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

 

[kyle555]

Are you asking about trunking them together or how to manage your dialplan? It sounds like you've got the trunking part down.

AAR (or ARS) and Uniform Dial Plan is what you're looking for. Either you dial the AAR or the ARS access code, then the extension on your other system and in the ARS or AAR table you have a match for that extension range that goes to a route pattern which as first choice has the trunk to your other system.

You can avoid explicitly needing to dial the AAR or ARS code if you let UDP handle it. So, "display dialplan parameters" and you'll see if CM takes a dialed number and tries to match "udp first then local extension" or vice versa. Either way, if say the 4xxx range had no extensions on your system and in "change uniform 0" you had 4xxx match "aar", then anytime you'd dial a 4xxx number, it'd hit AAR and pick the trunk.
Was that what you were looking for?

 

[robertw1984]

ok maybe i should have worded it better, sorry

atm i have an extension number that in turn dials a remote number via a cov path and it then dials out the system as its a landline number and then comes back into the VoIP server via the SIP trunk and then it goes to an extension number

instead of it going out and coming back in via the SIP trunk as its got to dial a landline number can it just link to the voip server via an extension number so its all internal

am i making sense?

 

[kyle555]

Yeah, you should be able to SIP or H323 trunk them together. I believe there's a forum FAQ for specifically that. Otherwise, it's relatively straightforward.

 

[robertw1984]

how do you do it please if not what should i search for to find out how to do it?

 

[kyle555]

here's a good start with links pointing back to this forum

http://forums.asterisk.org/viewtopic.php?p=53750

here's another bit:

http://www.mehrdust.com/archives/asterisk-tips-for...

 

[robertw1984]

can i set/config this all from the freepbx server or do i have to configure it from the avaya cm system

 

[kyle555]

both. Each needs a trunk/signaling pointing to the other to get their link up. Then you can play with numbering to decide what should route over it.

 

[robertw1984]

thanks Kyle for your help

im not comfortable configuring making changes to the avaya cm system as i havnt a clue, the only thing i do is configure on the gui like pickup groups/cov answer paths/cov paths/add new extensions

is there a gui way i can do this?

 

[kyle555]

no. System Manager maybe, but it's not going to be any easier or more intuitive.

 

[robertw1984]

does this video show what i want to do -

https://www.youtube.com/watch?v=Sius0AaLoBw

 

[kyle555]

Yeah, if you want to build a SIP trunk. I'd stick to someone who's done it from the asterisk side and on H323. The Avaya side is trivial for H323 and it's a far simpler config.

You technically shouldn't be sip connecting anything to CM without a SM on top. It'll probably work fine either way, but once you know the deal, you could h323 connect 2 CMs together in <5 mins. H323 is great. Its basic, inflexible, it has fixed values for almost everything. SIP does way more - and requires you know how to define and configure it.

 

[robertw1984]

mmm... i thought i wanted a SIP trunk from avaya to freepbx and vica versa?

what is h323, so i need to configure a h323 trunk from avaya to freepbx and then from the freepbx a SIP trunk to avaya?

 

[kyle555]

H323 is a voip protocol. It's Q931 signaling over IP - so basically the same structure as ISDN messages.

SIP is newer. It doesn't support any particular media specification. That's to say you and I could invent something better than 4K video and call it 5K video. SIP signaling lets you call that a media type and so long as you and I know how to handle the media, SIP can set that session up for us.

SIP also has a lot of flexibility. I guess if you've got a decent handle on the basics and have set up a SIP trunk on your Freepbx before, then yea, go for it.

 

[robertw1984]

thanks Kyle,

but surely all i do is create a SIP trunk from my avaya system and create a SIP trunk on my freepbx so the two can link up, i know what your trying to say that its not best to create a SIP trunk on the avaya system but if its there sureley it can be done or have i got this totally wrong?

i also attach a screenshot of the different trunks i can make on the freepbx system, what one of these should i use to link up to the avaya system

 

[telecomadmin12]

He is on R015x.01.2.416.4
Does he need session manager to connect a SIP trunk?

I can give you a working config for H.323 for both ends, the Asterisk and the Avaya CM.
I have done it, it's no problem.

Configuring the SIP on the Asterisk is easy as well, not sure about the Avaya CM part.
Can he run a SIP trunk from Avaya to the Asterisk box directly or does he need session manager or something in between with his CM version?

(I am by the way the one who answered your post on the FreePBX forum).

 

[robertw1984]

session manager? are you talking about the GUI where i make all the changes on the "Avaya CM Site Administration"?

do i need to make the trunk via H323, cant i make the trunks via SIP?

 

[Wanebo]

Usually you need a Session Manager in the mix when you are using SIP to connect to an Avaya CM. That's because pretty much everything out there speaks a dialect of SIP (including Avaya) and Session Manager offers the ability to use Adaptations to modify some of the SIPO signaling so that both sides can understand each other better.

 

[robertw1984]

mmm... how do i know if i have session manager?

 

[Wanebo]

System Manager is the GUI you refer to. You use it to modify elements, CM, AAM, AAC, Session Manager, etc.