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]

yes and yes. busy the sig group, change to tcp 5060. release the sig group. stat the sig group and you'll see if it's in service or not.

Its 'in service" if it gets any SIP response from freepbx - even 500 internal server error - that'll mean you at least got some communication.

Now, it may work with the "ip" in far end domain, but I reckon CM will send messages to xxxx@thatIP and expect xxxx@thatIP back from freepbx which is where you might have a problem.

CM would have every sig group on it's IP and port 5060 - like your trunk to your sip provider too. It's based on the TCP socket that it knows which SIP message is for which trunk. So, the TCP connection procr:5060 to freepbx:5060 is for that, the socket to provider:5060 it knows is for your provider.

 

[robertw1984]

Mmm, I get an error message when I try to change both near and far end to 5060

"5060 must demand maintenance busy the signaling group first

 

[kyle555]

busy sig 7
change sig 7
make your changes
release sig 7
stat sig 7

 

[robertw1984]

busy sig 7" doesnt work, it straight away deletes it and puts in "busyout", so i then type in "busyout sig 7" doesnt work, i then try the full command "busyout signaling-group 7" "busy signaling-group 7" non work

any ideas?

i have done command "display system-parameters customer-options" and page 3 "ds1 msp" is set to no and page 5 "processor and system msp" is set to no aswell

i have done command "change system-parameters customer-options" but obviously i havnt got the rights to change those options to yes

 

[telecomadmin12]

Then remove the signaling group and add it again.

You can have maintenance permissions turned on by Avaya for free.
Then you have access to maintenance commands.

 

[robertw1984]

can i turn it on myself, without having to call my service provider who look after our avaya system?

or

can i just make a new signaling group and new trunk group but will that cause issues?

cheers,

rob

 

[kyle555]

Yeah, go nuts.

a proper SIP design on CM requires you to have many SIP sig groups from procr to even 1 SM. You should have 1 for SIP phones to use, you should have 1 for voicemail, another for PSTN calls. CM can always pick which trunk/sig group to go out via it's route patterns. Inbound is trickier and there's a whole thing to it.

Consider - SIP voicemail and SIP PSTN trunks on SM. SM goes to CM where CM has a sig/trunk group for each of those applications to SM on the same IP/port pair.

When SM sends a call to CM from the PSTN or voicemail, how would CM know which sig group/trunk group it belongs to? It does it by domain matching. It works like this - make your highest numbered sig group @you.com and your lowest numbered sig groups @pstn.you.com and @voicemail.you.com.

CM will look at the invite, see it's on IP/port pairs of "procr:5060 and sm:5060". It will then answer with the sig group that has the lowest number in the system if that sig group's "far end domain" can fit in its entirety into the domain of the "to" header in the invite.

That means if sig group 1 is @you.com, it'll poach everything from @anythingelse.you.com because you.com in sig1 technically fits. That's why the most explicit subdomains need to be lowest numbered and why least specific subdomains need to be highest numbered.

So, unless your SIP PSTN trunks were already going from CM to your freepbx and working, you've got nothing to worry about. Hell, if that was working, you'd already have a working trunk between both systems! What you'd do is route the calls through AAR for private and ARS. We do that all the time - put a CM next to a CS1000 we're migrating from and leave the PRIs on the CS1000 and with 1 tie line between CM/CS1k, you can call CS1000 extensions and send your short 4 digits as CLID and call 9+1+NPA-NXX-XXXX and have CM send the 10 digit public CLID if you set up your call types in AAR and ARS just right.

Make that your next exercise!

 

[robertw1984]

the avaya engineer called me back and he remoted in on my machine to help me out

he got me to "remove sig 7", couldnt do that so he then got me to "ch trunk 7" and the thing i was missing was to obviously remove the "signaling group", leave it blank and make "number of members" to 0 and then i could "remove sig 7" and add it again so job done

now when i do stat sig 7 i get this -

is this right and now working correctly?

if this is correct how do i add a dial pattern so when people/users dial 2XX it goes down the trunk to the freepbx server?

cheers,

rob

 

[kyle555]

dialplan 101. You have avaya support on that? I thought you were messin around with a lab to try and figure some stuff out!

If that's a production system you're using for real phone calls, you might not want to punch in whatever some dude on the internet told ya...

 

[robertw1984]

i trust you guys as your tech savvy pros

 

[robertw1984]

Awesome found a good video on how to do aar dial plans -

https://youtu.be/putIGdNFS-4

 

[robertw1984]

am i missing anything off because on my avaya phone when i dial extension 201 i expect for it to start dialling to my freepbx voip system via the trunk but its not (my softphone on my pc connected to the freepbx)