Cisco Cisco IOS Software Release 12.2(11)YT

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      SIP Call Transfer and Call Forwarding Supplementary Services
Restrictions for SIP Call Transfer and Call Forwarding Supplementary Services
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Cisco IOS Release 12.2(11)YT
Write a Tool Command Language (TCL) Interactive Voice Response (IVR) 2.0 script that implements 
Cisco IOS call transfer and forward supplementary services functionality. 
Restrictions for SIP Call Transfer and Call Forwarding 
Supplementary Services
The SIP Call Transfer and Call Forwarding Supplementary Services feature is supported only 
through TCL IVR 2.0 and VoiceXML applications; it is not supported for TCL IVR 1.0 applications 
or the DEFAULT session application.
Although SIP Cisco IOS gateways currently support SIP URLs and TEL URLs, the Refer-To header 
and the Also header must be in SIP URL format to be valid. The TEL URL format cannot be used 
because it does not provide a host portion, and without one, the triggered Invite request cannot be 
routed. 
Cisco SIP customer premise equipment (CPE) such as 79xx and Analog Telephone Adaptors (ATAs) 
do not currently support TEL URLs.
The Refer-To and Contact headers are required in the Refer request. The absence of either header 
results in a 4xx class response to the Refer request. Also, the Refer request must contain exactly one 
Refer-To header. Multiple Refer-To headers result in a 4xx class response.
The Referred-By header is required in a Refer request. The absence of this header results in a 4xx 
class response to the Refer request. Also, the Refer request must contain exactly one Referred-By 
header. Multiple Referred-By headers result in a 4xx class response.
Only RLT on CAS or analog (FXS) ports is supported with SIP call transfers.
The Cisco AS5xxx platforms do not support hookflash detection for T1 CAS.
SIP call forwarding is supported only on e-phones—IP phones that are not configured on the 
gateway. FXS, FXO, T1, E1, and CAS phones are not supported.
In Cisco IOS Release 12.2(11)YT, when SIP with e-phones is used, DTMF is not supported. Voice 
can be established, but DTMF cannot be relayed in- or out-of-band. Custom scripting is also 
necessary for e-phones to initiate call forwarding. The standard configurations listed in this 
document work only when an e-phone is the recipient or final-recipient.
Information About SIP Call Transfer and Call Forwarding 
Supplementary Services
To configure the SIP Call Transfer and Call Forwarding Supplementary Services feature, you must 
understand the following concepts: