iPECS eMG80 & eMG800 & UCP
Administration and Programming Manual Issue 1.6
466
< DTMF >
1) 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)
2) 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.
Table 4.4.8.7-1 SIP PHONE ATTRIBUTES
ATTRIBUTE DESCRIPTION RANGE DEFAULT
Registration Mode
Initial registration of a SIP phone is accomplished
employing the parameters set in Station User Login
Manual
Register
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