PBX is down (CUCM) want to replace with FreePBX - router config questions

I am not sure if this is the right place to ask this but here it goes:

My CUCM is down and I need to replace it asap.

My current config goes something like this

EndUser <-------> CUCM <------> Cisco2801<-----PRI-----> TelcoProvider

I want it to look like this:

EndUser <-------> FreePBX <------> Cisco28XX <-----PRI-----> TelcoProvider

I did find this here:

But I am kind of a chicken looking at a wristwatch - not sure what I need to set on the Cisco side to make this doable.

Here is a copy of my current config of the router (washed a little of some info)

!
boot-start-marker
boot-end-marker
!
card type t1 0 2
enable secret 5 *****
!
no aaa new-model
clock timezone CST -6
clock summer-time CST recurring
network-clock-participate wic 2
dot11 syslog
ip cef
!
multilink bundle-name authenticated
!
isdn switch-type primary-ni
!
voice-card 0
 dsp services dspfarm
!
!
archive
 log config
  hidekeys
!
!
controller T1 0/2/0
 framing esf
 linecode b8zs
 cablelength long 0db
 pri-group timeslots 1-12,24
!
controller T1 0/2/1
 framing esf
 linecode b8zs
 cablelength long 0db
!
interface FastEthernet0/0
 ip address 10.2.7.5 255.255.255.0
 duplex auto
 speed auto
 h323-gateway voip interface
 h323-gateway voip bind srcaddr 10.2.7.5
!
interface FastEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
!
interface Serial0/2/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 no cdp enable
!
ip default-gateway 10.2.7.1
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.2.7.1
!
!
ip http server
no ip http secure-server
!
control-plane
!
disable-eadi
!
voice-port 0/2/0:23
!
dial-peer voice 1 pots
 incoming called-number .T
 direct-inward-dial
!
dial-peer voice 1000 voip
 destination-pattern [0-8]T
 session target ipv4:10.2.7.6
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 1001 voip
 destination-pattern [0-8]T
 session target ipv4:10.2.7.7
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 2000 pots
 destination-pattern 9.T
 port 0/2/0:23
!
dial-peer voice 2001 pots
 destination-pattern 911
 port 0/2/0:23
 prefix 911
!
call-manager-fallback
 max-conferences 4 gain -6
 transfer-system full-consult
 limit-dn 7910 1
 limit-dn 7940 2
 limit-dn 7960 2
 ip source-address 10.2.7.6 port 2000
 max-ephones 24
 max-dn 48
 default-destination 2000
!
!
line con 0
 password 7 *******
 login
line aux 0
line vty 0 4
 password 7 *******
 login
!
scheduler allocate 20000 1000
ntp clock-period 17178243
ntp master 4
ntp server *******
ntp server *******
ntp server *******
end

If someone has an idea of what I need to change and how I need to change it to make it work with FreePBX I would greatly appreciate it. Thanks!

This is a different wristwatch, but I think I can make some suggestions:

  1. Reduce the risk by installing a FreePBX system and installing Chan-SCCP-B to talk to your phones. There is an SCCP Manager by PhantomVI that allows you to do your thing with the phones pretty simply.

  2. Change your provider’s information to point at the Ethernet of the new FreePBX box and set up SIP to talk to the box from your provider. It looks like you’re already set up with a provider that delivered SIP to your service, so that should be relatively simple.

  3. Get your list of phone MAC addresses and extensions and start mapping. This is the hard part.

Drop the FreePBX box in in place of your CUCM server, turn it back on, and spin her up. This way, you don’t mess with the phones, there shouldn’t be a lot of training involved (although you will require some agile thinking to get everything perfect). It’s not going to be simple, but it beats the crap out of flashing all of the phones to SIP and then dealing with the differences, as well as doing all of the work to get the extension and phone config straight.

https://github.com/chan-sccp/chan-sccp/wiki/Setup-FreePBX is where I started the last time I did this (last weekend).

you’re going to need help setting up your cisco28xx to have a SIP trunk to the freepbx. the choices are mgcp , h323 and SIP.
right now you have h323 which technically can work with freepbx but i would switch to SIP

the phones will register to freepbx but they are all using sccp cisco protocol, and the guy above mentioned how to get that working without reflashing phones to SIP.
i think he’s mistaken you have a SIP trunk to the provider; all i see is PRI

so build a SIP trunk between the cisco GW and freepbx and the calls will terminate via the cisco GW if you keep using the existing dialplan.

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