CUBE Dial-peer question

[richardmoorehead]

I support a client with several CUBEs, one of which has many dial peers which I believe aren't being used at all.  When I run a "show dial-peer voice [interface]" I get an output like the example below (details scrubbed).  My question is, the last few lines list Last Disconnect and Last Setup TIme.  In what human readable format is that presented?  The Last Disconnect Time from the example is 2087895922.

I checked the Last Disconnect Time from a dial-peer that is constantly used and it showed 2086372430, which if the numbers increment like you'd think would be a point in the past compared to the example's.  I also ran a "show logging" and got a timestamp of 1377198297.  Am I barking up the wrong tree entirely?

Basically I am trying to find a way to display actual dial-peer throughput so I can clean up this config.

cube#show dial-peer voice 600
VoiceOverIpPeer600
peer type = voice, system default peer = FALSE, information type = voice,
description = `TEMP FAX TO ATT',
tag = 600, destination-pattern = `1234567890',
voice reg type = 0, corresponding tag = 0,
allow watch = FALSE
answer-address = `', preference=0,
CLID Restriction = None
CLID Network Number = `'
CLID Second Number sent
CLID Override RDNIS = disabled,
rtp-ssrc mux = system
source carrier-id = `', target carrier-id = `',
source trunk-group-label = `', target trunk-group-label = `',
numbering Type = `unknown'
group = 600, Admin state is up, Operation state is up,
incoming called-number = `', connections/maximum = 0/unlimited,
bandwidth/maximum = 0/unlimited,
DTMF Relay = enabled,
modem transport = passthrough, nse, payload type = 100, codec = g711ulaw, ,
URI classes:
Incoming (Request) =
Incoming (Via) =
Incoming (To) =
Incoming (From) =
Destination =
huntstop = disabled,
in bound application associated: 'DEFAULT'
out bound application associated: ''
dnis-map =
permission :both
incoming COR list:maximum capability
outgoing COR list:minimum requirement
outgoing LPCOR:
Translation profile (Incoming):
Translation profile (Outgoing):REMOVE-AREACODE
incoming call blocking:
translation-profile = `'
disconnect-cause = `no-service'
advertise 0x40 capacity_update_timer 25 addrFamily 4 oldAddrFamily 4
mailbox selection policy: none
type = voip, session-target = `ipv4:X.X.X.X',
technology prefix:
settle-call = disabled
ip media DSCP = ef, ip media rsvp-pass DSCP = ef
ip media rsvp-fail DSCP = ef, ip signaling DSCP = cs3,
ip video rsvp-none DSCP = af41,ip video rsvp-pass DSCP = af41
ip video rsvp-fail DSCP = af41,
ip defending Priority = 0, ip preemption priority = 0
ip policy locator voice:
ip policy locator video:
UDP checksum = disabled,
session-protocol = sipv2, session-transport = system,
req-qos = best-effort, acc-qos = best-effort,
req-qos video = best-effort, acc-qos video = best-effort,
req-qos audio def bandwidth = 64, req-qos audio max bandwidth = 0,
req-qos video def bandwidth = 384, req-qos video max bandwidth = 0,
dtmf-relay = rtp-nte,
RTP dynamic payload type values: NTE = 101
Cisco: NSE=100, fax=96, fax-ack=97, dtmf=121, fax-relay=122
CAS=123, TTY=119, ClearChan=125, PCM switch over u-law=0,
A-law=8, GSMAMR-NB=117 iLBC=116, AAC-ld=114, iSAC=124
lmr_tone=0, nte_tone=0
h263+=118, h264=119
G726r16 using static payload
G726r24 using static payload
RTP comfort noise payload type = 19
fax rate = disable, payload size = 20 bytes
fax protocol = system
fax-relay ecm disable
Fax Relay ans enabled
Fax Relay SG3-to-G3 Enabled
fax NSF = 0xAD0051 (default)
codec = g711ulaw, payload size = 160 bytes,
video codec = None
voice class codec = `'
voice class sip session refresh system
voice class sip rsvp-fail-policy voice post-alert mandatory keep-alive interval 30
voice class sip rsvp-fail-policy voice post-alert optional keep-alive interval 30
voice class sip rsvp-fail-policy video post-alert mandatory keep-alive interval 30
voice class sip rsvp-fail-policy video post-alert optional keep-alive interval 30
text relay = disabled
Media Setting = forking (disabled) flow-through (global)stats-disconnect (disabled)
Expect factor = 10, Icpif = 20,
Playout Mode is set to adaptive,
Initial 60 ms, Max 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Fax nominal 300 ms
Max Redirects = 1, signaling-type = cas,
VAD = disabled, Poor QOV Trap = disabled,
Source Interface = NONE
voice class sip url = system,
voice class sip tel-config url = system,
voice class sip rel1xx = system,
voice class sip anat = system,
voice class sip outbound-proxy = "system",
voice class sip associate registered-number = system,
voice class sip asserted-id pai,
voice class sip privacy system
voice class sip e911 = system,
voice class sip history-info = system,
voice class sip reset timer expires 183 = system,
voice class sip pass-thru headers = system,
voice class sip pass-thru content unsupp = system,
voice class sip pass-thru content sdp = system,
voice class sip copy-list = system,
voice class sip g729 annexb-all = system,
voice class sip early-offer forced = system,
voice calss sip delay-offer forced = disable,
voice class sip negotiate cisco = system,
voice class sip block 180 = system,
voice class sip block 183 = system,
voice class sip block 181 = system,
voice class sip preloaded-route = system,
voice class sip random-contact = system,
voice class sip random-request-uri validate = system,
voice class sip call-route p-called-party-id = system,
voice class sip call-route history-info = system,
voice class sip call-route url = system,
voice class sip privacy-policy send-always = system,
voice class sip privacy-policy passthru = enable,
voice class sip privacy-policy strip history-info = system,
voice class sip privacy-policy strip diversion = system,
voice class sip send 180 sdp = system,
voice class sip map resp-code 181 = system,
voice class sip bind control = system,
voice class sip bind media = system,
voice class sip bandwidth audio = system,
voice class sip bandwidth video = system,
voice class sip encap clear-channel = system,
voice class sip error-code-override options-keepalive failure = system,
voice class sip error-code-override cac-bandwidth failure = system,
voice class sip calltype-video = false
voice class sip registration passthrough = System
voice class sip authenticate redirecting-number = system,
voice class sip referto-passing = system,
redirect ip2ip = disabled
local peer = false
probe disabled,
Secure RTP: system (use the global setting)
mobility=0, snr=, snr_noan=, snr_delay=0, snr_timeout=0
snr calling-number local=disabled, snr ring-stop=disabled, snr answer-too-soon timer=0
voice class perm tag = `'
Time elapsed since last clearing of voice call statistics never
Connect Time = 85226396, Charged Units = 0,
Successful Calls = 62, Failed Calls = 31, Incomplete Calls = 31
Accepted Calls = 222, Refused Calls = 0,
Bandwidth CAC Accepted Calls = 0, Bandwidth CAC Refused Calls = 0,
Last Disconnect Cause is "10 ",
Last Disconnect Text is "normal call clearing (16)",
Last Setup Time = -2087972307.
Last Disconnect Time = -2087895922.

[Chris Deren]

I believe this is the uptime since router was last rebooted in ms, so 2087895922 would be approximately 579 days, if you issue "show version" you can find out uptime and then you can subtract the last disconnected time from the uptime to see when the last call ended.

[richardmoorehead]

Hi Chris, I did consider that, but the times don't match up:

uptime is 36 weeks, 3 days, 15 hours, 41 minutes
which equals 30,627,200,000,000 milliseconds, roughly

Remember my current logging timestamp was 1377198297.  I'm not sure how these numbers are calculated or if they are even related to logging timestamps.

[Chris Deren]

Simply test would be make a call that matches predictable dial peer and then issue the command to observe what shows up, wait 10-20 seconds and issue it again to see what changes, you should be come to a good conclusion then.

[richardmoorehead]

Hi Chris, good idea - I ran a test using a dial peer that was explicitly set for one DN.

- Incoming dial peer 404 (specific to a user's DN, also has port 5064 applied to the session target IP for unknown reason)

- Outgoing dial peer 312 (one of our commonly used 1-800 dial peers)

Dial-peer 312 timestamp before call: Last Disconnect Time = -2061761934.

Dial-peer 312 timestamp after call: Last Disconnect Time = -2061737261.

DIal-peer 404 timestamp before call: Last Disconnect Time = 2053675545. (note the lack of the minus symbol)

Dial-peer 404 timestamp after call: Last Disconnect Time = -2061752369.

Difference in 312 timestamps: 24673 (little less than half a minute, seems correct enough if MS is being used to calculate)

Difference in 404: Unsure how to calculate.

The numbers originally incremented, but it looks like there must have been a point they hit a buffer and began decrementing into the negatives.  I am totally stumped.  Again, this CUBE hasn't even been online a year.

I believe it's clear that dial-peer 404 isn't being used, but now I am really curious what these numbers actually mean.

[Jason Bree]

This guy had the same curiosity.  It looks like it is counting ms from boot.

http://www.packetpilot.com/what-is-that-timestamp-dial-peers-last-setupdisconnect-time/

I put his calculations in excel and got these formulas:

Last disconnect time    Drop 2 digits    weeks    Days    Hours    Minutes    Sec
1283340034    =LEFT(A5,LEN(A5)-2)    =B5/60/60/24/7    =(C5-INT(C5))*7    =(D5-INT(D5))*24    =(E5-INT(E5))*60    =(F5-INT(F5))*60

Jason

[richardmoorehead]

Jason, I reloaded the routers on Friday night and checking timestamps seems to show your findings are accurate.

I'm still not sure at what point the dial-peers hit a decrement number, but I'll keep checking every week or so until I notice it.