 mask 255.255.255.0
 dtmf-relay rtp-nte sip-notify
 codec g711ulaw
 no vad
!
voice register pool  11
 id network 172.26.11.0 mask 255.255.255.0
 dtmf-relay rtp-nte sip-notify
 codec g711ulaw
 no vad
!

sip-ua
 registrar ipv4:172.26.10.240 expires 600

(172.26.10.240 is the SRST gateway IP address).

### CUCM CONFIG ####

On UCM SRST reference configuration for gateway

SIP Network/IP Address 172.26.10.240
SIP Port 5060