connecting my VoIP to my CM system

[robertw1984]

hi all,

i have set up a freePBX server and now i want to connect it up to my current CM system so the two can link together for calls/transfers etc

atm the way i have linked the two servers up is via an extension number on my CM system, i have made a virtual extension number which dials a remote number via a call coverage path and it works

is there anyway of doing this please?

many thanks,

rob

 

[Wanebo]

Look in system manager and see. There is a listing for Session Manager. Click that and see if there is one configured.

 

[robertw1984]

i have "start gedi" in the general tab and "start emulation" on the advanced tab

EDIT - im in system manager (start gedi) and dont see a session manager anywhere, is there a command i need to type in the box to display it

 

[kyle555]

You'd need to deploy a session manager virtual machine and enroll it to SM. Then you'd trunk CM to SM and SM to your Asterisk.

Just build a SIP sig group in CM to the IP of your asterisk, then GEDI and no SM/SMGR required.

This is just a lab to learn on, right?

 

[telecomadmin12]

If you don't want to do that, use H.323, of which I can provide the config for both ends.

 

[Wanebo]

Sounds like you are in the Avaya Site Administration (ASA) program, not in a System Manager. There is no "Start GEDI in System Manager, but there is in ASA.

 

[robertw1984]

do you mean i need to create a new virtual machine just to run session manager on, you dont think session manager is on the system manager pc?

how do i build a SIP sig group in my CM

yes this is just a lab to learn on as theres alot to learn on avaya and i know the bare basics

telecomadmin12 - thanks, i will hold you to that but i really want to learn for myself, are you going to give me files?

 

[kyle555]

Correct, session manager is it's own virtual machine, managed by a separate virtual machine System Manager.

You'd "change node-names ip" to add a name - "anotherserver" and a IP. Then you'd "add sig 1" and fill out that form to add the signaling group. It can be H323, SIP, or ISDN if you've got PRIs. Once you pick one that is voip - like h323 or sip, you'd put the "far end node name" and port which would be "anotherserver" you added in the node-names screen.

It's the same whether you'd trunk CM to SM or your Asterisk, just the IP is different. Now, you can SIP trunk CM to anything, but having CM hang off Session Manager and have everything go through that is "better" or "right" or whatever you want to call it.

Avaya's architecture is very hierarchical. You need a VM for everything. They can have flexible footprints, so a Session Manager can use as little as 3 vCPU and up to 20 depending how much you want to do with it.

If you're learning, and got a CM and System Manager (SMGR), install a SM too if you can. CM to SM, SM to everything else.
Otherwise, if you want quick and dirty, just build a group with near-end "procr" and far end "thenameofyourAsterisk" as TCP and see how far you get.

 

[telecomadmin12]

If you decide to go with h323, this would be a starter on the FreePBX end:

https://community.freepbx.org/t/h323-trunk-with-avaya-cm/35473

 

[robertw1984]

ok thanks but what one have i got, i think i have CM not SM

so if i have CM only can i do CM > freepbx server but if i had SM i could do CM > SM > freepbx server, is that right?

i havnt got any software for any of our avaya gear as it was all set up for us on a xp machine (waaaaay old!)

EDIT - ok i have successfully entered in those commands and looks like im on to a winner

the signalling group i imagine i select SIP as im connecting it to my freePBX server and that is SIP?

http://files.engineering.com/getfil...08-4cc6-9949-35690e9c3a21&file=nodes_sigs.PNG

 

[kyle555]

Yup. you need to build a trunk group of type sip as well afterwards. There's not much to it.

 

[robertw1984]

So after I add make a new node and signalling group I need to then add a new trunk?

If so what's the command for that

Edit - awesome it's in that video YouTube link I posted earlier

I will do this tomorrow and let you know

 

[robertw1984]

im stuck on the "add trunk-group 7" what do i enter in the "TAC" field?

i imagine "service type" is tie and not public-ntwrk as its not going out, ie its an internal server?

here is a list of my trunk groups atm

http://files.engineering.com/getfil...d-445f-8f9a-e2e42fcb4e85&file=trunk_group.PNG

 

[montyzummer]

Tac is trunk access code in your case you can use 807 , to see where this is referenced look at "display dial ana" and look at the DAC entries.

ACSS (UC/SBCE/SM/SME)

Not that they mean a thing anymore , get a brain dump pass the test crash the system.

 

[robertw1984]

i dont see 807 on the list

 

[kyle555]

80=dac, that means in any given trunk you can use 800-809 as a "tac" or "trunk access code". Every trunk needs one by definition despite them not being used much anymore

 

[robertw1984]

mmm i still dont understand but i will research into it abit more

also dont i need to create a dial plan/pattern so when you press 4 digits ie 6XXX it goes to the extensions of the freepbx server via the trunk?

 

[robertw1984]

what you think, now i need to make the trunk the other end ie the freepbx side but i know how to do that as i have done one for my SIP provider

 

[kyle555]

1.Use TCP 5060. stat your sig and trunk group and see if it's up
2. if you have problems, make far end domain an actual domain and not ip. CM doesn't jive nice with IP addresses in there. say like mycmtrunk.com and you might need to create a DNS SRV record for the SIP domain mycomtrunk.com to point to your CM IP on 5060 so your freepbx resolves the domain to the IP to send back.

 

[robertw1984]

thanks kyle, i will create a dns a record for my freepbx and change the port to 5060, this aswell is the same port i connect to my SIP provider and my internal softphones and hardphones ie PJSIP

once i have configured both ends how can i determine if the trunk is online etc?

i will get back to you

 

[robertw1984]

when you say change it to port 5060, do you mean change both ends ie near end node and far end node?

and also use tcp, do you mean instead of tls for transport method?