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[Asterisk] PBX in a Flash with Multiple CC DIDs

Normally, one puts the incoming DID dial-plan entries in a context named something like from-carriername that is specified in the context for carriername in your sip.conf (or sip_custom.conf).

The thing you might be missing, something that took me by surprise when setting up CC on an * box for the first time, is that for some unknown reason, the incoming calls never end up in the from-callcentric context. They come in, instead, in the context named from-internet.

SO, in your from-internet context, you need to add the CC DID routing, for example:

exten => 17771234567,1,Goto(from-internal,1234,1)

Hope that helps.

reply to Trimline
Have you tried creating inbound routes in PBXiaF for each DID? I'm afraid that CC is one carrier with which I have never worked. Give it a meaningful (to you) name and then make sure you define the expected DID . You may have to watch your log to determine if CC delivers it with a leading 1 or just the 10 digit number (some VoSPs use an account name instead to make things more difficult )

reply to Trimline
Straight from CC when I asked that question some time back:

I'll try this, thanks. Did you have to set Allow Anonymous Sip to Yes under the General Settings tab?

I do run anonymous sip set to yes, but not for anything related to this particular set up. The DID from Callcentric is in the SIP invite and these settings will allow you to determine which DID the call is coming from, and direct it accordingly.

Edit: You probably are already aware of this, but I use the CID name prefix in the incoming route to have it prepend the caller id so I know which DID it is coming in on.

reply to Jet101
Jet: Thanks for the tips, but I found all kinds of issues with Asterisks 1.8.1 using Callcentric. It took me about 8 days of fussing around to finally get CC to play nice. I believe in technical solutions. I'll document my recipe (taken from bits and pieces from other attempts) in the event others need it. What I've read some folks have cancelled their CC due to this!.

Step One. Add the following to your sip_custom.conf :

context=from-sip-external-custom
srvlookup=yes
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas


Step Two: Add the following to your extensions_custom.conf :

[from-sip-external-custom]
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)
; CallCentric Check
exten => s,1,GotoIf($["${DID}"="1777yourDIDONE"]?callcentric)
exten => s,2,GotoIf($["${DID}"="1777yourDIDTWO"]?callcentric)
exten => s,3,GotoIf($["${DID}"="1777yourDIDTHREE"]?callcentric)
; Regular Check
exten => s,n,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?checklang:noanonymous)
; CallCentric DID Code
exten => s,n(callcentric),Set(Var_FROM_DOMAIN=${CUT(CUT(SIP _HEADER(TO),@,2),>,1)})
exten => s,n,GotoIF($["${Var_FROM_DOMAIN}" = "callcentric.com"]?callcentric-next)
exten => s,n,GotoIF($["${Var_FROM_DOMAIN}" = "ss.callcentric.com"]?callcentric-next)
exten => s,n,GotoIF($["${Var_FROM_DOMAIN}" = "66.193.176.35"]?callcentric-next:checklang)
exten => s,n(callcentric-next),Set(Var_TO_DID=${CUT(CUT(SIP_HEADER(TO),@,1) ,:,2)})
exten => s,n,GotoIF($["${Var_TO_DID}" = ""]?checklang)
exten => s,n,Set(DID=${Var_TO_DID})
; Regular script continues
exten => s,n(checklang),GotoIf($["${SIPLANG}"!=""]?setlanguage:from-trunk,${DID},1)
exten => s,n(setlanguage),Set(CHANNEL(language)=${SIPLANG})
exten => s,n,Goto(from-trunk,${DID},1)
exten => s,n(noanonymous),Set(TIMEOUT(absolute)=15)
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,Hangup
exten => i,1,Hangup
exten => t,1,Hangup


You may add as many Callcentric DID's as you have; see the line 1777yourDIDONE. In this case, I have three DID's. If you have more, just add S4, S5 etc., or remove the extras.

Step Three Your Extensions:
Your "Add Inbound DID" should simply be your 1777yourDIDONE number.

Step Four Your Trunk
Make sure your trunk definition has context=from-sip-external-custom - the setup for trunk would also include type=peer. The Registration would read:


1777XXXXX:password@callcentric.com/1777XXXX


Step Five Define your inbound/outbound routes.
If you want each Extension to use a separate outbound route (I do) go to »mirror.freepbx.org/modules/relea···modules/ and grab customcontexts-2.8.0rc1.1.tgz. This allows you to specify what outbound route each extension is permitted to take.

This is about as simple as it gets and it actually works!

Cheers