iPECS eMG80/100& eMG800 & UCP & vUCP
Administration and Programming Manual Issue 2.5
275
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
It 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 Phones will reject Request-URI if IP and port in the domain field are different from its contact IP and
port.
‘Normal’: IP and port number in the 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 a busy state that is presented by the system. Additional
VOIB/VOIU/VOIM DSP channel is required.486 Busy Message: there will be a ‘486 busy’ SIP signaling
response on the busy state.
Call Initiation Mode
For a SIP station, the system can establish multiple or single call sessions. With this option, the system can
control ‘call wait option on the system side. Normally, the SIP Phone has its ‘call-wait ON/OFF’ option by itself.
Options are as below:
‘Multiple’: The system initiates every SIP Phone call regarding its idle state for a new additional call to
SIP Phone. The ‘call-wait allow/deny’ is decided by SIP Phone itself.
‘Single’: system initiates only one call for a SIP Phone. The system side denies the ‘call wait. So, a
call to a busy SIP station will be implemented on a busy state call control.
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