Cisco Unified SRST Administration Guide (All Versions) - Setting Up Cisco Unified IP Phones using SIP

Publisher

Cisco Systems

URL

https://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_setting_up_using_sip.html

Published

2026-04-25

Updated

unknown

Accessed

2026-09-25

HTTP status

200

License

Cisco documentation; all rights reserved; no-redistribution; short excerpts and locators only

Source notes citing this source

• When a SIP phone registers and matches a voice register pool that has a proxy configured, SRST automatically creates two dial peers: one routing to the proxy and a fallback dial peer routing to the phone. in context
• A voice register pool uses id network/mask, id ip/mask or id mac to decide which SIP phone registrations the SRST router accepts, and can carry dial-peer attributes such as dtmf-relay and voice-class codec. in context

Cite this source record

APA

WarmTransfer. (2026, April 25). Cisco Unified SRST Administration Guide (All Versions) - Setting Up Cisco Unified IP Phones using SIP. WarmTransfer. https://warmtransfer.net/knowledge/sources/cisco-srst-admin-setup-sip-phones

BibTeX

@misc{warmtransfer-cisco-srst-admin-setup-sip-phones,
  title  = {Cisco Unified SRST Administration Guide (All Versions) - Setting Up Cisco Unified IP Phones using SIP},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/knowledge/sources/cisco-srst-admin-setup-sip-phones},
  note   = {Cisco Systems, accessed 2026-09-25}
}