One Access

Business-Class Router Solutions All Telecommunication Services in One Access
4.10
Voice Call Routing

VRT (Voice Routing Table)
www.oneaccess-net.com
contact@oneaccess-net.com
v0.11 2007
Copyright OneAccess Networks All rights reserved
VoIP: Call routing
Overview
Applicable to any call between any type of ports (local or voip).

Any call routing is possible except VoIP -> VoIP call
Implicit Routing
Easy to configure : no called number configuration

No number processing
Routing Table
Incoming call routing : optional process done for each incoming call (from
VoIP network or local port) allowing number translations (calling & called
numbers) and voice profile selection
Outgoing call routing : for numbering plan definition (overlap or in-bloc),
selection of the remote voip device if no gatekeeper, number translations
(calling & called)
Possibility to configure a backup route in case of outgoing call failure
Possibility to modify specific ISDN Information Elements :
Type of number, Numbering plan, Bearer capability
Call Routing Architecture (from local port)
Incoming call routing

(number translations)
Outgoing call routing
PSTN Interfaces
(ISDN, analog)
VoIP Network
(SDP media properties)
Call Routing Architecture (from VoIP network)
Outgoing call routing
Incoming call routing

(select voip profile,
number translations)
PSTN Interfaces
(ISDN, analog)
VoIP Network
(SDP media properties)
VoIP: Call Routing
Up to 200 routing rules can be configured
Type of rule: incoming (initial routing when receiving the call) or

outgoing (2nd step of routing)
Number to be checked
Called or calling number
Wildcards can be used
Related port (local or VoIP)

Actions: Number translations, ISDN I.E. forcing
Definition of backup routes
The rule order is important: the rule #1 is checked

first
Insertion & deletion of rules by using the rule index
VoIP: Call Routing
Routing Process for call coming from a local port
1. The incoming rules related to the local port are checked down the
rule list
If matching the rule (number & port), the number translations are
applied
The check of incoming routing rules can be processed again until the
last one. This possibility allows to make several number translations on
the calling and called number.
2. The outgoing rules are checked from the first one

If fully matching the rule, the number translations of the rule are applied
and the call is routed to the specified port (local or voip).
Backup route can be configured to define alternative rules in case
of failure.
If partially matching the rule (e.g.: incomplete number in overlap mode),
the call is not further routed and the GW waits for another digit
VoIP: Call Routing
Routing Process for call coming from VoIP
1. The incoming rules related to voip are checked from the

first one
Upon a match (number only), the number translations are applied

The voip port (coder profile,etc) is determined
Check of routing table can be processed again until the end of the
table
2. The outgoing rules are checked from the first one
Outgoing rules with voip port are excluded (no voip-voip routing)
Upon a full match, the number translations are applied and the call
is routed to the specified port (local or voip).
Backup can be configured to define alternative rules in case of failure.
Upon a partial match (ex : uncomplete number in overlap mode), the

routing process is fully cancelled (done again with the next digit).
VoIP: Call Routing
Routing on local port
Configuration of a port group

Same number for several ports
Incoming Calls are distributed with a configurable priority
Possibility of call hunting
automatic backup inside the port group

higher priority is given to working ports
Several numbers can be configured for each port
Voice Routing (1)
6. Voice routing
one200(configure)>voice-routing
one200(configure)>voice-routing
one200(voice-route)>
one200(voice-route)> ??
display-routes
-- Show
display-routes
Show voice
voice routing
routing table
table
exit
-- Exit
exit
Exit from
from command
command node
node
insert
Insert
VOIP
voice
route
insert
- Insert VOIP voice route
move
move
- Move
Move VOIP
VOIP voice
voice route
route
no
no
no
- no
route
-- Set
route
Set VOIP
VOIP voice
voice route
route
test-route
Test
voice
routing
test-route
- Test voice routing table
table
<cr>
<cr>
Voice Routing (2)
one200(voice-route)>route
one200(voice-route)>route 11
one200(conf-voice-route)>
one200(conf-voice-route)> ??
dial-peer
-- Set
dial-peer
Set route
route dial
dial peer
peer
exit
Exit
from
command
exit
- Exit from command node
node
force-bearer-cap
Set
force
bearer
capability
force-bearer-cap
- Set force bearer capability fields
fields
force-clir
Set
caller
line
identity
request
force-clir
- Set caller line identity request
force-numplan
force-numplan
- Set
Set origin/destination
origin/destination numplan
numplan
insert-calling
Set
route
calling
insertion
insert-calling
- Set route calling insertion
insert-prefix
insert-prefix
- Set
Set route
route prefix
prefix insertion
insertion
insert-suffix
Set
route
suffix
insertion
insert-suffix
- Set route suffix insertion
no
no
- no
no
prefix
Set
prefix
- Set route
route prefix(prefix
prefix(prefix [number-type][length][timer]
[number-type][length][timer]
[overlap])
[overlap])
prefix-type
Set
prefix-type
- Set route
route direction
direction and
and type
type of
of call
call
startup-file
Set
restart
equipment
with
the
file
startup-file
- Set restart equipment with the file name.
name.
suppress-prefix
Set
route
prefix
suppression
suppress-prefix
- Set route prefix suppression
translate
translate
- Set
Set route
route number
number translate
translate
wildcard
Set
wildcard
value
wildcard
- Set wildcard value for
for prefix
prefix
<cr>
<cr>
10
Voice Routing (3)
Registration to registrar server (user=phone).
E164 Prefix is mapped to a pots-group or a voip resource. To register it,

<ua-sip> must be added. Corresponding prefix is used as user portion of
sip uri of To/From/Contact header field
To register a username instead of a E164 prefix (no user=phone), sipusername must be validated. It overrides prefix value for
To/From/Contact header fields for register method. It overrides prefix
value for From/Contact header fields for Invite method.
Registrar ip address or domain name must be configured at the sipgateway (reg-dns-add)
401 or 407 challenge for register and/or invite method. Sipauthentication <username> <password> are provided to calculate
MD5 response.
11
Voice Routing (4)
authentication
CLI(voice-route)>route 1
CLI(voice-route)>dial-peer pots-group 0 ua-sip
CLI(voice-route)>sip-username <user portion of sip uri>
CLI(voice-route)>sip-authentication [<username> <password>]
CLI(voice-route)>prefix 1000 length 4
CLI(voice-route)>prefix-type outgoing called last
CLI(voice-route)>exit
CLI(configure)>exit
12
Voice Routing (5)
one200(voice-route)>display-routes
1 - 20. all length 4 / voip 1 / mixed
- called - next
2 -1000 all length 4 / pots 0 / outgoing - called
- last
one200>show running-config
...
voice-routing
route 1
dial-peer voip 1
prefix 20. length 4
prefix-type mixed called next
exit
route 2
dial-peer pots-group 1 ua-sip
prefix 1000 length 4
prefix-type outgoing called last
exit
exit
...
13
Voice Routing (6)
Prefix suppression
1234
Pots-group 0
12011234
VoIP 0
route 1
dial-peer pots-group 0
prefix 120A. length 4
suppress-prefix 4 called
wildcard A 0123
exit
14
Voice Routing (7)
Codec Profile for incoming call
Pots-group 0
VoIP 1
Incoming
Table
986902739
VoIP 2
986911111
Pots-group 4
route 14
dial-peer voip 1
prefix-type incoming called
exit
route 15
dial-peer voip 2
prefix . length 0
prefix-type incoming called
exit
15
Voice Routing (8)
Modification of Calling and Called numbers 1/4
Calling : 019869027391111
Called : 12345678
Calling : 019869027352222
Called : 12345678
Incoming
Table
Calling : 986902739
Called : 120012345678
Outgoing
Table
Calling : 986902735
Called : 12345678
Pots-group 0
16
Voice Routing (9)

route 2
dial-peer pots-group 0
translate .98690273A. 98690273A calling
prefix . length 0
prefix-type incoming calling next
exit
route 3
dial-peer pots-group 0 alias
insert-prefix 1200 called
prefix-type incoming calling
exit
alias added to register the number 1200 in the RAS
17
Voice Routing (10)

Calling : 986902739
Called : 120012345678
Calling : 019869027391111
Called : 12345678
Incoming
Table
Calling : 986902739
Called : 120012345678
Outgoing
Table
Voip 1
Calling : 019869027352222
Called : 12345678
Calling : 986902735
Called : 12345678
Voip 0
Pots-group 0
Calling : 986902735
Called : 12345678
18
Voice Routing (11)

route 4
dial-peer voip 1
prefix 1200. length 4
exit
route 5
dial-peer voip 0
prefix . length 0
exit
19
Business-Class Router Solutions All Telecommunication Services in One Access
Voice Toubleshooting & Statistics
www.oneaccess-net.com
contact@oneaccess-net.com
v0.11 2007
Voice routing table testing

One_Training(voice-route)>display-route
1 - . all length 10 / pots 0 / incoming - called - last
/ +prefix-calling 0
10 - . all timer / voip 0 / outgoing - called - last
20 - . all length 9 / pots 0 / outgoing - called - last
/ -prefix-called 1
One_Training(voice-route)>
One_Training> configure terminal
One_Training(configure)>voice-routing
One_Training(voice-route)>test-route 0141877422 260402020 from-voip 0
--> test_route : Calling=0141877422 Called=260402020
Route match : Calling=0141877422 Called=60402020
Incoming Routes =
Outgoing Routes = 20,
Incoming call from voip id:0 Send towards Isdn on local port: 5/0
One_Training(voice-route)>test-route 260402020 0141877422 from-pots 0
--> test_route : Calling=260402020 Called=0141877422
Route match : Calling=0260402020 Called=0141877422
Incoming Routes = 1,
Outgoing Routes = 10,
Outgoing call from local port: 5/0 -> Send towards H323 on voip id:0
21
VoIP Statistics
BRI statistics
CLI#
CLI# show
show voice
voice voice-port
voice-port bri
bri index
index 00
voice port
voice port
protocol descriptor
protocol descriptor
current state
current state
config state
config state
layer 1 status
layer 1 status
attached vmoabri dial peer
attached vmoabri dial peer
number of voice communication
number of voice communication
bri Tx frames on D channel
bri Tx frames on D channel
bri Rx frames on D channel
bri Rx frames on D channel
5/0
5/0
BRI_NT
BRI_NT
activated
activated
up
up
activated
activated
0
0
0
0
40
40
41
41
Outgoing calls
Outgoing calls
Outgoing calls failures
Physical Interface down
Cause Class 0 (normal event)
Normal Cause (16)
: 2
Normal Cause (16)
: 2
User busy (17)
: 3
User busy (17)
: 3
No answer (18)
: 0
No answer (18)
: 0
Cause Class 2 (unavailable ressources)
Cause Class 3 (unavailable service)
Cause Class 4 (service not provided)
Cause Class 5 (invalid message)
Cause Class 6 (protocol error)
Cause Class 7 (interworking)
: 102
: 102
: 5
: 5
: 0
: 0
: 0
: 0
: 5
: 5
Incoming calls
Incoming calls
Incoming calls failures
Remote failure
Remote failure
Unknown number
Unknown number
DSP unavailable
DSP unavailable
Not specified
Not specified
: 54
: 54
: 7
: 7
: 0
: 0
: 5
: 5
: 0
: 0
: 2
: 2
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
22
VoIP Statistics
PRI statistics
CLI#
CLI# show
show voice
voice voice-port
voice-port pri
pri index
index 00
voice port
voice port
physical type
physical type
protocol descriptor
protocol descriptor
current state
current state
config state
config state
layer 1 status
layer 1 status
number of voice communications
number of voice communications
pri AIS occurence
pri AIS occurence
pri RDI occurence
pri RDI occurence
5/0
5/0
E1
E1
E1_PRI
E1_PRI
activated
activated
up
up
deactivated
deactivated
0
0
0
0
0
0
Outgoing calls
Outgoing calls
Normal Cause (16)
: 0
Normal Cause (16)
: 0
User busy (17)
: 3
User busy (17)
: 3
No answer (18)
: 0
No answer (18)
: 0
: 67
: 67
: 3
: 3
: 0
: 0
: 0
: 0
: 3
: 3
Incoming calls
Incoming calls
Remote failure
Remote failure
Unknown number
Unknown number
DSP unavailable
DSP unavailable
Not specified
Not specified
: 23
: 23
: 2
: 2
: 2
: 2
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
: 0
23
VoIP Statistics
FXS statistics
CLI#
CLI# show
show voice
voice voice-port
voice-port fxs
fxs index
index 00
voice port
voice port
current
current state
state
config
config state
state
attached
attached vmoa
vmoa fxs
fxs dial
dial peer
peer
voice
voice communication
communication
Outgoing
Outgoing calls
calls
User
User busy
busy
No
No answer
answer
Incoming
Incoming calls
calls
Incoming
calls
failures
Remote
Remote failure
failure
Unknown
Unknown number
number
DSP unavailable
DSP unavailable
Not
Not specified
specified
5/0
5/0
on
on hook
hook
up
up
00
no
no
:: 32
32
: 3
: 3
:: 22
:: 11
:: 66
:: 00
:: 00
:: 00
: 0
: 0
:: 00
24
VoIP Statistics
Dial-peer VoIP Statistics (1)
one200>
one200> show
show voice
voice dial-peer
dial-peer voice
voice voip
voip type
type index
index <port
<port id>
id> [reset]
[reset]
or
or
one200>
one200> show
show voice
voice dial-peer
dial-peer voice
voice voip
voip type
type global[reset]
global[reset]
type
type may
may be
be ::
--current : statistics on current calls
current : statistics on current calls
--outgoing : outgoing calls only
outgoing : outgoing calls only
--incoming : incoming calls only
incoming : incoming calls only
--user-plan : voice & fax only
user-plan : voice & fax only
--all (default) : all the statistics are provided
all (default) : all the statistics are provided
25
VoIP Statistics
Dial-peer VoIP Statistics (2): Outgoing Calls

Dial Peer
Current Calls
1
0
Outgoing Calls
Outgoing Calls
RAS Call Failures
Gatekeeper Unavailable
Admission Rejects
H225/Q931 Call failures
Normal Cause (16)
User busy (17)
No answer (18)
H245 Call failures
Incompatible capabilities
Protocol errors
Internal call failures
DSP unavailable
Max-bandwidth exceeded
Max-connection exceeded
Not specified
4
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
26
VoIP Statistics
Dial-peer VoIP Statistics (3): Incoming Calls

Incoming calls
RAS Call failures
Gatekeeper Unavailable
Admission Rejects
Local Port Call failures
H245 Call failures
Incompatible capabilities
Protocol errors
Internal call failures
DSP unavailable
Unknown number
Channel / port unavailable
Max-bandwidth exceeded
Max-connection exceeded
Not specified
3
1
0
0
0
1
0
0
0
0
0
0
0
0
0
0
27
VoIP Statistics
Dial-peer VoIP Statistics (4): Voice and Fax
RTP statistics
Number of transmitted packets
Number of received packets
Number of transmitted bytes
Number of received bytes
Number of excessive jitter events
Number of lost packets
Number of invalid packets
Number of calls with frame error rate
total
<0.01<0.1<0.5<1<5>=5
3
0
0
2
1
Modem passthrough
Number of switching to modem mode
T38 FAX Calls
Number of outgoing fax
Number of incoming fax
Number of failures
Request Mode failure
Pre-message procedure failure
Page failure
Number of transmitted packets
Number of received packets
Number of transmitted bytes
Number of received bytes
Number of lost packets
1237
1234
101484
101098
3
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
0
28
VoIP Statistics
Events
vxTarget>event
vxTarget>event
filter
-- Add/remove
filter
Add/remove events
events filters
filters
manager
Add
a
SNMP
manager
manager
- Add a SNMP manager
no
no
- No
No
recover
Recover
recover
- Recover events
events from
from memory
memory
vxTarget>event
filter
vxTarget>event filter
add
-- Add
add
Add an
an event
event filter
filter
remove
Remove
a
events
remove
- Remove a events filter
filter from
from the
the table
table
vxTarget>event
filter
add
vox
vxTarget>event filter add vox
ALL
-- All
ALL
All families
families from
from vox
vox group
group
GEN
GEN
GEN
- GEN
VOATM
VOATM
- VOATM
VOATM
VOIP
VOIP
VOIP
- VOIP
vxTarget>event
filter
add
vox voip
voip
<subfam>
<ALL
|
<subfam>
- <ALL | ControlPlan
ControlPlan || UserPlan>
UserPlan>
<fam2>
<GEN
|
VOATM>
<fam2>
- <GEN | VOATM>
vxTarget>event
filter
add
vox voip
voip all
all show
show
29
VoIP Statistics
Voice call history, active calls
Gives statistics on the current voice calls and the last 100 calls
vxTarget>show
vxTarget>show voice
voice voip-call
voip-call any
any ind
ind 11
11 -- Call
Call from
from remote
remote voip:
voip: 0,
0, to
to local
local port:
port: 5/1
5/1 call-id:
call-id: 44 active
active
calling
:
110,
called
:
111
calling : 110, called : 111
setup
setup time:
time: 01/02/00
01/02/00 04h58m31s
04h58m31s
01/02/00
01/02/00 04h58m31s
04h58m31s
RTP
Source
RTP Source ip
ip :192.168.1.110
:192.168.1.110 rtp:16384
rtp:16384 /Dest
/Dest ip
ip :192.168.1.111
:192.168.1.111 rtp:16386
rtp:16386 (active)
(active)
Play
time
(voice)
:
00h00m39s
Play time (voice) : 00h00m39s
Tx
Tx Coder
Coder :: G729
G729 // 20
20 ms
ms ;; Rx
Rx Coder
Coder :: G729
G729
RTP
Packets
RX
/
TX
:
1988
/
1989
RTP Packets RX / TX : 1988 / 1989
RTP
RTP Packet
Packet lost&discarded
lost&discarded RX
RX // TX
TX (RTCP
(RTCP reported)
reported) :: 00 // 395307
395307
Number
of
Excessive
Jitter
events
:
3
Number of Excessive Jitter events : 3
30
VoIP Statistics
RTP sessions history
Gives complete statistics about the 200 last RTP sessions
CLI>
CLI> show
show voice
voice rtpcall
rtpcall full
full any
any ind
ind 22
22 -- 01/04/01
01/04/01 00h47m24s
00h47m24s RTP
RTP 192.168.1.1:16384
192.168.1.1:16384 192.168.1.111:16386
192.168.1.111:16386
Play
time
(voice)
:
00h00m46s
Play time (voice) : 00h00m46s
Tx
Tx Coder
Coder :: G729
G729 // 20
20 ms
ms ;; Rx
Rx Coder
Coder :: G729
G729
VAD
enabled
local
/
remote
:
no
/
no
VAD enabled local / remote : no / no
ERL
ERL :: 15
15 dB
dB
ACOM
:
32
dB
ACOM : 32 dB
RTP
RTP Packets
Packets received
received (DSP
(DSP // Uplink)
Uplink) :: 2337
2337 // 2337
2337
lost
:
0
lost : 0
out
out of
of sequence
sequence :: 00
invalid
invalid :: 00
RTP
RTP Packets
Packets transmitted
transmitted (DSP
(DSP // Uplink)
Uplink) :: 2338
2338 // 2338
2338
lost
(RTCP
reported)
:
0
lost (RTCP reported) : 0
Jitter
Jitter parameter
parameter :: 100
100 ms
ms
Number
of
Excessive
Jitter
Number of Excessive Jitter events
events :: 11
31
VoIP Statistics
RTP sessions history (continue)

Excessive
Excessive Jitter
Jitter events
events ::
2|
2|
1|
1| **
------------------------------------------------------------------------------00 30"
30" 1'
1' 2'
2' 4'
4' 8'
8' 12'
12' >16'
>16'
Jitter
Jitter received
received (uplink)
(uplink) ::
Max
Max delay
delay :: 93
93 ms
ms
Delays
(ms)
>50
Delays (ms)
>50 >100
>100 >150
>150 >200
>200 >300
>300
Nb
22
00
00
00
00
Nb of
of occur.
occur.
Interarrival
Interarrival max
max jitter
jitter :: 99 ms
ms
Jitter
received
(DSP)
:
Jitter received (DSP) :
Max
Max delay
delay :: 93
93 ms
ms
Delays
(ms)
>50
Delays (ms)
>50 >100
>100 >150
>150 >200
>200 >300
>300
Nb
22
00
00
00
00
Nb of
of occur.
occur.
Interarrival
Interarrival max
max jitter
jitter :: 99 ms
ms
Frames
Frames with
with aa delay
delay >50
>50 ms
ms ::
2|
2|
1|
**
1| **
------------------------------------------------------------------------------00 30"
30" 1'
1' 2'
2' 4'
4' 8'
8' 12'
12' >16'
>16'
Jitter
Jitter transmitted
transmitted (uplink)
(uplink) ::
Max
Max delay
delay :: 66 ms
ms
Delays
(ms)
>50
Delays (ms)
>50 >100
>100 >150
>150 >200
>200 >300
>300
Nb
of
occur.
0
0
0
0
00
Nb of occur.
0
0
0
0
Interarrival
Interarrival max
max jitter
jitter :: 11 ms
ms
(RTCP
reported)
:
2
ms
(RTCP reported) : 2 ms
32
VoIP: Internal Call Generator
Possibility to generate and / or terminate one or

several VoIP calls
Two services: RTP loopback or BERT testing
Use of virtual and routable dial-peer pots
dial-peer
dial-peer voice
voice pots
pots 44
service
service bert2047
bert2047 both
both 33
pots-group
pots-group 00
exit
exit
voip-call
voip-call 11
pots
pots 44
called
called 0198723000
0198723000
calling
calling 3000
3000
bearer
data
bearer data
duration
duration 180
180
timeout
10
timeout 10
exit
exit
One200>
One200> start
start identifier
identifier 11
33
VoIP: ISDN capture
Possibility to capture the signalling traffic over

ISDN BRI & PRI interfaces: layer 1 to 3
For VoIP side: use of the IP capture possibilities
CLI>conf t
CLI>logging buffered debug
CLI>exit
CLI>debug isdn all layer 1to3
00:07:21.271
00:07:21.271
00:07:21.271
00:07:21.271
00:07:21.271
00:07:21.272
00:07:21.343
00:07:21.343
00:07:21.343
00:07:21.343
00:07:21.343
00:07:21.343
line:5/0 L1 frame sent.

line:5/0 L2 tx UI P/F=0 NR=4 NS=2 C/R=1.
hex: 02 ff 03
line:5/0 L3 tx SETUP callref:8.
hex1: 08 01 08 05 04 03 80 90 a3 18 01 89 70 04 a1 31
hex2: 31 31 a1
line:5/0 L1 frame received.
line:5/0 L2 rx SABME P/F=1 C/R=0.
hex: 00 01 7f
line:5/0 L1 frame sent.
line:5/0 L2 tx UA P/F=1 NR=4 NS=2 C/R=0.
hex: 00 01 73
34
Call factory over IP
For debug, a SETUP can be sent on VoIP.

One_training>auto-call
<called>
- called number: up to 21 characters <0..9, #, *>
One_training>auto-call 0141877422
<calling>
- calling number: up to 21 characters <0..9, #, *>
<pots-number>
- pots: 0..29
<bearer>
- bearer capability < voice | data | voiceband >
overlap
- units in milliseconds: 0..2000 <0 means no overlap used>
<cr>
One_training>auto-call 0141877422 2408882005
17:50:17.677 Info vox factory test 1 call-id: 4, ident: auto-call, CALL IN PROGRESS Calling=2408882
005 Called=0141877422.
one100_interopBW>17:50:17.678 Info vox voip controlplan 3 Incoming call on local pots: 0, calling:
2408882005, called: 0141877422, call-id: 4.
17:50:17.710 Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: 2408882005, called:
0141877422, call-id: 4.
17:50:27.660 Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED cause=no codec.
17:50:27.661 Info vox factory test 1 call-id: 4, ident: auto-call, CALL FAILED on pots cause=[Norma
l call clearing].
35
Auto call to ISDN
For debug, a SETUP can be sent on a ISDN local

port. One_training>isdn test call 85841 ( data call/unrestricted )
02:27:34.914 line:5/0 L1 event received PH_AR State:F3.
02:27:34.923 line:5/0 L1 event received EV_LOST_FRAMING State:F4.
02:27:34.923 line:5/0 L1 event received EV_INFO_2 State:F5.
02:27:34.925 line:5/0 L1 event received EV_INFO_4_8(PH_AI) State:F6.
02:27:34.925 line:5/0 L1 event received MPH_AI State:F7.
02:27:34.925 line:5/0 L1 frame sent.
02:27:34.925 line:5/0 L2 tx SABME P/F=1 C/R=0.
02:27:34.925
hex: 00 83 7f
02:27:34.938 line:5/0 L1 frame received.
02:27:34.938 line:5/0 L2 rx UA P/F=1 C/R=0.
02:27:34.939
hex: 00 83 73
02:27:34.939 line:5/0 L1 frame sent.
02:27:34.940 line:5/0 L2 tx INFO P=0 NR=0 NS=0 C/R=0.
02:27:34.940
hex: 00 83 00 00
02:27:34.940 line:5/0 L3 tx SETUP callref:4.
02:27:34.940
hex1: 08 01 04 05 04 02 88 90 18 01 83 70 06 81 38 35
02:27:34.940
hex2: 38 34 31 a1
02:27:34.940
Called Number : 85841
36
Fax relay T38 debug
Fax T38 processing and traces: display of the T30

messages.
One_training>trace filter add vox up ifp 2 show
01:42:38.100 Info vox voip controlplan 3 Outgoing call on voip id: 0, calling: 110,
called: 111, call-id: 29.
01:42:40.397 Info vox voip controlplan 3 Alert in band received, call-id: 29.
01:42:40.917 Info vox voip userplan 3 VoIP RTP transmission started, coder: G729, call-id: 29.
01:42:40.997 Info vox voip userplan 3 VoIP RTP reception started, coder: G729, call-id: 29.
01:42:52.444 Info vox voip controlplan 3 Call connected, call-id: 29.
01:42:57.537 Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G729, call-id: 29.
01:42:57.537 Info vox voip userplan 3 VoIP RTP reception stopped, coder: G729, call-id: 29.
01:42:57.665 Info vox voip userplan 3 Fax T38 starting call-id: 29 .
01:43:04.825 VOX.up.ifp.2 ------ NSF / CSI / DIS ------>
01:43:07.717 VOX.up.ifp.2 <-------- TSI / DCS ---------01:43:07.837 VOX.up.ifp.2 <------- TCF (v29_9600) -------01:43:11.814 Info vox voip userplan 1 T38 Pre-message procedure OK, call-id: 29.
01:43:11.814 VOX.up.ifp.2 ------------ CFR ------------>
01:43:12.437 Info vox voip userplan 1 T38 Transmitting page 1, call-id: 29.
01:43:12.437 VOX.up.ifp.2 <------ PAGE (v29_9600) -------01:51:10.502 VOX.up.ifp.2 <--------- PPS-EOP ----------01:51:13.599 Info vox voip userplan 1 T38 page 1 OK, call-id: 29.
01:51:13.599 VOX.up.ifp.2 ------------ MCF ------------>
01:51:15.282 VOX.up.ifp.2 <----------- DCN ------------01:51:15.846 Info vox voip controlplan 3 Call Disconnection received on local port: 5/2,
cause: (16)[Normal call clearing], call-id: 29.
37
Modem / fax pass-through event
Example of event in case of modem/fax call.
One_training>event filter add vox all show

00:11:21.485 Info vox voip controlplan 3 Incoming call on voip id: 0, calling: 111,
called: 110, call-id: 3.
00:11:21.486 Info vox voip controlplan 3 Outgoing call on local port: 5/0, calling: 111,
called: 110, call-id:3.
00:11:21.550 Info vox voip controlplan 3 Alert received, call-id: 3.
00:11:21.631 Info vox voip controlplan 3 Call connected, call-id: 3.
00:11:25.142 Info vox voip userplan 1 Fax/Modem Passthrough starting call-id: 3.
00:11:25.142 Info vox voip userplan 3 RTP new transmission coder: G711 A Law, call-id: 3.
00:11:25.341 Info vox voip userplan 3 RTP new reception coder: G711 A Law, call-id: 3.
00:12:23.533 Info vox voip controlplan 3 Call Disconnection received on local port: 5/1,
cause: (16)[Normal call clearing], call-id: 3.
00:12:23.552 Info vox voip userplan 3 VoIP RTP transmission stopped, coder: G711 A Law, call-id: 3.
00:12:23.552 Info vox voip userplan 3 VoIP RTP reception stopped, coder: G711 A Law, call-id: 3.
38

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Business-Class Router Solutions All Telecommunication Services in One Access

Voice Call Routing

VoIP: Call routing

Applicable to any call between any type of ports (local or voip).

Easy to configure : no called number configuration

Call Routing Architecture (from local port)

Incoming call routing

Outgoing call routing

Copyright OneAccess Networks All rights reserved

Call Routing Architecture (from VoIP network)

Outgoing call routing

Incoming call routing

Copyright OneAccess Networks All rights reserved

VoIP: Call Routing

Up to 200 routing rules can be configured

Type of rule: incoming (initial routing when receiving the call) or

Related port (local or VoIP)

The rule order is important: the rule #1 is checked

Insertion & deletion of rules by using the rule index

Copyright OneAccess Networks All rights reserved

VoIP: Call Routing

Routing Process for call coming from a local port

2. The outgoing rules are checked from the first one

Copyright OneAccess Networks All rights reserved

VoIP: Call Routing

Routing Process for call coming from VoIP

1. The incoming rules related to voip are checked from the

Upon a match (number only), the number translations are applied

2. The outgoing rules are checked from the first one

Backup can be configured to define alternative rules in case of failure.

Upon a partial match (ex : uncomplete number in overlap mode), the

Copyright OneAccess Networks All rights reserved

VoIP: Call Routing

Routing on local port

Configuration of a port group

automatic backup inside the port group

Several numbers can be configured for each port

Copyright OneAccess Networks All rights reserved

Voice Routing (1)

Copyright OneAccess Networks All rights reserved

Voice Routing (2)

Copyright OneAccess Networks All rights reserved

Voice Routing (3)

Registration to registrar server (user=phone).

E164 Prefix is mapped to a pots-group or a voip resource. To register it,

Registrar ip address or domain name must be configured at the sipgateway (reg-dns-add)

Copyright OneAccess Networks All rights reserved

Voice Routing (4)

Copyright OneAccess Networks All rights reserved

Voice Routing (5)

Voice Routing (6)

Copyright OneAccess Networks All rights reserved

Voice Routing (7)

Codec Profile for incoming call

Copyright OneAccess Networks All rights reserved

Voice Routing (8)

Modification of Calling and Called numbers 1/4

Copyright OneAccess Networks All rights reserved

Voice Routing (9)

Modification of Calling and Called numbers 2/4

Voice Routing (10)

Modification of Calling and Called numbers 3/4

Copyright OneAccess Networks All rights reserved

Voice Routing (11)