does not traverse the WAN or PSTN
Occurs between two tels on same data ntwk
Occurs when a user dials an access code (9) to connect to the PSTN
FXO. E1 or T1.
Auto connects one tel to another when the first tel goes off-hook
can work btw any type of signaling inc, E&M, FXO, FXS or any combo of ana and digi interfaces.
Originates at one PBX and terminates at another
form of toll-bypass
-Intercluster trunk calls
occurs when calls are routed by two separate CUCMs
when calls originates on an internal ntwk and are routed to an external ntwk (PSTN)
Local call- One staff member calls another staff member at the same office. The
call is switched between two ports on the same voice-enabled router.
On-net call -One staff member calls another staff member at a remote office. The
call is sent from the local voice-enabled router, across the IP network,
and is terminated on the remote office voice-enabled router.
Off-net call - A staff member calls a client who is located in the same city. The call is sent from the local voice-enabled router, which acts as a gateway, to
the PSTN. The call is then sent to the PSTN for call termination.
PLAR call - A client picks up a customer service telephone located in the lobby of
an office and is automatically connected to a customer service representative
without dialing any digits. The call is automatically dialed based on the PLAR configuration of the voice port. In this case, as soon as the handset goes off hook, the voice-enabled router generates the prespecified digits to place the call.
PBX-to-PBX call - One staff member calls another staff member at a remote office. The
call is sent from the local PBX, through a voice-enabled router, across
the IP network, through the remote voice-enabled router, and terminated
on the remote office PBX.
Intercluster trunk call - One staff member calls another staff member at a remote office using IP phones. The call setup is handled by a Cisco Unified
Communications Manager server at each location. After the call is set
up, the IP phones generate IP packets carrying voice between sites.
On-net to off-net call - One staff member calls another staff member at a remote office while the IP network is congested. When the originating voice-enabled
router determines that it cannot complete the call across the IP network,
it sends the call to the PSTN with the appropriate dialed digits
to terminate the call at the remote office via the PSTN network.
Three types of Analog Voice Interfaces supported by Cisco devices:
connects directly to end-user equip - tels, fx, mods
used for trunk or tie line connection to a PSTN or PBX that does not support E&M signaling.
preferable to above options as it delivers better answer and disconnect supervision.
E-lead and M-lead
ANALOG SIGNALING: TECHNIQUES
detection of changes to the status of a loop or trunk
no way to prevent GLARE - CO and subscriber seize same line, same time - use ground-start to prevent glare.
does not provide switch-side disconnect supervision for FXO calls.
passing dialed digits - tone or pulse - to PBX or CO. These digits provide path to CPE
audible tones to use (incoming calls etc)
-call progress tones
mainly used btw PBXs or network to network telephony switches.
Types I, II, III, V supported
SSDC5 mainly found in England
E&M ADDRESS SIGNALING:
3 E&M access types:
suffers from glare
The calling side seizes the line by going off-hook
on its E lead, waits for a minimum of 150 ms and then sends address information as
DTMF digits or as dialed pulses. This signaling approach is used for E&M tie trunk
most commonly used
In wink-start, the calling
side seizes the line by going off-hook on its E lead; it then waits for a short temporary off-hook pulse, or “wink,” from the other end on its M lead before sending
address information as DTMF digits. The switch interprets the pulse as an indication
to proceed and then sends the dialed digits as DTMF or dialed pulses. This signaling
is used for E&M tie trunk interfaces. This is the default setting for E&M voice ports.
the calling station seizes the line by going off-hook on its E lead. After a timed interval, the calling side looks at the status of the called side. If the called side is on-hook, the calling side starts sending information as DTMF digits. Otherwise, the calling side waits until thecalled side goes on-hook and then starts sending address information. This signaling approach is used for E&M tie trunk interfaces.
ANALOG VOICE PORTS CONFIG:
Are used to interconnect GWs or PBX to other GWs.
can be analog or digital
Signaling can be done using either the voice channel (in-band) or an extra dedicated channel (outof-band). The available features depend on the signaling protocol in use between the devices.
Centralized Automated Message Accounting (CAMA)
Five options exist:
■ KP-0-NXX-XXXX-ST: 7-digit ANI transmission. The Numbering Plan Area (NPA),
or area code, is implied by the trunk group and is not transmitted.
■ KP-0-NPA-NXX-XXXX-ST: 10-digit transmission. The E.164 number is fully
■ KP-0-NPA-NXX-XXXX-ST-KP-YYY-YYY-YYYY-ST: Supports CAMA signaling with
ANI/Pseudo ANI (PANI).
■ KP-2-ST: Default transmission when the CAMA trunk cannot get a corresponding
Numbering Plan Digit (NPD) in the look-up table or when the calling number is
fewer than 10 digits. (NPA digits are not available.)
■ KP-NPD-NXX-XXXX-ST: 8-digit ANI transmission, where the NPD is a single MF
digit that is expanded into the NPA. The NPD table is preprogrammed in the sending
and receiving equipment (on each end of the MF trunk). For example: 0=415, 1=510,
05551234 = (415) 555-1234, 15551234 = (510) 555-1234
The NPD value range is 0–3.
When you use the NPD format, the area code needs to be associated with a single digit.
You can preprogram the NPA into a single MF digit using the ani mapping voice port
command. The number of NPDs programmed is determined by local policy as well as by
the number of NPAs the PSAP serves. Repeat this command until all NPDs are configured
or until the NPD maximum range is reached.