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Table D-1
Parameter
call_hold_ringback
call_stats
Cisco SIP IP Phone Administrator Guide, Release 6.x and 7.x
D-2
Configuration File Parameters (continued)
Description
(Phone-specific; optional) If you have a call on hold and are talking
on another call, when you hang up the call, this parameter causes the
phone to ring, letting you know that you still have another party on
hold.
Valid values are as follows:
Default is 0.
(Optional) Includes RTP statistics in BYE requests and responses.
Valid values are 0 (disable) and 1 (enable). Default is 0.
If this parameter is enabled, the phone inserts the headers
RTP-RxStat and RTP-TxStat as follows:
where the following apply:
Appendix D
0—Off by default, but can be turned on and off locally using the
user interface.
1—On by default, but can be turned on and off locally using the
user interface.
2—Off permanently and cannot be turned on and off locally
using the user interface. Specify this parameter in the
phone-specific configuration file.
3—On permanently and cannot be turned on and off locally
using the user interface. Specify this parameter in the
phone-specific configuration file.
RTP-RxStat:
Dur=a,Pkt=b,Oct=c,LatePkt=d,LostPkt=e,AvgJit=f
RTP-TxStat: Dur=g,Pkt=h,Oct=i
Dur—Total number of seconds since the beginning of reception
or transmission.
Pkt—Total number of RTP packets received or transmitted.
Oct—Total number of RTP payload octets received or
transmitted (not including RTP header).
LatePkt—Total number of late RTP packets received.
LostPkt—Total number of lost RTP packets received (not
including late RTP packets).
AvgJit—Average jitter, which is an estimate of the statistical
variance of the RTP packet inter-arrival time, measured in
timestamp unit and calculated according to RFC 1889.
a, b, c, d, e, f, g, h, and i—Integers.
Configurable Parameters for the SIP IP Phone

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