Archive for the ‘System Configuration’ Category

With the following setup you can simulate SLA (shared line appearance) on the Positron G-124 IP PBX using the Yealink T26 or T28 IP-Phone.

The following is going to assume that the default settings in “PBX Settings” were not changed and you have successfully added at least two phones to the system.

Do the following on all phones:

1) In User Templates make sure “Enable BLF” is checked (apply config if it was changed)
2) In Phone Provisioning click “Configure” on the the phone you want to setup.
3) On Key 1 configure the following: Type: BLF, Line: 1, Value: 701, Pickup Value: *7701
4) Do the same as above on keys 2-4 except increment 701 in both values by one each time (702, 703, etc)
5) Next on Key 10 configure as follows: Type: Call Park, Line: 1, Value: 700
6) Save the settings and reboot the phone

The steps above setup four “Shared Lines” but you can add or subtract the number in step 4.

When speaking on the phone press Key 10 (“Hold”), Key 1 will change to red on all the phones.
Now the call can be “picked up” by any extension in the office.

Many times when setting up the Adtran Total Access 900 Series VoIP Gateway the customer will ask me to force a specific Caller ID for 911.
This way even if the PBX sets a different Caller ID when the emergency call goes out the call will still be routed to the local PSAP.

At first I configured it as follows:
voice trunk T01 type sip
match dnis “911” replace ani “2125551212” name “Emergency”

This worked fine until one customer had a their PBX misconfigured by their vendor and certain calls had no Caller ID.
When this happened and 911 was called the Caller ID was not set with the above rule.
It seems that the Gateway only manipulates the SIP Header when there is value to replace; in this case since there was none it skipped the rule.

I found settings below that can fix this.
For 911 calls only you can use the following:
voice trunk T01 type sip
caller-id-override emergency-outbound 2125551212

The setting “emergency-outbound” is hard coded for 911 calls. If your area has a different emergency number or you want to override none 911 calls with no Caller ID you need to use the if-no-cpn option:
voice trunk T01 type sip
caller-id-override number-inbound 2125551212 if-no-cpn
match dnis “911” replace ani “2125551212” name “Emergency”

Recently a customer who installed a new Asterisk based PBX had the following issue with their door phone.

They hooked up their existing analog door phone to the FXO port on the PBX and had it route to a ring group.
When someone rings the bell the system sees it as an incoming call and rings the phones in the ring group. The person that answers the phone can then “buzz” the person into the office using their keypad.

The problem was that it took about five seconds before the phones in the ring group would ring.

After some research I figured out that the issue was that the system was waiting for caller id which was not supplied by the door phone. After three to four seconds it timed out and continued.

The fix was very simple; in the zapta.conf set usecallerid=no and the delay is minimal.

Here is a great post by Mark Holloway that lists the most commonly used debug commands for the Adtran Total Access:

http://www.markholloway.com/blog/?p=315

Debug Output

Command Description
show debugging shows current enabled debug commands
debug sip stack messages shows sip debug information
debug isdn l2-formatted shows isdn debug information (easier use)
debug voice verbose shows all voice messaging
debug isdn verbose shows all isdn messaging
debug interface T1 0/4 rbs shows robbed bit signaling from CAS circuit
undebug all turns off all debugging

Utility Commands

Command Description
clear count media counters on media-gateway reset
clear counters clears counters on interfaces
no events turns off messages to the current session
ping ping an IP address
traceroute trace the router path to a destination IP address
wall sends a broadcast message to all current users
sip trunk-registration force-register forces sip re-registration

Show Commands

Command Description
show sip trunk-registration shows registration status
show media-gateway summary shows media gateway channel status
show media-gateway channel 0/2.1 shows media gateway channel details
show qos map shows qos map
show flash lists contents of the flash
show run voice [verbose] lists running configuration for voice provisioning
show hosts shows the hosts table with timers
show event-history shows events and terminal commands typed
show interfaces shows interface details
show system shows uptime, firmware version and timing
show thresholds shows thresholds of T1 interfaces
show users shows current users
show voice quality-stats shows call information including lost packets
show voice quality-stats active shows call information including lost
show voice quality-stats active realtime shows call information including lost
show ip traffic TCP and UDP statistics

As with most enterprise equipment the web GUI is usually either incomplete or full of bugs. (I got locked out of gateway once when trying to set the password through the web GUI)

In the Adtran Total Access web GUI you can enabled T38 faxing by checking the T38 check box under the T1 trunk.

While this does enable T38 it does not allow for the full configuration.

Here is a list of all the T38 options available through the terminal:

  • modem-passthrough
  • t38
  • t38 error-correction [fec | redundancy]
  • t38 redundancy [high-speed | low-speed] <value>
  • t38 max-rate <value>
  • t38 fallback-mode g711
  • t38 max-buffer <value>
  • t38 max-datagram <value>

Samples config:

(config)#voice trunk t02
(config-T02)#modem-passthrough
(config-T02)#t38
(config-T02)#t38 redundancy high-speed 3
(config-T02)#t38 redundancy low-speed 2
(config-T02)#t38 fallback-mode g711
(config-T02)#t38 max-rate 9600
(config-T02)#exit