How Outbound calls are routed from a branch

[rahul14]

I am seeking assistance in understanding how outbound calls from a branch works. I want to know which command/config on the router or Call Manager causes an outbound call to be routed through a branch router fxo or the T1/SIP trunk at head office and what is shown on the recipients caller id. I have a new branch that I want whenever a call is made, it request a FAC from the Call Manager at HQ however  it should utilize the FXO lines and the recipient would only see the number associated with the FXO line and not HQ trunk lines. What happens now is sometimes the FXO lines are used and other times the T1 trunk line at HQ used. It appears to algorithm however not sure which one and where this is being done. See snippets below. Any assistance would be appreciated.

trunk group POTSGROUP
 voice-class cause-code 1
 hunt-scheme round-robin

!
!

voice-port 0/0/1
 trunk-group POTSGROUP
 supervisory disconnect dualtone mid-call
 timeouts interdigit 7
 timeouts call-disconnect 3
 timeouts wait-release 3
 connection plar opx 8032
 description POTS LINE
 caller-id enable
!
voice-port 0/0/2
 trunk-group POTSGROUP
 supervisory disconnect dualtone mid-call
 timeouts interdigit 7
 timeouts call-disconnect 3
 timeouts wait-release 3
 connection plar opx 8032
 description POTS LINE
 caller-id enable
!
voice-port 0/0/3
 trunk-group POTSGROUP
 supervisory disconnect dualtone mid-call
 timeouts interdigit 7
 timeouts call-disconnect 3
 timeouts wait-release 3
 connection plar opx 8032
 description POTS LINE
 caller-id enable
!

[Earl Granger IV]

Your dial-peer commands are what determines the destination.  You can post your dial-peer configuration as well.

[rahul14]

Thanks for responding. See below.

NB I have both pots and voip dial peer. The VoIP dial peeer has a destination pattern to a translation pattern(8032). If I have both POTS and VoIP how does it know which one to choose and is the translation pattern used somewhere in the Call Manager config to choose between local FXO lines at the branch or T1/SIP trunk at HQ?

dial-peer voice 3 pots
 trunkgroup POTSGROUP
 description Local
 destination-pattern 9[2-9]......
 forward-digits 7
 no sip-register

dial-peer voice 510 voip
 description Outgoing To Primary CUCM
 preference 1
 destination-pattern 8032
 session target ipv4:10.50.1.11
 incoming called-number 9.T
 voice-class codec 100 
 voice-class h323 100
 dtmf-relay h245-alphanumeric
 ip qos dscp cs3 signaling
 no vad

dial-peer voice 511 voip
 description Outgoing To Secondary CUCM
 preference 2
 destination-pattern 8032
 session target ipv4:10.50.1.12
 incoming called-number 9.T
 voice-class codec 100 
 voice-class h323 100
 dtmf-relay h245-alphanumeric
 ip qos dscp cs3 signaling
 no vad

[michael o'nan]

I would suggest signing up at ine.com and watching their free CCNA Voice videos. I believe it would benefit you more to learn the fundamentals with demonstration.

[rahul14]

Ok thanks Michael. I have watched CCNA voice videos before however just having a bit of challenge understanding when a call is made from a branch how does it choose between the FXO at branch or the T1 and SIP trunk at HQ. If i want to say make all the outbound calls from a couple branches office use their local FXO lines and only use the t1 or SIP trunk at HQ only when their FXO lines are not available. How could this be accomplished?

[michael o'nan]

Are you using Call Manager for call control or CME? If CUCM are the gateways MGCP, SIP, or H.323? If MGCP then the call routing decisions will be made on CUCM. If you are using SIP or H.323 then your dial peer selects the best path for call routing as set by you. Your dial peers above are confusing as you have  incoming called-number 9.T

I believe this would only work if you were prefixing the 9 to match the dial peer and then it would send the call to CUCM. The voip dial peer is for routing calls to things like CUCM, CUC, CUCME, or a SIP/H323 destination. The POTS dial peers are for things like FXO, PRI, and BRI.

[rahul14]

Call Manager at HQ is being used for Call Control. Gateways are h.323.

The aim is for these 2 branches to use their FXO lines only when making a external call but there should be a request for their FAC code before they can make the call.

The other branches would use their FXO lines and use the t1 at HQ  only when the FXO lines are engaged.

[michael o'nan · Accepted solution]

That gives me a better idea...

Create FAC codes and levels.

Create route patterns and point them to the route list of your gateway or use SLRG. On the route pattern you will tell it what level FAC to use.

Your route list could contain the T1 at HQ under your FXO RG. This would allow the call to roll to the T1 should the FXO be filled up at that site.

As long as the phones are registered to CUCM, they will be required to use FAC. If this were CME or SRST scenario this wouldn't be the case.

Here's a decent link on the setup of FAC:

http://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-communications-manager-callmanager/81541-fac-config-ex.html

[rahul14]

Thanks its getting clearer. The route group for those 2 branches only contain their respective gateways. I notice however that only 2 of the 4 FXO ports have physical lines connected to it and when there is a hunt on the other 2 empty ports the outgoing call is placed over the WAN to the T1/SIP
 

[michael o'nan]

Looks like you have a trunkgroup called POTSGROUP. You may want to make sure that the two unused FXO ports aren't in that group.

[rahul14]

Oh ok kool. So by having the POTSGROUP in a voice port that is not being used causes it to to try and route over the WAN? I thought since there was no physical lines in those ports it would automatically skip those ports and select the ports that have the lines connected.

[michael o'nan]

You have this outbound PSTN dial peer:

dial-peer voice 3 pots
 trunkgroup POTSGROUP
 description Local
 destination-pattern 9[2-9]......
 forward-digits 7
 no sip-register

I can't see your FXO config but I am assuming you have trunk-group POTSGROUP on all 4 FXO ports? The FXO hardware can't detect if a line is actually plugged in or not.

The POTSGROUP has nothing to do with it routing the call over the WAN. Your route list is setup to route calls out FXO on the remote site gateways. If FXO is unavailable it goes to the next group in your route list which is the PRI at HQ.

[rahul14]

Yes the POTSGROUP is on all FXO ports. Ok understood. There is no route list associated with the route group in which those gateways are assigned. The route lists include route groups for the PRI/SIP trunks at HQ.

I have shut down the voice ports that do not have lines connected and it now only goes out the local FXO.