Routing incoming calls to Asterisk without UDP

[obukev]

We have an Avaya Definity G3si Ver.6 that handles all of our campus inbound and outbound DID's. We have had no Maintenance agreement with Avaya since the mid 90's, and when the Definity was upgraded to Ver.6, Uniform Dialing Plan was not purchased as one of the options. I have successfully integrated the Definity with Asterisk in that I have a functioning DS1 circuit pack communicating with the Asterisk server, and I can make outgoing calls via the VoIP phones, but I am having trouble routing calls from "the outside world" to the Asterisk server. All configurations I have seen point to the use of UDP and aar tables, but since I do not have UDP stroked to y, I obviously cannot get in and change the system-parameters customer-options without paying for the option... which in turn means that I would have to upgrade to at least Ver.11 before Avaya would touch anything on the switch. UDP was a purchased option until Ver.9 where it then became available by default. I am looking at minimum $30K to upgrade software and Processor boards, and being a non-profit university, funds like that do not come readily available.

What I need is an alternative to UDP. Is there any way I can route incoming calls to the Asterisk server without UDP?

We have purchased a new set of DID's from AT&T to transition to VoIP.

If you need to see my configuration, I have it posted in the following thread:

(the last post contains my working config on the Definity)

http://www.tek-tips.com/viewthread.cfm?qid=1420308&page=1

Any help would be appreciated. Everyone here has been so much help for me.... I am truely grateful.

 

[tlpeter]

Voice only works on UDP
Voip calls are send by udp, tcp gives to much delay
That is the way it is and i think you do not have a choice


ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[kwing112000]

Peter,
he is talking about uniform dial plan. not udp data packets. maybe you could use ars to get the call there i am not sure.

 

[tlpeter]

Oke i didn't say anything


ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[t3chhelp]

You could try creating a x ported station and call forward to a ARS route to your Asterisk server.

 

[obukev]

PHQONEMAN,

I have tried to figure out what you suggested, and the only way I was able to get anything to halfway work was this:

1) Create a new coverage path and stroke all criteria to 'y'; then set Path 1 to r1

2)I then go to change coverage remote and in r1 I put the TAC of the correct trunk group followed by the extension.

I wanted to be able to dump the call to the Asterisk server by just using the TAC associated with the trunk group for the Asterisk tie, but when I do that it does not pass along the information for whom I am calling.... let me explain...

Lets say I own DID's 585-1000 through 2999. I set up the x ported station with the setup I described above...

With the coverage remote(r1) set to my TAC(891) only, and I call 585-1001, it dumps the call to 891 without sending the called number.... 585-1001.... The incoming call makes it to the Asterisk server, but has no information to know what phone extension to send it to. The calling party's CallerID is passed, but the called number is not passed. Do you have any ideas for that, or....

...can you give me some instructions on call forwarding to an ARS route?

 

[SloTCMS30]

obukev,

I'm not real sure how functional your version 6 box is, can you make use of VDN's and Vectors. You might be able set up a MAIN 'transfer' VDN that all stations cover to. In the VDN:

Set the 'Allow VDN Override? to 'y'
Create a vector to your AsteriskNOW server...

:-D

Of course, this is just a guess on my part...


“Do not spoil what you have by desiring what you have not; remember that what you now have was once among the things you only hoped for.”

- Epicurus

 

[obukev]

Thanks for the suggestion, but I do not have vectoring enabled on the Definity. I have decided to setup each station on the Definity as an x ported station and create its own coverage path. I then change the coverage path and set point1 to a coverage remote number. I then do a change cov r and use the same number assigned to point1 in the coverage path to point to the TAC+the extension on asterisk.

If anyone needs info on my setup, reply to this forum, and I will help as best as I can.

 

[t3chhelp]

If you still want to use the call forwarding method, just assign a station with console permissions in the COS. Then use the call foward code, but you will need to put in the xported station then the number you want to foward to. ie foward code *88, xport station 1234, foward to number 95551234#. Hope that helps.

 

[obukev]

PHQNEMAN,

I would definately like to take advantage of this if I can configure it properly. I have created extension 4002 as an x ported station; I have changed cos 12, giving it console permissions, but I did not see any place to assign that station in the COS. I may really be exposing my ignorance here, but am I missing a step? I know the forwarding code to be *5, and to deactivate it is #5. I am just unsure where to setup the forwarding statement.

 

[ahannigan]

i have the same exact thing, did you get it to work?

 

[obukev]

Im not sure if what I am doing is the norm, but I do have something suitable working for our environment. I would be happy to explain to anyone what I have done on the phone. I will post later all of my configurations via a white paper, but6 it is not complete, and i dont want to post in pieces. Email me with your phone number, and I will call you.

kevin dot armstrong at okbu dot edu

 

[jimbojimbo]

Not sure how you are set up on your tie but if you take the incoming call handling and insert the tac code of the tie for the new DID numbers you might be able re-direct the calls from the inbound trunk group to the asterisk tie. Internally users would have to dial the tie trunk tac then the digits. I think you will probably have to be robbed-bit.

 

[rbordeaux]

Not sure if this will work but you might test it. I have an Avaya CM and a Nortel IVR set up, the way I send calls from one to another is using CAS.

I defined the DS1 as:

display ds1 02b0801
DS1 CIRCUIT PACK

Location: 02B08 Name: MPS1-2
Bit Rate: 2.048 Line Coding: hdb3

Signaling Mode: CAS

Interconnect: pbx Country Protocol: 1

Interface Companding: alaw CRC? n
Idle Code: 01010100





Slip Detection? n Near-end CSU Type: other


Then (as is a E1) I set each port of the E1 as follows (from 81001 to 81030):

display station 81001 Page 1 of 3
STATION

Extension: 81001 Lock Messages? n BCC: 0
Type: DS1FD Security Code: TN: 1
Port: 02B0801 Coverage Path 1: COR: 1
Name: IVR Line 01 Coverage Path 2: COS: 1
Hunt-to Station: Tests? y

STATION OPTIONS
Loss Group: 4
Off Premises Station? y
R Balance Network? n


Survivable COR: internal
Survivable Trunk Dest? y


Now every time I dial one of the DS1 extensions the call is sent to the IVR. In asterisk maybe with a little scripting if you might be able to "identify" each line. If each "DS1FD" extension is also a DDI the call should go from your Definity to Asterisk on a "known" port. I know is limited to 30 ports per E1 (or 24 per T1) and even less if you want to transfers calls back from Asterisk to Avaya, or maybe you can set up an incoming and an outgoing ds1) but maybe you can try it.

I have no idea how to set up this on the Asterisk side, I don't have any E1 boards.

My 2c.

 

[vamosalavaya]

Hi,
ARS or AAR should be better than your coverage call forward
Most simple ! Ask to user to dial AAR or ARS feature access code before Asterisk extensions and create AAR or ARS entry that send calls to a route pattern including your asterisk trunk group.
If you want to keep your PortX station , make one coverage path per station with remote coverage (and so ARS and Route Patter and Trunk.. )
Other way ... change you dial plan and assign a range to your asterisk extension shorter than your dial plan normal extension.
Regards

Working on Avaya S8700 and tampering with Asterisk !

http://benasterisk.homelinux.com