5 Replies Latest reply on Sep 24, 2018 2:34 PM by jimtgs

    Looking to add VOIP capability to old PBX

    jimtgs New Member

      We have a Mitel SX-200EL with a PRI circuit to the telco and a spare T1 port.  We want to link the T1 port to a VOIP asterisk style PC for VOIP phones. We want to do T1 E&M signalling to a Total Access type unit and then register the total access as a SIP trunk to the asterisk. The end result would be to route calls via the T1 to the asterisk to voip phones. It may not be perfect but we're willing to settle. I have complete control of the MItel and can use PRI as the link instead of T1 E&M.

       

      Can this be done and which Total access should I be considering?

        • Re: Looking to add VOIP capability to old PBX
          markfreeman Employee

          Yes this can be done. You can use a 904:

           

          Total Access 904

          P/N: 4212904L1

           

          You don't care about FXS interfaces. That box has one PRI/T1 interface and one Eth interface.

           

          I would recommend using PRI over EM because with PRI you get Caller ID and EM you don't.

           

          But basically you terminate the T1 into the 900 and then you create a SIP trunk to Asterisk. This is a basic sip to PRI application turned around. Done all the time.

           

          Let me know if you have any questions.

           

          here is sample config of PRI to SIP:

          IP Business Gateway PRI to SIP PSTN Gateway Sample Configuration

           

          -Mark

            • Re: Looking to add VOIP capability to old PBX
              jimtgs New Member

              Is it possible to register a SIP client to the 904. Not a PBX but for example a basic IP phone?  That way we remove the 'asterisk' server.  For example I could route a certain incoming set of digits over the PRI to a specific SIP phone? 

               

              I did not know this was 'done all the time'!

               

              Thank you!!!!

                • Re: Looking to add VOIP capability to old PBX
                  jayh Hall_of_Fame

                  Absolutely. Create a voice user with the extension just like an analog station.

                   

                  Instead of "connect fxs 0/1" configure the following for the user:

                   

                   

                  sip-authentication - Configure sip authentication information for this user
                  sip-identity       - Configure registration for this user
                  sip-keep-alive     - SIP keep-alive configuration
                  sip-register       - Configure registration settings

                   

                  Configure the SIP phone to register to the IP of the Adtran device.

                • Re: Looking to add VOIP capability to old PBX
                  jimtgs New Member

                  I can make calls from my VOIP pbx to the PRI to the PBX

                  I cannot make calls from the PBX to the VOIP pbx. I get a fast busy

                  I dial 9 plus the 4 digit extension of a sip extension on my voip pbx 9 4101 and I get a fast busy.

                  I get a fast busy no matter what I dial on the PBX.  I added some entries on my pbx dial plan to allow 4101.

                  I know the PRI card is good, incoming works, not outgoing.

                  I cannot tell if my sip trunk between systems is good but calls go through. it shows no sip trunk registrations.

                  both of these units are sitting on a bench, connected by a switch, no firewall, no nat.

                  !

                  !

                  ! ADTRAN, Inc. OS version R10.9.4.E

                  ! Boot ROM version 14.04.00

                  ! Platform: Total Access 904 (2nd Gen), part number 4212904L1

                  ! Serial number CFG1073449

                  !

                  !

                  hostname "TA904"

                  enable password mark

                  !

                  clock timezone -5-Eastern-Time

                  !

                  ip subnet-zero

                  ip classless

                  ip routing

                  !

                  !

                  name-server 192.168.10.1 8.8.8.8

                  !

                  !

                  no auto-config

                  !

                  event-history on

                  no logging forwarding

                  no logging email

                  !

                  no service password-encryption

                  !

                  username "admin" password "password"

                  !

                  !

                  no ip firewall alg msn

                  no ip firewall alg mszone

                  no ip firewall alg h323

                  !

                  !

                  !

                  !

                  !

                  no dot11ap access-point-control

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  interface eth 0/1

                    ip address 192.168.10.23 255.255.255.0

                    media-gateway ip primary

                    no shutdown

                  !

                  !

                  !

                  !

                  interface t1 0/1

                    shutdown

                  !

                  interface t1 0/2

                    tdm-group 1 timeslots 1,24 speed 64

                    no shutdown

                  !

                  !

                  interface pri 1

                    description pri 1

                    isdn name-delivery

                    connect t1 0/2 tdm-group 1

                    no shutdown

                  !

                  !

                  interface fxs 0/1

                    shutdown

                  !

                  interface fxs 0/2

                    shutdown

                  !

                  interface fxs 0/3

                    shutdown

                  !

                  interface fxs 0/4

                    shutdown

                  !

                  !

                  isdn-group 1

                    connect pri 1

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  ip route 0.0.0.0 0.0.0.0 192.168.10.1

                  !

                  no tftp server

                  no tftp server overwrite

                  http server

                  no http secure-server

                  no snmp agent

                  no ip ftp server

                  no ip scp server

                  no ip sntp server

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  sip

                  sip udp 5060

                  no sip tcp

                  !

                  !

                  !

                  voice feature-mode network

                  voice transfer-mode local

                  voice forward-mode local

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  voice trunk T01 type isdn

                    description "pri"

                    resource-selection linear ascending

                    connect isdn-group 1

                    no early-cut-through

                    modem-passthrough

                    t38

                    rtp delay-mode adaptive

                  !

                  voice trunk T10 type sip

                    description "FreePBX"

                    sip-server primary 192.168.10.250

                    registrar primary 192.168.10.250

                    grammar from host local

                    authentication username "ta908" password "123456789"

                    transfer-mode network

                  !

                  !

                  voice grouped-trunk 10

                    description "sip pbx"

                    trunk T10

                    accept $ cost 0

                  !

                  !

                  voice grouped-trunk 1

                    description "pri mitel"

                    trunk T01

                    accept $ cost 0

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  sip authenticate

                  !

                  !

                  sip default-call-routing switchboard

                  !

                  sip registrar

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  !

                  ip rtp symmetric-filter

                  !

                  !

                  !

                  line con 0

                    login

                  !

                  line telnet 0 4

                    login

                    password adtran

                    no shutdown

                  line ssh 0 4

                    login local-userlist

                    no shutdown

                  !

                  !

                  !

                  !

                  !

                  end

                    • Re: Looking to add VOIP capability to old PBX
                      jimtgs New Member

                      I addded the dial plan entries and now I can make calls from each side.  From PBX to IP PBX calls are Anonymous. Most like PRI Card setting.

                       

                      Still don't know why the sip trunk registration doesn't show?? It works.

                       

                      I am most likely going to switch to T1 RBS since we have that port readily available.