Configuring Adtran SBC to use with Mitel 250 & Telnet SIP Trunks

[DialToneDreams]

Hello all. 15 year phone guy but new to Mitel....    This Adtran setup is a little more "IT" (command line) than I usually get involved in but I made a few commands work but there is so much that I don't know.

I have 1 public IP for the Adtran and a Destination Telnet SIP server destination public IP. I believe I either need to setup a 1to1 NAT or a 1 to Many NAT (in case I need a port forward in lieu of a Border Gateway).  Is it as simple as that? No other SIP configs needed on Adtran? Any tips or gotchas I should lookout for? Got any commands you know that work you're willing to part with?

Thanks for any help!

[johnp]

I wpuld get to the Adtran forum, there should be some info that helps. My thought was that the Adtran does sip to Telnetww, and Mitel does sip to Adtran. You haven't mentioned what system. A SBC should provide a level of security, and you can access it via a webpage that may help with this setup

[DialToneDreams]

Oops...     Adtran NetVanta 3140. I did find the GUI. I could try the Adtran forums but figured I'd try here in case someone had a Mitel specific experience to share.

[acejavelin]

We use the Netvanta 3140 and Adtran TA908's... A LOT... we are not only a Mitel dealer, but a hosted SIP service provider as well and the Adtran is our go to for SIP trunking SBC.

I can possibly give you some sample setup configs, but here is a real general idea... You need a public IP address direct to the Adtran with no firewall or router ahead of it, and a LAN IP on the voice VLAN, both static and dedicated to the Adtran. Then you make two trunk groups, one for the connection to your SIP service provider and one to the Mitel, then in your digit acceptance tables you put the DID's so they are allowed with no cost to the PBX trunk group and all others go to the SIP provider trunk group. This is a VERY simplified overview. We never use the GUI except to load a configuration file and diagnostics, or make a quick tweak on a unit in service already. We basically have two templates, one for "dumb" PBX's which don't handle PAI, BTN, and redirect headers properly (most phone systems, including the MiVO 250), and "smart" PBX's which do handle them correctly like the 3300 and most Asterisk boxes.

Oh yeah, and make sure you have a Netvanta 3140 with a SBC session license of sufficient size, they come in 5, 10, 25, 50, and larger increments. Without the SBC Session license you will never get anywhere. You need one that is a part number like 4700341F2#10 which is a NetVanta 3140 10 SESSION SBC unit.

And you don't need an MBG, but if you have one you most likely don't need the Netvanta either as the MBG handles everything for you.

[DialToneDreams]

Thanks for the tips ace... it is greatly appreciated. I talked to the SIP carrier, Mitel and Adtran today and only inched closer to success. I gotta return to the site tomorrow. I think the SIP carrier told me wrong... I got Adtran to help me but pursued the NAT route like the carrier said. So where I stand now is no SIP traffic hitting the Adtran from outside and no SIP traffic hitting the Adtran from the PBX. My Adtran can now ping the outside world though... yay!

I guess I'll have to figure out those trunks. I did see a few commands online but none describing the entering of DID's. I'm starting to miss the simplicity of SIP on Toshiba 

Many thanks for pointing me in the right direction! If you get bored and have a sample config I will totally ask the universe to reward your generosity!

 
I do have an Adtran SBC 10 lisc and my Topology is as you say I will need:   Comcast carrier router  <---> public IP for Adtran on GIG0/2  ~~~~ private LAN IP for Adtran on GIG0/1   <---> Switch <---> Mitel 250 w private LAN IP

[acejavelin]

Thanks for the tips ace... it is greatly appreciated. I talked to the SIP carrier, Mitel and Adtran today and only inched closer to success. I gotta return to the site tomorrow. I think the SIP carrier told me wrong... I got Adtran to help me but pursued the NAT route like the carrier said. So where I stand now is no SIP traffic hitting the Adtran from outside and no SIP traffic hitting the Adtran from the PBX. My Adtran can now ping the outside world though... yay!

I guess I'll have to figure out those trunks. I did see a few commands online but none describing the entering of DID's. I'm starting to miss the simplicity of SIP on Toshiba 

Many thanks for pointing me in the right direction! If you get bored and have a sample config I will totally ask the universe to reward your generosity!

 
I do have an Adtran SBC 10 lisc and my Topology is as you say I will need:   Comcast carrier router  <---> public IP for Adtran on GIG0/2  ~~~~ private LAN IP for Adtran on GIG0/1   <---> Switch <---> Mitel 250 w private LAN IP

Here is a sample configuration... I cleaned it up, used "***" to hide sensitive information and changed numbers and domain names to clean it up, but I have to very careful as not go give away sensitive information. Not that "carrier.com" does not exist, but is an example. Hope it helps, it should give you some idea of what I am talking about...

Code: [Select]

!
!
! ADTRAN, Inc. OS version R12.3.4
! Boot ROM version R11.5.0
! Platform: NetVanta 3140, part number 4700341F2#25
! Serial number CFG***
!
!
hostname "NV3140"
enable password encrypted 2***9
!
!
clock timezone -6-Central-Time
!
ip subnet-zero
ip classless
ip default-gateway <PUBLIC IP ADDRESS GATEWAY>
ip routing
ipv6 unicast-routing
!
!
name-server 8.8.8.8 4.2.2.2
!
!
no auto-config
auto-config authname adtran encrypted password 1***c
!
event-history on
no logging forwarding
no logging email
!
service password-encryption
!
username "Admin" password encrypted "1***c"
!
banner motd #
     *** Important *** 
       
This Managed VoIP SIP Session Border Controller is Provided by
*** for use with *** Managed VoIP Services.

Please contact *** for any configuration changes.

The gig 0/1 interface is enabled with an address of 10.1.1.253
The gig 0/2 interface is enabled with an address of <PUBLIC IP ADDRESS>
#
!
ip policy-timeout udp 5060 600
!
ip firewall
no ip firewall alg msn
no ip firewall alg mszone
no ip firewall alg h323
!
!
no dot11ap access-point-control
!
!
ip flow top-talkers
!
interface gigabit-eth 0/1
  description Local LAN
  ip address  10.1.1.253  255.255.0.0
  ip access-group LAN in
  ip flow ingress
  no shutdown
  media-gateway ip primary
!
!
interface gigabit-eth 0/2
  description WAN
  ip address  <PUBLIC IP ADDRESS> <PUBLIC IP SUBNET MASK>
  ip access-group "CARRIER ACCESS" in
  ip flow ingress
  no shutdown
  media-gateway ip primary
!
!
interface gigabit-eth 0/3
  no ip address
  shutdown
!
!
ip access-list standard "CARRIER ACCESS"
  permit 24.***.***.240 0.0.0.15 log
  permit 64.***.***.0 0.0.0.255 log
  permit host 8.8.8.8 log
  permit host 4.2.2.2 log
  permit hostname time.carrier.com
  deny   any log
!
ip access-list standard LAN
  permit 10.0.0.0 0.255.255.255 log
  permit 192.168.0.0 0.0.255.255 log
  permit 172.16.0.0 0.15.255.255 log
!
!
ip route 10.1.0.0 255.255.0.0 10.1.2.1
!
no tftp server
no tftp server overwrite
http server
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 forward-mode network
!
!
voice dial-plan 1 local NXX-XXXX
voice dial-plan 2 local NXX-NXX-XXXX
!
!
voice codec-list TRUNKS
  codec g711ulaw
!
!
voice ani-list DIDs
  ani 9994150051
  ani 9994150052
  ani 9994150053
  ani 9994150225
  ani 9994150226
  ani 9994150227
  ani 9994150228
  ani 9994150229
  ani 9994150230
  ani 9994150231
!
voice trunk-list PBX_Trunks
  trunk T03
!
voice trunk-list CARRIER_Trunks
  trunk T01
  trunk T02
!
!
voice trunk T01 type sip
  description "CARRIER SIP TRUNK"
  sip-server primary voice.carrier.com
  registrar primary voice.carrier.com
  outbound-proxy primary voice.carrier.com
  domain "voice.carrier.com"
  register 9994150051 auth-name "9994150051" password encrypted "3***f"
  register range 9994150052 9994150053
  register range 9994150225 9994150231
  trust-domain
  codec-list TRUNKS both
  authentication username "9994150051" password encrypted "2***a"
!
voice trunk T02 type sip
  description "CARRIER DIVERSION"
  match ani "$" add diversion "9994150051" unconditional no off
  sip-server primary voice.carrier.com
  registrar primary voice.carrier.com
  outbound-proxy primary voice.carrier.com
  domain "voice.carrier.com"
  trust-domain
  codec-list TRUNKS both
  authentication username "9994150051" password encrypted "2***d"
!
voice trunk T03 type sip
  description "LOCAL SIP LINK"
  sip-server primary 10.1.1.132
  codec-list TRUNKS both
!
!
voice grouped-trunk CARRIERSIP
  description "CARRIER SIP Trunks"
  trunk T01
  accept $ cost 0
  reject 976-XXXX
  reject 1-900-NXX-XXXX
  reject 1-976-NXX-XXXX
  reject 1-NXX-976-XXXX
  deny list CARRIER_Trunks
  permit list DIDs
  !deny all other trunks
  !deny all other ani
!
!
voice grouped-trunk SIP_DIVERSION
  description "CARRIER Diversion Link"
  trunk T02
  accept $ cost 0
  reject 976-XXXX
  reject 1-900-NXX-XXXX
  reject 1-976-NXX-XXXX
  reject 1-NXX-976-XXXX
  deny list DIDs
  permit list PBX_Trunks
  !deny all other trunks
  !deny all other ani
!
!
voice grouped-trunk PBXSIP
  description "PBX SIP trunks"
  trunk T03
  accept 999-415-005X cost 0
  accept 999-415-022X cost 0
  accept 999-415-023X cost 0
!
!
ip rtp nat-session timeout 300
!
!
line con 0
  login
  password encrypted 3***3
!
line telnet 0 4
  login
  password encrypted 2***7
  no shutdown
line ssh 0 4
  login local-userlist
  no shutdown
!
sntp server time.carrier.com version 3
sntp wait-time 3600
!
!
end

[johnp]

What does the diversion do? how does that work?...thanks

[acejavelin]

What does the diversion do? how does that work?...thanks

It's a way to trick a carrier switch... Many SIP carriers don't allow you to send an originating number that is not a DID number, unless it is a "diversion" such as a call forward or something like that... Some phone systems (including the MiVO) don't send a proper SIP Diversion message in the Invite, this is a work-around. Basically if the an originating number is sent by the phone system that is a DID assigned to the Adtran, it just completed the call normally... If a different originating number is sent that is NOT a number on that service, the Adtran inserts an unconditional diversion message into the Invite message. Essentially a way of inserting a "billing number" into the message so any originating caller ID can be sent, or an original caller's number can be passed through to the destination of a forwarded or twinned call.

Does that make sense, or have had a few to many Two Hearted Ale's this evening already to be answering this question?

[acejavelin]

Here would be a more simplified version, without the "diversion" trick... This would be what we would use for a MiVoice Business or similar "smart" PBX that handles BTN, PAI, and diversion properly. This would vary based on your carrier's needs as to which one is needed, if your carrier allows ANY originating caller ID to be sent, this might work for any PBX but most carriers where I am will only accept an originating Caller ID if it is a number they are hosting for that customer.

Code: [Select]

!
!
! ADTRAN, Inc. OS version R12.3.3
! Boot ROM version R11.5.0
! Platform: NetVanta 3140, part number 4700341F2#10
! Serial number CFG1596969
!
!
hostname "NV3140"
enable password encrypted 2***9
!
!
clock timezone -6-Central-Time
!
ip subnet-zero
ip classless
ip default-gateway <PUBLIC IP GATEWAY>
ip routing
ipv6 unicast-routing
!
!
name-server 8.8.8.8 4.2.2.2
!
!
no auto-config
auto-config authname adtran encrypted password 1***c
!
event-history on
no logging forwarding
no logging email
!
service password-encryption
!
username "Admin" password encrypted "1***c"
!
banner motd #
     *** Important *** 
       
This Managed VoIP SIP Session Border Controller is Provided by
*** for use with *** Managed VoIP Services.

Please contact *** NPA-NNX-XXXX for any configuration changes.

The gig 0/1 interface is enabled with an address of 10.10.50.30
The gig 0/2 interface is enabled with an address of <PUBLIC IP ADDRESS>
#
!
ip policy-timeout udp 5060 600
!
ip firewall
no ip firewall alg msn
no ip firewall alg mszone
no ip firewall alg h323
!
!
no dot11ap access-point-control
!
!
ip flow top-talkers
!
interface gigabit-eth 0/1
  description Local LAN
  ip address  10.10.50.30  255.255.255.0
  ip access-group LAN in
  ip flow ingress
  no shutdown
  media-gateway ip primary
!
!
interface gigabit-eth 0/2
  description WAN
  ip address  <PUBLIC IP ADDRESS>  <PUBLIC IP SUBNET>
  ip access-group "CARRIER ACCESS" in
  ip flow ingress
  no shutdown
  media-gateway ip primary
!
!
interface gigabit-eth 0/3
  no ip address
  shutdown
!
!
ip access-list standard "CARRIER ACCESS"
  permit 24.***.***.240 0.0.0.15 log
  permit 64.***.***.0 0.0.0.255 log
  permit host 8.8.8.8 log
  permit host 4.2.2.2 log
  permit hostname time.carrier.com
  deny   any log
!
ip access-list standard LAN
  permit 10.0.0.0 0.255.255.255 log
  permit 192.168.0.0 0.0.255.255 log
  permit 172.16.0.0 0.15.255.255 log
!
!
ip route 10.0.0.0 255.0.0.0 10.10.50.1
!
no tftp server
no tftp server overwrite
http server
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 forward-mode network
!
!
voice dial-plan 1 local NXX-XXXX
voice dial-plan 2 local NXX-NXX-XXXX
!
!
voice codec-list TRUNKS
  codec g711ulaw
!
!
voice trunk T01 type sip
  description "Carrier SIP TRUNK"
  sip-server primary voice.carrier.com
  registrar primary voice.carrier.com
  outbound-proxy primary voice.carrier.com
  domain "voice.carrier.com"
  register 9994751345 auth-name "9994751345" password encrypted "3***"
  register range 9994751346 9994751349
  trust-domain
  codec-list TRUNKS both
  authentication username "9994751345" password encrypted "3***"
!
voice trunk T02 type sip
  description "LOCAL SIP LINK"
  sip-server primary 10.10.50.10
  codec-list TRUNKS both
!
!
voice grouped-trunk carrierSIP
  description "Carrier SIP Trunks"
  trunk T01
  accept $ cost 0
  reject 976-XXXX
  reject 1-900-NXX-XXXX
  reject 1-976-NXX-XXXX
  reject 1-NXX-976-XXXX
!
!
voice grouped-trunk PBXSIP
  description "PBX SIP trunks"
  trunk T02
  accept 999-475-XXXX cost 0
!
!
ip rtp nat-session timeout 300
!
!
line con 0
  login
  password encrypted 3***3
!
line telnet 0 4
  login
  password encrypted 2***7
  no shutdown
line ssh 0 4
  login local-userlist
  no shutdown
!
sntp server time.carrier.com version 3
sntp wait-time 3600
!
!
end


[johnp]

It's a way to trick a carrier switch... Many SIP carriers don't allow you to send an originating number that is not a DID number, unless it is a "diversion" such as a call forward or something like that... Some phone systems (including the MiVO) don't send a proper SIP Diversion message in the Invite, this is a work-around. Basically if the an originating number is sent by the phone system that is a DID assigned to the Adtran, it just completed the call normally... If a different originating number is sent that is NOT a number on that service, the Adtran inserts an unconditional diversion message into the Invite message. Essentially a way of inserting a "billing number" into the message so any originating caller ID can be sent, or an original caller's number can be passed through to the destination of a forwarded or twinned call.

Does that make sense, or have had a few to many Two Hearted Ale's this evening already to be answering this question?

I looked again and see what you are doing. As to the Two Hearted, I myself don't drink many ipa's but this is one I will have now and then. I like Bell's Expedition Stout and have toured the brewery several times.

[NEPhoneGuy]

The two config files you uploaded for the NetVantas are great information.

Adtran has an active and super helpful forum in my experience also for the NetVanta and TA908e/924es