underdog

Hello,

The forums have been great on helping me to get this setup. I just setup a tie trunk to the VoIP gateway I installed and all is working great but I have a problem setting up a route. If someone could help I’ll be glade to post the progress I did to get a Trixbox which was Asterisk@home with a Sangoma four port ISDN-Pri card on the forums with a step by step guide.

Definity G3si+m R4

I’m dialing out from the PC to the switch 8620 the 8410d rings I pick up and talk works great and also receive caller-id. My softphone number is lets say 4000 how do I pick up the 8410d and dial my softphone? Please step by step newbi on the routing.

Thanks,
Steve

underdog

Could someone explain the AAR? If I call ext. 3000 I need it to look on the pri trunk 22 for that ext.

Thanks

PBXtech.info

 

martinyoung

Do you have "Universal Dial Plan" enabled on your cust options?

Caveat:I have not done an R4 for years, so the commands might differ little.

Type "ch udp 3000" this will take you to the UDP screen with your cursor blinking at 3000. Type "u" and press tab and you will go to a new field. Enter a three digit number, we'll say 221 for now, and press enter.

Do a "ch aar an 2" and you will be on the aar analysis screen. If you already have entries there, use the down arrow to get to a vacant line. Enter the UDP code you assigned (221, for example), the minimum and maximum digits you need (and I don't know how many you need, so let us assume four), a new route pattern that you are not presently using and call type aar. Press enter.

Go to the new route pattern you created and assign the trunk group you want (22), FRL will usually be 0 and delete three digits.

Your call will work this way;

Dial 3000, the PBX will examine the UDP table and see that you added a UDP code so the PBX changes your dialed number to 221-3000.

It now checks the AAR Analysis table and sees that any calls starting with 221 will be directed to route pattern xx.

At route pattern xx it sees that the call is to use trunk group 22 and that you want to delete three digits, so the 221 is stripped off and 3000 is sent out the trunk group. If you need more than four digits then there are other choices to do on the AAR Analysis table and the route pattern.

__________________
Marty
Retired Avaya DSIC tech

underdog

Thanks for the help

I don't have this enabled if I get this enabled by Avaya will this hurt anything I have now?

martinyoung

Not a bit, it will give you the ability to use the AAR tables.

__________________
Marty
Retired Avaya DSIC tech

underdog

Thank you very much this helps out alot!!! I was getting worried there....:D

PBXtech.info

 

underdog

Marty,

One more for ya,

I have this all working inbound/outbound calls but if I dial a long distance number it won't work... We do have a carrier that send a tone and we have to put a number in to let the call go out. What I need to know what is it sending out to get the tone? Like a code number or something?

Thanks

oozenoz

You are describing an authorization code. The presentation of an auth code tone is not triggered by your system, it is provided as a service by your long distance provider. (You can also set up authorization codes on your own system). In either case the implementation of auth codes would not affect calling number presentation.

If you have dedicated trunks going to the long distance carrier, check the number analysis in the same way you did for your local calls to make sure you are actually sending something. If you are routing your long distance calls through your local telco, your caller ID is being stripped by either the local telco or the long distance carrier - so you will need to contact them to get it sorted.

__________________
- calvin

martinyoung

You set up CO provided auth codes with the vendor. If you do not know the auth code (or codes) you will need to contact them.

I prefer to do auth codes from within the PBX for a few reasons.

You can be more flexible with them. If you have 1000 users, you can give each one an auth code to control every call they make. You can, for instance allow some users to make long distance calls over the more expensive local circuits if all the long distance circuits are full. Others you could block.

If a code becomes public knowledge (and they will), you can delete it.

You can add and remove codes as needed. Lawyers, for example, would use an auth code as an account code. When Accounting looks at the Call Accounting records from the PBX, they can compare auth codes with the client database and bill the call to the correct client.

__________________
Marty
Retired Avaya DSIC tech

oozenoz

You can also create or change codes much more quickly.

Some carriers will make you wait several days - sometimes even a week or more - to get auth codes created or deleted.

__________________
- calvin

underdog

Thanks for the information...

How I see it, we have one trunk going out on a PRI to local phone company (Bellsouth) , I dial out from the PBX long-dis ie 1-545-XXX-XXXX i get a tone I then dial the right account code and I get out.....

Okay this is the problem, I hooked up asterisk on a its own PRI I dial the same number and it just rings. can you tell me how that works? What is the PBX doing that asterisk/PRI is not doing? I can dial local numbers with no problems just not long-dis. ohh this hurts


Thanks

martinyoung

I know absolutely nothing about Asterisk so I cannot help with it but I did a Google search for Asterisk and there is a lot on line about it including some updates and patches. I would start there.

__________________
Marty
Retired Avaya DSIC tech

oozenoz

Are the auth codes programmed in the PBX or programmed by your LD carrier?
a) PBX - you need to program auth codes in the Asterisk.
b) LD carrier - make certain the PRI the Asterisk is using is using the same carrier and that they have assigned the auth codes to that PRI as well. (This is not something they would do automatically)

__________________
- calvin

underdog

This is the route I'm going..


Phone--->>VOIP BOX---->>PRI--->>DefinityG3----->>PRI--->>>Bellsouth

I have a PRI going to the definity and then the definity goes out to bellsouth.

If I pick the phone up off the definity it works I get the tone to input the Auth code but if I dial out on the VoIP box I get no tone only ring ring then a hang up. What do you think the switch is doing? I also checked and the call goes out/in the PRI only. It's like I need to share/open the definity PRI so the VoIP box can share it.

Thanks

martinyoung

What is your connection between the Definity and your VoIP box? I know you show it as a PRI but what does the Definity think it is? If the Definity sees it as another PBX then your PRI is a tie trunk. In that case you need to check the COR and FRL of both trunk groups, from the Definity to the outside world and to the tie line. Make sure both are compatible. Keep in mind that an incoming tie trunk call that transfers to another trunk group drops one FRL and becomes one step more restricted.

Also verify that the COS permits trunk to trunk connection.

You can do a trace in the Definity on the incoming circuit from the VoIP box, that also might tell you what is happening.

__________________
Marty
Retired Avaya DSIC tech