Cisco CP-7911G-CH1 Administration Guide - Page 22

Cisco Unified IP Phone 7905G, and Cisco, Not Supported, Phone Features, General

Page 22 highlights

Cisco Unified SIP SRST Cisco Unified SCCP and SIP SRST Feature Overview Table 2 Restrictions from Cisco SIP SRST from the Present Version to Version 3.0(continued) Cisco Unified SRST Cisco IOS Version Release Version 4.0 12.4(4)XC Version 3.4 12.4(4)T Version 3.2 12.3(11)T Version 3.1 12.3(7)T Version 3.0 12.2(15)ZJ 12.3(4)T Restrictions Not Supported • MOH is not supported for a call hold invoked from a SIP phone. A caller hears only silence when placed on hold by a SIP phone. • As of Cisco IOS Release 12.4(4)T, bridged call appearance, find-me, incoming call screening, paging, SIP presence, call park, call pickup, and SIP location are not supported. • SIP-NAT is not supported. • Cisco Unity Express is not supported. • Transcoding is not supported. Phone Features • For call waiting to work on the Cisco ATA and Cisco IP Phone 7912 and Cisco Unified IP Phone 7905G with a 1.0(2) build, the incoming call leg should be configured with the G.711 codec. Note Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G, and Cisco Analog Telephone Adaptor (ATA) 186 are not capable of dual registration; thus they are not supported and have limited functionality with Cisco Unified SIP SRST. General • Call detail records (CDRs) are only supported by standard IOS RADIUS support; CDRs are not supported otherwise. • All calls must use the same codec, either G.729r8 or G.711. • Calls that have been transferred cannot be transferred a second time. • URL dialing is not supported. Only number dialing is supported. • The SIP registrar functionality provided by Cisco Unified SIP SRST provides no security or authentication services. • SIP IP phones that do not support dual concurrent registration with both their primary and their backup SIP proxy or registrar may be unable to receive incoming calls from the Cisco Unified SIP SRST gateway during a WAN outage. These phones may take a significant amount of time to discover that their primary SIP proxy or registrar is unreachable before they initiate a fallback registration to their backup proxy or registrar (the SIP SRST gateway). • SIP-phone-to-SIP-trunk support requires Refer and 302/300 Redirection to be supported by the SIP trunk (Version 3.0). Cisco Unified SCCP and SIP SRST System Administrator Guide 22 OL-13143-04

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Cisco Unified SCCP and SIP SRST Feature Overview
Cisco Unified SIP SRST
22
Cisco Unified SCCP and SIP SRST System Administrator Guide
OL-13143-04
Version 4.0
Version 3.4
Version 3.2
Version 3.1
Version 3.0
12.4(4)XC
12.4(4)T
12.3(11)T
12.3(7)T
12.2(15)ZJ
12.3(4)T
Not Supported
MOH is not supported for a call hold invoked from a SIP phone. A caller hears only
silence when placed on hold by a SIP phone.
As of Cisco IOS Release 12.4(4)T, bridged call appearance, find-me, incoming call
screening, paging, SIP presence, call park, call pickup, and SIP location are not
supported.
SIP-NAT is not supported.
Cisco Unity Express is not supported.
Transcoding is not supported.
Phone Features
For call waiting to work on the Cisco ATA and Cisco IP Phone 7912 and Cisco
Unified IP Phone 7905G with a 1.0(2) build, the incoming call leg should be
configured with the G.711 codec.
Note
Cisco Unified IP Phone 7905G, Cisco Unified IP Phone 7912G, and Cisco
Analog Telephone Adaptor (ATA) 186 are not capable of dual registration; thus
they are not supported and have limited functionality with Cisco Unified SIP
SRST.
General
Call detail records (CDRs) are only supported by standard IOS RADIUS support;
CDRs are not supported otherwise.
All calls must use the same codec, either G.729r8 or G.711.
Calls that have been transferred cannot be transferred a second time.
URL dialing is not supported. Only number dialing is supported.
The SIP registrar functionality provided by Cisco Unified SIP SRST provides no
security or authentication services.
SIP IP phones that do not support dual concurrent registration with both their
primary and their backup SIP proxy or registrar may be unable to receive incoming
calls from the Cisco Unified SIP SRST gateway during a WAN outage. These
phones may take a significant amount of time to discover that their primary SIP
proxy or registrar is unreachable before they initiate a fallback registration to their
backup proxy or registrar (the SIP SRST gateway).
SIP-phone-to-SIP-trunk support requires Refer and 302/300 Redirection to be
supported by the SIP trunk (Version 3.0).
Table 2
Restrictions from Cisco SIP SRST from the Present Version to Version 3.0(continued)
Cisco Unified SRST
Version
Cisco IOS
Release
Restrictions