Interconnecting a Definity Prologix with Asterisk through H.323 2

[jcpierri]

[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 ):

[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 ):

[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 ):

[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.

 

[victor5908]

Great job Kerchak-Have you set up Asterisk as a gateway
off of a Definity with the Definity providing processor functions?Is that what you are refering to in this example?
If so what is the cost involved?
I'm also interested in Asterisk as a stand alone-Any input is much appreciated-VChip

 

[jcpierri]

victor5908,

My Definity is a Prologix model and, as Avaya told me I can't use any SIP devices on it, I decided to see how far I could go using only free stuff, before considering any kind of upgrade.

I need to use those ATA-186 and intend to try some other even less expensive SIP devices in the future.

Asterisk's highest deployment cost seems to be T1 (or E1) interface cards, so, employing just H.323 as the link to Prologix reduced the cost to just the machine where Asterisk runs (actually an old Pentium-Pro we get around), and nothing more.

As our Prologix users are all connected through VoIP (4612 hardphones and some softphones), this is the way Asterisk users will connect too (ATAs and softphones).

I know there is a long way for Asterisk, before it get all stable features commercial systems use to show, but, so far so good ...

 

[victor5908]

Thanks a lot Kerchak-We will definitely use this in the near future.

 

[m4ilm4n]

Now this is wicked cool! I've been trying to figure out how to integrate Asterix with our 8300, and this looks like a good starting point. Thanks!

 

[Moog73]

Thank you! I has almost given up!

 

[Cicada13]

Kerchak,

I am an Asterisk newbie but have been using Definity systems for a while. I understand from the above config how Asterisk extensions know how to route to the Definity i.e. "exten => _1XXX,1,Dial(OOH323/${EXTEN}@definity)".

But I don't understand how the Definity extensions know how to route to Asterisk extensions? Is this done through the ATAs?

Thanks,
Cicada13

 

[jcpierri]

Cicada13,

I prefer to use Definity ARS feature to route to Asterisk, this way:

[tt][blue]
change ars analysis 27 Page 1 of 2
ARS DIGIT ANALYSIS TABLE
Location: all Percent Full: 17

Dialed Total Route Call Node ANI
String Min Max Pattern Type Num Reqd
27xx 4 4 102 natl n
[/blue][/tt]

[tt][green]
change route-pattern 102 Page 1 of 1
Pattern Number: 102 Pattern Name: Asterisk

Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC
No Mrk Lmt List Del Digits QSIG
Dgts Intw
1: 84 0 n user
2: n user
3: n user
4: n user
5: n user
6: n user

BCC VALUE TSC CA-TSC ITC BCIE Service/Feature BAND No. Numbering LAR
0 1 2 3 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest none
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none
5: y y y y y n n rest none
6: y y y y y n n rest none
[/green][/tt]

[tt][red]
change uniform-dialplan 27 Page 1 of 2
UNIFORM DIAL PLAN TABLE
Percent Full: 1

Matching Insert Node Matching Insert Node
Pattern Len Del Digits Net Conv Num Pattern Len Del Digits Net Conv Num
27 4 0 ars n n

[/red][/tt]

You can also use AAR instead of ARS or none and access Asterisk just dialing its TAC ...

 

[Cicada13]

Kerchak,

Excellent - thanks. You've saved me a lot of headaches with our deployment of Asterisk. We've used Avaya's R300 (now defunct) for calls between offices and it now looks like we have a third, more robust solution.

Merry Xmas,
C.

 

[nurofen]

Hello Kerchak,
Thanks very much for the information.
I have some questions to you, if you don´t mind.
In the configurstion you posted here, are you able to make calls from a SIP phone in asterisk to an extension in Prologix?
I copy the config, but still having some problems with the ooh323 stack (no channel registerd); I am a newbie in asterisk and need time to figure out what is going on wrong with my machine

In any case, thanks again.

 

[whatsanavaya]

i had to change the OOH323.conf file to ulaw.

alaw did not work.

 

[chkbrt]

Is it possible to route all outbound off-net calls through the definity using this setup ? And vise versa using an existing definity DID ? I am looking for cheap way to implement a remote office without buying additional hardware. thanks for any info.

 

[rbordeaux]

The info above was very helpfull but I still cannot make it work.

When dialing from an extension to asterisk I get :
23:17:37 denial event 1178: Normal call clearing D1=0x86b4 D2=0xf10
23:17:37 idle trunk-group 44 member 2 cid 0x5d

When dialing from asterisk to an extension in the avaya I get:
Sep 5 00:03:28 NOTICE[3652]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'OOH323' (cause 66 - Channel not implemented)

Any ideas?

 

[whatsanavaya]

i have the trunk with supplementary protocol = c, yhat might be it

 

[jcpierri]

Enter Asterisk and give a "show channels" command. If OOH323 is available you should see it there.

As OOH323 come with asterisk-addons package, if you're seeing messages like "Channel not implemented" or "no channel registered" when trying to use OOH323, it means you dont have asterisk-addons installed.

Get it from

http://ftp.digium.com/pub/asterisk/asterisk-addons-1.2-current.tar.gz

After following the install instructions, enter Asterisk and give a "show channels" command again. If OOH323 is available you should see it now.

 

[rbordeaux]

Thanks. I was using oh323 rather than ooh323. I did an installation of ooh323 but still not working. I will play around. If you can paste any ooh323.conf config that would be usuefull.

Btw the show channels command doesn't show much. The show channeltypes shows ooh323.


rbast*CLI> show channels
Channel Location State Application(Data)
0 active channels
0 active calls
rbast*CLI>

 

[jcpierri]

My bad.
Where I said "show channels" please read "show channeltypes".

The configuration I posted above (/etc/asterisk/ooh323.conf) is the only one ooh323 need, AFAIK.

 

[TreyS]

Thanks for the info above. I have a Cisco 7940 SIP phone connected to my * and built the trunk group in the AVAYA system as outlined above. I can call the Cisco phone from any AVAYA extension and the Cisco phone rings. The problem is that the talk path is only one way. I can talk on the AVAYA phone and hear it on the Cisco phone but not the other way around.

I also need to work on my dialplan in the * because I cannot call an AVAYA phone from the Cisco phone. It just plays a busy signal on the Cisco phone.


Any suggestions on the one way talk path?

Thanks again for all the help!

Trey

 

[mikeydidit]

You should add this to the FAQ list kerchak, so you work and time will not be lost and can help others.

Very nice job.

"You don't stop playing because you get old. You get old because you stopped playing."

 

[eguander]

Hi Kerchak ,in the first time congratulations for your handbook , it's very good.We have configured the avaya s8700 and the asterisk likely you explain but doesn't work.When you call asterisk appears " network failed" , but the signaling group pass ! Can you help us? anything idea ? Thanks .