SIGTRAN = SS7 information over IP
QSIG = Used to communicate to PBX. Based on q.931
Basic items needed to set up t1 CAS
#network-clock-participate wic 1
#controller t1 0/0/1
#framing esf
#linecode b8zs
#clock source line
#ds0-group 0 timeslots 1-12 type e&m-wink-start
Basic items needed to set up t1 CCS
isdn switch-type primary-5ess
controller t1 0/0/1
pri-group timeslots 1-4 (service nfas-d|mgcp)-multiple pri’s and 1 d chann for CCM to use the pri
transparent ccs – pass any d channel proprietary information
ds0-group 23 timeslot 24 ext-sig
Order for matching incoming dial-peer
Incoming called-number . (DNIS)* most common
answer-address (ANI)
destination-pattern (ANI)
port assignment (pots only) port the call came in on
pots dial peers strip all matched digits
example 41..called 4100 will send 00 only
use to solve:
prefix 41
forward-digits 4
no digit-strip
default failover between dial peers is 15 seconds h225 is responsible
To change this create:
voice class h323 10
h225 timeout tcp establish 5
apply to dial-peer
voice-class h323 10
g729 is more complex than g729a (less CPU usage)
connection tieline emulates a temporary connection to a pbx.
QoS
one way delay is 150 – 200 ms
jitter needs to be less than 30 ms
less than 1% loss
Local Voice Busyout LVBO
voice port 1/0/0:0
busyout monitor serial 0/0/1
Will show ports as busy on voice port if serial link goes down. CAC
Measurement based CAC
AVBO
Under voice port – busyout monitor probe 192.168.1.1 icpif 20 (g.711 default)
RSVP
Under dial-peer
req-qos guaranteed-delay
acc-qos guaranteed-delay
under interface
ip rsvp bandwidth 500(total) 30(Per call)
QoS
Priority queuing polices also(strict priority). If you give it 256k that is guaranteed but no more!
**** show policy-map interface Ethernet 0******
See how many packets in or out of an interface. See 5 minute rate also
/5\(…\)/ /65125\1/ 5XXX —> 65125XXX
Troubleshoot h.323
VTSP
debug vtsp dsp You can see the DTMF digits the DSP is seeing.
Can see jitter buffers setup, playing of dial tone. Can see if VAD is on.
CCAPI
debug voip ccapi inout
What dial peer was matched?
show dialplan number 7028473
Will show what dial peer a number will use rather than making a test call.
debug cch323 h225 | h245
debug ip tcp transitions
Debug RAS
Gatekeeper information.
h.245 codec bitmask
0×1 g711u
0×2 g711a
0×4 g729ietf
0×8 g729a
0×10 g726r16
0×20 g726r24
0×40 g726r32
0×80 g728
0×100 g723r63
0×200 g723r53
0×800 g729b
0×1000 g729ab
0×2000 g723ar63
0×4000 g723ar53
0×8000 g729
All Codecs = 0×4eff7 (all bits set) Usually sent by requesting gateway
VAD_off 0×1
VAD_on 0×2
FAX Cap bits
0×1 None
0×2 Voice
0×4 14400
0×8 9600
0×10 7200
0×20 4800
0×40 2400
0×80 1200
http://www.cisco.com/warp/public/129/isdn_disc_code.html
Cause Code Origination Point
The first byte (most significant) after 0x indicates the point in the circuit path where the disconnect cause code appears. Consider the sample output in the Introduction section. 82 indicates that the call disconnects from the local Telco switch. Here is a list of cause code origination points that help you interpret where the call disconnects from:
80—the router
81—the private network near the local user (possibly a local PBX)
82—the public network near the local user (local Telco switch)
83—the transit network (in the ISDN cloud)
84—the public network near the remote user (remote Telco switch)
85—the private the network near the remote user (possibly a remote PBX)
87—the international network
8A—a network beyond the internetworking point
