10-24-2014 05:27 PM - editado 03-17-2019 12:40 AM
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
em 10-24-2014 10:19 PM
Review the SRST configuration guide, it covers SCCP SRST and SIP SRST.
em 01-25-2015 08:18 AM
Do we need to create individual pool for each SIP phones in the site?
em 10-24-2014 11:02 PM
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:
em 08-27-2019 10:08 AM
@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 sip3. 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 g711ulaw4. 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 105. 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:
Good
em 01-25-2015 12:41 PM
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
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