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iPECS eMG80/100& eMG800 & UCP & vUCP
Administration and Programming Manual
Issue 2.3
483
DTMF
INFO (OUT BAND) type DTMF
Presented in SIP signaling message.
INFO (SIMPLE DTMF) / INFO (NORTEL NETWORKS) / INFO (DTMF RELAY) / INFO
(TELEPHONE EVENT)
Default: INFO (DTMF RELAY)
TONE (INBAND) type DTMF
Presented in RTP packet
Additional VOIU/VOIB (VOIM) DSP channel is required to detect DTMF in RTP
INBAND / 2833
CO DIAL TONE
This is to avoid double play of CO dial tone. ‘Set’ if SIP phone plays CO dial tone by itself. If not, there will be
another CO dial tone from external.
Request URI Type
Some SIP Phone will reject Request-URI if IP and port in domain field is different from its contact IP and port.
‘Normal’: IP and port number in Request URI domain field will be the real IP and port number of the
SIP phone.
‘KT-FMC’: IP and port number in Request URI domain field will be system IP and port
Busy Serve
System Busy Tone: there will be a ‘busy-tone’ on busy state that is presented by system.Additional
VOIB/VOIU/VOIM DSP channel is required.486 Busy Message: there will be ‘486 busy’ SIP signaling
response on busy state.
Call Initiation Mode
For a SIP station, system can establish multiple or single call sessions. With this option system can control
‘call-wait’ option in system side. Normally, the SIP Phone has its ‘call-wait ON/OFF’ option by itself.Options are
as below:
‘Multiple’: for a new additional call to SIP Phone, system initiates every call for the SIP Phone
regarding it is on idle state. The ‘call-wait allow/deny’ is decided by SIP Phone itself.
‘Single’: system initiates only one call for a SIP Phone. The ‘call-wait’ is denied by system side. So, a
call to a busy SIP station will be implemented on a busy state call-control.