More ShoreTel Troublehsooting

The TMSNCC logs the call accumulations recorded at the end of a call in the G-MST entry, this points to the RTP stream of the call with the following summary

Packet info
s: sent packets
r: received packets
l: lost packets

Jitter Buffer Info (call quality)
j: # of times jitter buffer had to adjust size trying to handle voice streams
u: # of times the jitter buffer did not get enough data to pass on a voice stream

o: # of times the jitter buffer received data outside of its maximum jitter buffer

09:11:49.557 ( 4184: 5288) G-MST: 200000BD "00070001-33cd-5927-c6e1-005056a27011" ("",""),(0, 0),0(Null),rsn:1,22:06:23.979 (UTC),pl:20,(s:16279, r:16279, l:0),(j:3,u:6435,o:11) flgs:0x00000000 "sip:@:5441",vpn:0

G-MST: 2 --> Relates to Network Side of the call
G-MST: 4 --> Relates to Trunk Side of the call


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