iPECS eMG80/100& eMG800 & UCP & vUCP
Administration and Programming Manual Issue 2.5
278
Table 4.4.8.7-1 SIP PHONE ATTRIBUTES
DTMF dialing signals sent by the SIP phone must be defined for the
system to detect the tones properly. For Inband DTMF, a VoIP
channel is required.
INBAND
2833
INFO(SIMPLE DTMF)
INFO(NORTEL NETWORKS)
INFO(DTMF RELAY)
INFO(TELEPHONE-EVENT)
The Short Message Service Protocol (type) must be selected to
support SMS.
AUTO,
Text/plan,
Text/plan(KR),
Xnipm+xml
When the user of a SIP phone dials a CO/IP Line access code with
Enblock dialing, the system can provide a virtual dial tone to the
user.
For compatible SIP phones, the system supports SIP
Subscribe/Notify. When enabled here, the system sends Message
Waiting notifications to the SIP phone.
OFF, message-summary
The SIP Request header Domain field can use the SIP phone’s IP
address and port (Normal), or for ‘KT-FMC,’ the Request URI
Domain field will be system IP and port.
To indicate a busy condition to the SIP phone, the system can
provide RTP packets with a busy tone or the SIP 486 Busy
message. Providing a busy tone requires a VoIP DSP channel in
the system.
System Busy Tone, 486 Busy Message
The system can route calls to the SIP phone while busy (Multiple).
In this case, the SIP phone determines if Call Waiting is supported.
Otherwise, if the SIP phone is busy, the system routes call based
on the busy treatment (Single).
Pre Audio
Connection For
DTMF
The system normally provides the 183 Session Progress SIP
message to establish a “Pre-audio” connection.
The “Pre-audio” connection permits the system to send tones (CO
dial tone or ringback tone) to the SIP phone.
In addition, the SIP phone can send DTMF tones the user dials in
response to CO dial tone or a remote IVR message.
Some SIP phones may require the 200 OK messages, which
“answer” the call, to allow dialing after the call has been initiated.
183 Session Progress,200 OK