SIP Phone SRST Configuration

[shinesalim666]

Dear Team,

We are trying to configure SRST on router.  Phone are registering to call manager and routing through proper gatewat.

But when ever Connection fails to HO our phones are not registering. We are using SIP phones in branch. 

I searched in internet and came to know that there are other commnads er need to apply for SIP phone SRST. I found several documents but all are confusing. 

Please find the SRST configuration on router 

call-manager-fallbac9
 secondary-dialtone 0
 max-conferences 0 gain -6
 transfer-system full-consult
 timeouts interdigit 4
 ip source-address X.X.X.X port 2000
 max-dn 12
 system message primary Your Current Options SRST Mode
 transfer-pattern .T
 call-forward pattern .T
 time-zone 35
 date-format dd-mm-yy

Kindly advise how can we configure SRST for SIP phones.

Thanks in Advance,

Shine Salim

[Jaime Valencia]

Review the SRST configuration guide, it covers SCCP SRST and SIP SRST.

[shyam2014]

Do we need to create individual pool for each SIP phones in the site?

[Ayodeji Okanlawon]

This configuration is for sccp phones, SIP phones require a different config..

1. SIP phones need a Registrar

voice service voip

sip

registrar server

2. You need to enable sip to sip calls

voice service voip
  allow-connections sip to sip

3. Configure voice class codec so that multiple codecs are supported

voice class codec 1
  codec preference 1 g729r8
  codec preference 2 g711alaw
  codec preference 3 g711ulaw

4. Configure voice register global and specify how many phones and dn can register in srst

voice register global
timeouts interdigit 5
system message "SRST Mode"
max-dn 10---change this value to suit you
max-pool 10

5. Configure voice register pool to apply specific details for phones

The id network specify the subnet that is allowed to register to these gateway

voice register pool  1
id network x.x.x.x mask y.y.y.y
dtmf-relay rtp-nte
voice-class codec 1
 

Details from the srst admin guide below:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide.pdf

[fsfiguer95]

---

@Ayodeji Okanlawon wrote:

This configuration is for sccp phones, SIP phones require a different config..

1. SIP phones need a Registrar

voice service voip

sip

registrar server

2. You need to enable sip to sip calls

voice service voip
  allow-connections sip to sip

3. Configure voice class codec so that multiple codecs are supported

voice class codec 1
  codec preference 1 g729r8
  codec preference 2 g711alaw
  codec preference 3 g711ulaw

4. Configure voice register global and specify how many phones and dn can register in srst

voice register global
timeouts interdigit 5
system message "SRST Mode"
max-dn 10---change this value to suit you
max-pool 10

5. Configure voice register pool to apply specific details for phones

The id network specify the subnet that is allowed to register to these gateway

voice register pool  1
id network x.x.x.x mask y.y.y.y
dtmf-relay rtp-nte
voice-class codec 1
 

Details from the srst admin guide below:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide.pdf

---

Good

[Chris Deren]

Many good threads on this topic on this forum, your config is for SCCP SRST, see config guide for SIP SRST:

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cusrst/admin/sccp_sip_srst/configuration/guide/SCCP_and_SIP_SRST_Admin_Guide/srst_41.html