IP Based 'Trunk-side' - CommPartners Connect

SIP DS0 Service 

IP Based ‘Trunk-side’ Replacement 

Nationwide SIP Origination and Termination 

Adtran Total Access 912/924 – VoIP Gateway 

The scope of this document is to provide our partners who are deploying the “SIP DS0” service a guide to 

properly install, configure, turn-up and troubleshoot the Adtran Total Access 912/924 VoIP Gateway. This 

document will be broken down into five sections according to the process that is involved when deploying 

the SIP DS0 service from Commpartners. 

Deployment Scenario #1 

In the deployment scenario below, the customer can have the ability to replace the ISDN-PRI trunks that are 

currently being delivered from the incumbent telephone operator and substitute the circuits with a SIP trunk 

from Commpartners. The customer would also have the ability to replace any existing analog POTS trunks 

as the TA912 has the capability to provide SIP DS0 service. The network connectivity in this deployment 

scenario allows the customer to utilize their existing broadband internet access to deploy VoIP services with 

the Adtran TA912. 

Deployment Scenario #2 

In the second deployment scenario below, the customer would still have the ability to replace the ISDN-PRI 

trunks that are currently being delivered from the incumbent telephone operator and substitute the circuits 

with a SIP trunk from Commpartners. The customer would also have the ability to replace any existing 

analog POTS trunks as the TA912 has the capability for our partners to drop to a T1 WAN circuit directly 

to the customer premise to deploy VoIP services with the Adtran TA912.

I. Installation of the Adtran TA912/924 VoIP Gateway 

Site Requirements: 

(1) Adtran TA912/924 VoIP gateway 

(1) Adtran TA912/924 Software/Firmware Version 10.02.00 

(1) Open PRI-ISDN Interface on existing legacy PBX switch 

(1) 10/100 Ethernet port to Broadband Router or T1 WAN Circuit 

(1) 1RU of free space on existing 19”network rack installation 

(1) 120V-AC Power Connection 

Front View: 

Rear View:

II. Configuration Wizard Setup through HTTP Interface 

Once the user has logged into the TA912/924 through the serial interface, the TA can be easily setup 

through the HTTP server interface which will allow the user to simply fill in the required configuration 

fields with the “point-and-click” interface. First the user will need to enable the HTTP service and setup the 

IP address of the Ethernet port: 

AOS Configuration (CLI) 

User Access Login 

Password: 

TA912>conf t 

TA912>en 

Password: 

TA912#conf t 

TA912(config)#ip http server 

TA912(config)#login 

TA912(config)#username admin password adtran 

TA912(config)# TA912(config)#interface ethernet 0/1 

TA912(config-eth 0/1)# ip address 24.120.226.51 255.255.255.128 

TA912(config-eth 0/1)# media-gateway ip primary 

TA912(config-eth 0/1)#no shut 

TA912(config-eth 0/1)#CTRL+Z 

TA912(config)# ip route 0.0.0.0 0.0.0.0 24.120.226.1 

TA912(config)#copy run star 

Building configuration... 

Done. Success! 

TA912# 

IAD MANAGEMENT: 

enable 

SET THE TIME CLOCK: 

clock set 16:40:00 3 june 2005 

clock timezone –8 

NAME THE IAD: 

conf t 

hostname “TA912” 

PASSWORD PROTECT THE ENABLE MODE: 

enable password adtran 

PASSWORD PROTECT TELNET ACCESS: 

line telnet 0 

login 

password adtran 

PUT AN IP ADDRESS ON THE ETHERNET PORT: 

interface e 0/1 

ip address x.x.x.x 255.255.255.0 

no shutdown

Once the user has enabled the HTTP interface , setup a valid username and password and associated 

IP address for the Ethernet Interface, then the TA912/924 can now be setup over the HTTP interface. 

Proceed to open your internet web browser and type in the IP address of the TA912/924 Ethernet 

address into the address field. You will then be prompted for the user name and password: 

Once the user has entered the username and password, then we can begin the configuration wizard: 

Proceed to click on the “Config Wizard”. This will allow the user to configure the TA912/924 with 

the basic FXS basic port configuration with the Ethernet WAN or T1 WAN interfaces. For 

deployments with the Legacy PBX integration for the PRI interface, please see section #4. For 

deployments with the BroadSoft DSX-1 integration, please refer to Section #6.

The first prompt will ask the user to fill in the host name of the system along with other basic system 

parameters. Be sure to fill in the DNS server field along with the type of Local Dialing Plan to be 

used: 

Now specify which type of WAN Access will be used. Either T1 or Ethernet can be utilized.

If the T1 WAN access is to be used, then specify whether a DHCP served address will be used. If a 

static IP address is going to be used, then proceed to fill in the IP address and Subnet Mask 

information. 

Confirm that the VoIP Parameters are set to the default values as shown below:

Configuration parameters for the SIP trunk with the Commpartners network are included below: 

Begin the configuration for each individual FXS analog port as seen below:

Verify that all of the configuration settings are correct and proceed to click on the “finish” tab to 

confirm: 

If the admin would like to configure additional parameters then proceed to click on the “System” tab: 

The “Passwords” section will allow the administrator to assign additional user accounts and 

passwords:

The “IP Services” section will allow the admin to open or close the HTTP server ports as seen below: 

The admin can also allow for a DHCP host server for LAN clients as well:

Configuration of a Primary DNS server is performed in the “Hostname/DNS” fields below: 

The ability to Add/Modify/Delete user accounts is performed below under the “Stations” settings:

If the configuration wizard was used for the initial setup, then these fields should already be set: 

The SIP Dial Plan can be manually configured below to control the dial plan parameters:

Additional dial plan parameters can be modified to allow or deny certain call types for station users: 

The admin can also setup dial plan templates to allow/deny call types for different groups if needed:

Be sure to create the codec order so that “G.729” is the primary voice codec while “G.711” is selected 

at the fallback codec for faxing (T.38 FoIP to be support in Q1 2006): 

The admin shouldn’t require any changes in this section but verification of the “FlashHook” 

parameters is recommended as seen below:

Verify that the “SIP Registrar” services are enabled as seen below: 

Setup the “Default Gateway” if required when using an external broadband router for the WAN:

Enable “IP Routing” when utilizing an external broadband router for the WAN: 

Configuration of the IP Address for the Ethernet port is required if the admin has not configured the 

port already in the configuration wizard: 

The system configuration can be saved to an external configuration file and the admin will have the 

ability to restore the configuration if needed:

The ability to upgrade the system firmware is performed in the system configuration as well: 

III. AOS (Adtran Operating System) Provisioning 

If the admin would like to provision the TA912/924 directly through the CLI interface then the CLI 

examples below contain the proper syntax for configuring the system: 

LAYER 1 CONFIGURATIONS: 

en 

conf t 

T1 PROVISIONING: 

clock source t1 0/1 

clock source internal secondary 

interface t1 0/1 

framing esf 

coding b8zs 

lbo long 0 

tdm-group 1 timeslots 1-24 speed 64 

no shutdown 


framing esf 

coding b8zs 

lbo short 15 

tdm-group 2 timeslots 1-8 speed 64 

no shutdown

FXS PROVISIONING: 

interface fxs 0/1 

signal loop-start 

impedance 600r 

ring-voltage 50 

tx-gain –6.0 

rx-gain –3.0 

no shutdown 







no shutdown 







no shutdown 







no shutdown 


en 

conf t 

PPP PROVISIONING: 

interface ppp 1 

cross-connect 1 t1 0/1 1 ppp 1 

no shutdown 


en 

conf t 

ETHERNET IP PROVISIONING: (Already done in the Management section above) 

interface e 0/1 

ip address 66.251.103.1 255.255.255.224 

no shutdown 

T1 / PPP IP PROVISIONING: 


ip address 66.251.102.142 255.255.255.252 

IP ROUTING PROVISIONING: 

ip routing 

ip route 0.0.0.0 0.0.0.0 66.251.102.141

NAT: 

en 

conf t 

ip firewall 

ip access-list standard TRUSTED 

permit any 

exit 

ip policy-class PRIVATE 

nat source list TRUSTED address 66.251.102.142 overload 

interface eth 0/1 

access-policy PRIVATE 

PAT: 

en 

conf t 

ip access-list extended UNTRUSTED 

permit tcp any host 66.251.102.142 eq 80 

exit 

ip policy-class PUBLIC 

nat destination list UNTRUSTED address 66.251.103.10 


access-policy PUBLIC 

SIP VOICE CONFIGURATIONS: (MATCH THE PROVIDER’S NETWORK) 

en 

conf t 

ip sip 

ip sip qos dscp 46 

ip rtp qos dscp 46 


media-gateway ip primary 

exit 

voice feature-mode network 

voice flashhook mode transparent 

DEFINE NUMBER COMPLETE TEMPLATES FOR LOCAL 10-DIGIT PSTN CALLS: 

voice dial-plan 1 always-permitted 1-nxx-nxx-xxxx 

DEFINE VOICE CODECS: 

voice codec-list VOICE 

codec g729 

codec g711ulaw 

default 

voice codec-list FAXMODEM 

codec g711ulaw 

IV. PBX Trunk Side Replacement (Stripped POTS) 

This section provides the admin with the AOS commands to provision the TA912/924 with the ISDN PRI 

or T1 CAS PBX Trunk Side Replacement. The intent is to show the admin an example for how the 

configuration would be performed using the CLI as a reference: 

DSX-1 Options: 

Separate DSX-1 DS0s each with their own phone number and own trunk group. This configuration will 

only allow that one PBX phone number to use that one DSX DS0 for inbound calls. For outbound calls, the 

PBX will have to be configured for that phone extension to always grab the same DS0. This way the 

correct Caller ID number will be sent to the called party.


framing esf 

coding b8zs 

lbo short 15 

tdm-group 11 timeslots 1 speed 64 



no shutdown 

! 

! 

! 

voice user 8188271681 

description “PBX” 

sip-identity 8188271681 T11 register 

codec-group VOICE 

exit 

voice trunk T11 type t1-rbs supervision wink role network 

description “DSX-1” 

no reject-external 

caller-id-override number 8188271681 

did digits-transferred 4 


connect t1 0/2 tdm-group 11 

exit 

voice grouped-trunk DSX-DS0-1 

trunk T11 

accept 8188271681 

exit 

! 

! 

! 





exit 








exit 


trunk T12 

accept 8188271682 

exit 

! 

! 

! 








exit


trunk T13 

accept 8188271683 

exit 





exit 

V. BroadSoft Application (Class V POTS) 

The Adtran TA912/924 can be configured as a gateway within BroadSoft which will provide the full Class 

V feature sets for the 24 analog FXS ports as individual BroadSoft seats. Below are the AOS commands 

and BroadSoft screen captures to enable this type of functionality: 

NETWORK T1 SIP TRUNKING TO BROADSOFT: 

voice trunk T01 type sip 

description “SIP” 


sip-server primary 63.123.133.30 

registrar primary 63.123.133.30 


exit 

voice grouped-trunk SIP 

no description 

trunk T01 

accept $ cost 0 

ENABLE SIP FOR ANALOG PHONES: 


description “Analog Phone” 

connect fxs 0/1 

sip-identity 1688 T01 register auth-name 1688 password adtran 



description “Analog Phone” 





description “Fax Machine” 



codec-group FAXMODEM 


description “Dialout Modem” 


sip-identity 8188271691 T01 register auth-name 8188271691 password adtran codec-group 

FAXMODEM 

exit 

ENABLE SIP FOR DSX-1 INTERFACE: 




caller-id-override number 8188271682 (This number is associated with the Broadsoft user profile, 

if it does not match a user, then get fast busy from Broadsoft)

did digits-transferred 4 (Number of DID digits passed to the DSX-1 port for the PBX to route the 

call to a phone.) 



exit 

voice grouped-trunk DSX 

trunk T02 

accept 818827168[2,3] cost 0 (The inbound phone numbers will be matched to the accept number 

and pattern, the call will then be routed to the DSX-1 port by the switchboard feature of the TA916.) 

exit 

voice user 8188271682 (This allows the DSX-1 phone number to be registered on the Broadsoft) 




voice user 8188271683 (This allows the DSX-1 phone number to be registered on the Broadsoft) 




(The Broadsoft needs a RESOURCE profile with the WAN IP address of the TA916. The Broadsoft USER 

profiles will then reference the device with that RESOURCE profile. The TA916 should not have any 

VOICE USER profiles defined for the DSX-1 phone numbers.) 

ENABLE SIP FOR IP PHONES: 

en 

conf t 

ip sip registrar 

ip rtp firewall-transversal (Only if using Private IPs on the Ethernet port) 

interface ethernet 0/1 

media-gateway ip primary 

exit 

voice user 8188271692 (The 8188271692 example must match the IP phone’s phone number or username) 

description “SIP Phone” 

connect sip 

sip-identity 8188271692 T01 register auth-name 8188271692 password adtran (Sending to 

Broadsoft. If this command is left out, the SIP phone registration will pass through the TA900 and register 

with the Broadsoft. 

Refer to: voice trunk T01 type sip 

sip-server primary 66.251.102.20 

registrar primary 66.251.102.20 

The proxy of the SIP phone must be the TA900 Ethernet port IP address. The “voice user 8188271692” 

and “sip-identity 8188271692” need to have the same 8188271692 value in this example. 

Examples: 

adtran) 

SIP identity defined, but user not registered in the TA900: 


sip-identity 8188271692 T01 

SIP identity defined, user is registered in the TA900 without authentication: 



SIP identity defined, user is registered in the TA900 with authentication: 


sip-identity 8188271692 T01 register authentication-name brian password 

exit

DSX Resource Configuration 

Analog Resource Configuration / SIP Phone Configuration

Analog User – Station Configuration 

DSX User Configuration

VI. Troubleshooting the Adtran TA912/24 

SIP Phone - User Configuration 

The admin can find the appropriate debug commands below in case an issue arises with the voice service 

trunks. In addition, any issues that cannot be resolved by our partners and resellers directly can direct you’re 

inquires to: SE@commpartners.us 

TROUBLESHOOTING COMMANDS: 

Show voice 

Ani 

Available 

Current-mode 

Dial-plan 

Did 

Directory 

Extensions (shows idle or busy) 

Grouped-trunk 

Ring-group 

Service-mode 

Spre 

Switchboard 

Trunk 

Connects (shows what virtual trunk T02 is connected to what interface T1 0/2) 

Users (shows analog users) 

Show sip 

Location 

Resources 

Statistics 

Trunk-registration (shows if a fxs voice user is registered) 

User-registration (shows if a SIP phone user is registered)

Show media-gateway 

Channel 

Info 

Session 

Summary 

Active (shows active RTP sessions) 

Show qos map 

Show rtp resources 

Debug 

Persistence-manager 

Sip 

Cldu 

Location 

Manager 

Registrar 

Registration 

Stack 

Debug 

Errors 

Exceptions 

Info 

Messages 

Verbose 

Warnings 

Voice 

Debug sip stack messages (This is the best sip debug command)

IP Based 'Trunk-side' - CommPartners Connect

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