[highlight] - This is how I connected a Definity Prologix to Asterisk using a H.323 ip trunk:[/highlight]
Dialplan:
Definity extension range: 1000 to 2699
Asterisk extension range: 2700 to 2799
Network IP addresses:
Definity (C-LAN) : 192.168.1.50
Asterisk (ETH0) : 192.168.1.51
Definity (IP trunk - page 1):
[blue][tt]
display trunk-group 84 Page 1 of 10
TRUNK GROUP
Group Number: 84 Group Type: isdn CDR Reports: y
Group Name: Asterisk COR: 95 TN: 1 TAC: *11
Direction: two-way Outgoing Display? y Carrier Medium: IP
Dial Access? y Busy Threshold: 99 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n TestCall ITC: unre
Far End Test Line No:
TestCall BCC: 4
TRUNK PARAMETERS
Codeset to Send Display: 6 Codeset to Send National IEs: 6
Max Message Size to Send: 260 Charge Advice: none
Supplementary Service Protocol: b Digit Handling (in/out): enbloc/enbloc
Trunk Hunt: cyclical QSIG Value-Added? y
Digital Loss Group: 13
Incoming Calling Number - Delete: Insert: Format:
Bit Rate: 19200 Synchronization: async Duplex: full
Disconnect Supervision - In? y Out? y
Answer Supervision Timeout: 0
[/tt][/blue]
Definity (IP trunk - page 2):
[blue][tt]
display trunk-group 84 Page 2 of 10
TRUNK FEATURES
ACA Assignment? n Measured: none Wideband Support? n
Internal Alert? n Maintenance Tests? n
Data Restriction? n NCA-TSC Trunk Member:
Send Name: y Send Calling Number: y
Used for DCS? n Hop Dgt? n
Suppress # Outpulsing? n Format: public
Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Send Called/Busy/Connected Number: y
Modify Tandem Calling Number? n
Send UUI IE? y
Send UCID? n
Send Codeset 6/7 LAI IE? y
Path Replacement with Retention? n
Path Replacement Method: better-route
SBS? n Network (Japan) Needs Connect Before Disconnect? n
[/tt][/blue]
Definity (IP trunk - page 4):
[blue][tt]
display trunk-group 84 Page 4 of 10
TRUNK GROUP
Administered Members (min/max): 1/4
GROUP MEMBER ASSIGNMENTS Total Administered Members: 4
Port Code Sfx Name Night Sig Grp
1: T00208 83
2: T00230 83
3: T00283 83
4: T00284 83
5:
6:
7:
8:
9:
10:
11:
12:
13:
14:
15:
[/tt][/blue]
Definity (Signaling Group):
[green][tt]
display signaling-group 83
SIGNALING GROUP
Group Number: 83 Group Type: h.323
Remote Office? n Max number of NCA TSC: 0
SBS? n Max number of CA TSC: 0
Trunk Group for NCA TSC:
Trunk Group for Channel Selection: 84
Supplementary Service Protocol: b
T303 Timer(sec): 10
Near-end Node Name: C-Lan Far-end Node Name: Asterisk
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region: 4
LRQ Required? n Calls Share IP Signaling Connection? n
RRQ Required? n
Media Encryption? n Bypass If IP Threshold Exceeded? n
DTMF over IP: out-of-band Direct IP-IP Audio Connections? n
IP Audio Hairpinning? n
Interworking Message: PROGress
[/tt][/green]
Definity (Node Names):
[red][tt]
display node-names ip Asterisk
IP NODE NAMES
Name IP Address
Asterisk 192.168.1 .51
[/tt][/red]
Asterisk ( /etc/asterisk/ooh323.conf ):
Asterisk ( /etc/asterisk/extensions.conf ):
The SIP phones are both connected using each one a Cisco ATA-186.
Asterisk ( /etc/asterisk/sip.conf ):
I already tested with codecs A-Law, U-Law and G723.1 and they all work great.
For those who don't want voicemail on Asterisk I suggest using G723.1, because it allows huge bandwidth savings (better than G729 with equal voice quality) and don't need any commercial license if Asterisk just switches traffic from Definity to ATAs and the other way around.
Dialplan:
Definity extension range: 1000 to 2699
Asterisk extension range: 2700 to 2799
Network IP addresses:
Definity (C-LAN) : 192.168.1.50
Asterisk (ETH0) : 192.168.1.51
Definity (IP trunk - page 1):
[blue][tt]
display trunk-group 84 Page 1 of 10
TRUNK GROUP
Group Number: 84 Group Type: isdn CDR Reports: y
Group Name: Asterisk COR: 95 TN: 1 TAC: *11
Direction: two-way Outgoing Display? y Carrier Medium: IP
Dial Access? y Busy Threshold: 99 Night Service:
Queue Length: 0
Service Type: tie Auth Code? n TestCall ITC: unre
Far End Test Line No:
TestCall BCC: 4
TRUNK PARAMETERS
Codeset to Send Display: 6 Codeset to Send National IEs: 6
Max Message Size to Send: 260 Charge Advice: none
Supplementary Service Protocol: b Digit Handling (in/out): enbloc/enbloc
Trunk Hunt: cyclical QSIG Value-Added? y
Digital Loss Group: 13
Incoming Calling Number - Delete: Insert: Format:
Bit Rate: 19200 Synchronization: async Duplex: full
Disconnect Supervision - In? y Out? y
Answer Supervision Timeout: 0
[/tt][/blue]
Definity (IP trunk - page 2):
[blue][tt]
display trunk-group 84 Page 2 of 10
TRUNK FEATURES
ACA Assignment? n Measured: none Wideband Support? n
Internal Alert? n Maintenance Tests? n
Data Restriction? n NCA-TSC Trunk Member:
Send Name: y Send Calling Number: y
Used for DCS? n Hop Dgt? n
Suppress # Outpulsing? n Format: public
Outgoing Channel ID Encoding: preferred UUI IE Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Send Called/Busy/Connected Number: y
Modify Tandem Calling Number? n
Send UUI IE? y
Send UCID? n
Send Codeset 6/7 LAI IE? y
Path Replacement with Retention? n
Path Replacement Method: better-route
SBS? n Network (Japan) Needs Connect Before Disconnect? n
[/tt][/blue]
Definity (IP trunk - page 4):
[blue][tt]
display trunk-group 84 Page 4 of 10
TRUNK GROUP
Administered Members (min/max): 1/4
GROUP MEMBER ASSIGNMENTS Total Administered Members: 4
Port Code Sfx Name Night Sig Grp
1: T00208 83
2: T00230 83
3: T00283 83
4: T00284 83
5:
6:
7:
8:
9:
10:
11:
12:
13:
14:
15:
[/tt][/blue]
Definity (Signaling Group):
[green][tt]
display signaling-group 83
SIGNALING GROUP
Group Number: 83 Group Type: h.323
Remote Office? n Max number of NCA TSC: 0
SBS? n Max number of CA TSC: 0
Trunk Group for NCA TSC:
Trunk Group for Channel Selection: 84
Supplementary Service Protocol: b
T303 Timer(sec): 10
Near-end Node Name: C-Lan Far-end Node Name: Asterisk
Near-end Listen Port: 1720 Far-end Listen Port: 1720
Far-end Network Region: 4
LRQ Required? n Calls Share IP Signaling Connection? n
RRQ Required? n
Media Encryption? n Bypass If IP Threshold Exceeded? n
DTMF over IP: out-of-band Direct IP-IP Audio Connections? n
IP Audio Hairpinning? n
Interworking Message: PROGress
[/tt][/green]
Definity (Node Names):
[red][tt]
display node-names ip Asterisk
IP NODE NAMES
Name IP Address
Asterisk 192.168.1 .51
[/tt][/red]
Asterisk ( /etc/asterisk/ooh323.conf ):
Code:
[general]
faststart=yes
h245tunneling=yes
gatekeeper = DISABLE
[definity]
type=friend
context=intern
ip=192.168.1.50
port=1720
disallow=all
allow=alaw
canreinvite=no
Asterisk ( /etc/asterisk/extensions.conf ):
Code:
[general]
autofallthrough=yes
[intern]
exten => 2701,1,Dial(SIP/2701)
exten => 2702,1,Dial(SIP/2702)
exten => _1XXX,1,Dial(OOH323/${EXTEN}@definity)
exten => _2XXX,1,Dial(OOH323/${EXTEN}@definity)
The SIP phones are both connected using each one a Cisco ATA-186.
Asterisk ( /etc/asterisk/sip.conf ):
Code:
[general]
context=intern
disallow=all
allow=alaw
srvlookup=yes
canreinvite=no
[2701]
type=friend
secret=123123
qualify=yes
host=dynamic
callerid="Kerchak"
pickupgroup=1
[2702]
type=friend
secret=456456
qualify=yes
host=dynamic
callerid="Nobody"
pickupgroup=1
I already tested with codecs A-Law, U-Law and G723.1 and they all work great.
For those who don't want voicemail on Asterisk I suggest using G723.1, because it allows huge bandwidth savings (better than G729 with equal voice quality) and don't need any commercial license if Asterisk just switches traffic from Definity to ATAs and the other way around.