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5055 SIP Phone User Guide (Rev. A)
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emerg_port= 5060
##################################################################################
# Media configuration
##################################################################################
# audio codec to offer
# the codec(s) you choose here, will be listed in the INVITE or OPTION's SDP
# choice 0-g711 uLaw
# 1-g711 ALaw
# 2-g729A
# 3-g729A and g711 uLaw
# 4-all of the above codecs
audio_codec=0
# audio codec packet size
# currently this parameter is only applicable to the g711 codec
# choice [10, 20, 30] ms
audio_pkt_size= 20; ms
# dtmf type
# defines the way the DTMF digits will be sent across
# choice [0-automatic, 1-outband & inband, 2-outband only, 3-inband only
# automatic means when a "telephone-event" is being received from the peer party, send
the
# DTMF digit in the outband-only mode
#
dtmf_type= 0
#dtmf payload(96-127)
dtmf_payload= 96
#Media port configuration start port
start_port= 8000
#Media port configuration start port
end_port= 19998
##################################################################################
# feature configuration
##################################################################################
# auto answer mode
# choice [0-disabled, 1-enabled]
auto_answer= 0
# auto answer reason code
# 0- in a meeting
# 1- out of town
# 2- at lunch
# 3- on vacation
# 4- in a conference
# 5- in lab
# 6- back in 5 minutes
# 7- gone home
# 8- on a course
# 9- off sick
# 10- other reason
reasons= 0
# other reason string
# when reasons = 10, the sip will copy this string to the "reject reason" field.
other_reason= i am busy now!
# do not disturb
# choice [0 - disable, 1-enable]
# when enabled, all incoming calls will be rejected, or forwarded to the voicemail
do_not_disturb= 0
# call forwarding no answer mode
# choice [0-disabled, 1- enabled]
# when enabled, the incoming call will be forwarded to the "noans_fwd_addr"