Digital trunks typically offer much greater capacity, quality, and reliability than analog trunks. As such, most midsize and larger businesses use digital service connections to the PSTN. This rise in popularity has also caused the cost of this service to decline over time in most geographies, making its use even more attractive.
In this section, you will learn how to configure, verify, and troubleshoot the four types of digital service that Cisco voice gateways support. These include ISDN PRI, ISDN BRI, E1 R2, and T1 CAS trunk types.
Configuring E1/T1 Physical Layer Connections
All these trunk typeswith the exception of ISDN BRIare delivered on either an E1 or T1 digital circuit. Before you configure voice signaling methods, be sure that the interface is configured so that the circuit is operating properly on the physical signaling level (Layer 1).
E1 or T1 voice circuits are channelized, meaning they transmit information in timeslots or channels. E1 circuits have 30 timeslots available, and T1 circuits have 24 timeslots. The timeslots are carried in frames. The beginning of the frame is marked by a specific bit sequence. A more complete description of the circuit characteristics is described in Chapter 5, "Circuit Options."
At a physical level, the interface must be able to identify the beginning of each frame in the bit stream and understand the frame format that is being used so that it can read the information in the timeslots correctly. This means several things:
For T1, D4 framing is usually associated with AMI line coding, and ESF framing is usually associated with B8ZS line coding. The PSTN service provider can furnish details regarding which framing and line coding mechanism is being used on the circuit that you are installing.
Now you can begin setting up the physical characteristics of the circuit at the T1/E1 controller level. From global configuration mode, enter the controller T1 port/slot or controller E1 port/slot command to enter controller configuration mode.
Configure the framing using the framing type command. For T1, the valid options for type are sf or esf. The default for T1 is SF. For E1, the valid options for type are crc4 and no-crc4. The default for E1 is CRC4.
Configure the line coding using the linecode type command. For T1, the valid options for type are ami and b8zs. The default for T1 is AMI. For E1, the valid options for type are ami and hdb3. The default for E1 is HDB3.
It is not normally necessary to configure the clock source for PSTN connections. The default setting is line clock, which allows the interface to obtain clocking from the circuit. This is typically the correct setting for a PSTN circuit.
Example 6-10 demonstrates configuring the physical interface for an E1 circuit.
[View full width] Shanghai#config t ! ! Enter controller configuration mode Shanghai(config)#controller E1 1/0 ! ! Configure framing Shanghai(config-controller)#framing crc4 ! ! Configure line coding Shanghai(config-controller)#linecode hdb3 *May 12 15:48:28.527: %CONTROLLER-5-UPDOWN: Controller E1 1/0, changed state to up |
After you have configured the proper framing and line coding, the controller comes up.
Verifying and Troubleshooting
You can verify that no physical alarms exist on this circuit by using the show controller T1 slot/port or show controller E1 slot/port command. Both commands show you the current state of the controller, configuration information, and whether any alarms are detected. Both commands also provide error counters for the previous 24 hours in 15-minute intervals.
Example 6-11 demonstrates sample output from the show controller e1 command. You can see in this example that this interface has had errors. Slip seconds have been recorded in the current interval, indicating a possible clocking issue.
[View full width] Shanghai#show controller e1 1/0 E1 1/0 is up. Applique type is Channelized E1 - balanced No alarms detected. alarm-trigger is not set Version info Firmware: 20050620, FPGA: 16, spm_count = 0 Framing is CRC4, Line Code is HDB3, Clock Source is Internal. CRC Threshold is 320. Reported from firmware is 320. Data in current interval (864 seconds elapsed): 0 Line Code Violations, 0 Path Code Violations 24 Slip Secs, 0 Fr Loss Secs, 0 Line Err Secs, 0 Degraded Mins 24 Errored Secs, 0 Bursty Err Secs, 0 Severely Err Secs, 0 Unavail Secs Data in Interval 1: 3 Line Code Violations, 29 Path Code Violations 21 Slip Secs, 0 Fr Loss Secs, 3 Line Err Secs, 0 Degraded Mins 22 Errored Secs, 0 Bursty Err Secs, 4 Severely Err Secs, 0 Unavail Secs ! Additional output omitted |
Table 6-1 describes all the fields shown.
Field |
Description |
---|---|
E1 1/0 Is Up |
The E1 controller 0 in slot 1 is operating. The controller state can be up, down, or administratively down. Loopback conditions are shown by (Locally Looped) or (Remotely Looped). |
No Alarms Detected |
Any alarms that the controller detects are displayed here. Possible alarms are as follows:
|
Data in Current Interval (864 Seconds Elapsed) |
This field shows the current accumulation period, which rolls into the 24-hour accumulation every 15 minutes. The accumulation period is from 1 to 900 seconds. The oldest 15-minute period falls off the back of the 24-hour accumulation buffer. |
Line Code Violations |
This field indicates the occurrence of either a bipolar violation (BPV) or Excessive Zeros (EXZ) error event. |
Path Code Violations |
This field indicates a frame synchronization bit error in the D4 and E1-noCRC formats, or a cyclic redundancy check (CRC) error in the ESF and E1-CRC formats. |
Slip Secs |
This field indicates the replication or deletion of the payload bits of a DS1 frame. A slip can occur if there is a difference between the timing of a synchronous receiving terminal and the received signal. |
Fr Loss Secs |
This field indicates the number of seconds an "out-of-frame" error is detected. |
Line Err Secs |
A line errored second (LES) is one in which one or more "line code violation" errors are detected. |
Degraded Mins |
A degraded minute is one in which the estimated error rate exceeds 1E-6 but does not exceed 1E-3. |
Errored Secs |
In ESF and E1-CRC links, an errored second is a second in which one of the following is detected: one or more path code violations; one or more "out-of-frame" defects; one or more Controlled Slip events; a detected AIS defect. For D4 and E1-noCRC links, the presence of bipolar violations (BPV) also triggers an errored second. |
Bursty Err Secs |
Count of the number of seconds with fewer than 320 and more than 1 "path coding violation" error, no "severely errored frame" defects, and no detected incoming AIS defects. Controlled slips are not included in this parameter. |
Severely Err Secs |
For ESF signals, count of the number of seconds with one of the following errors: 320 or more "path code violation" errors; one or more "out-of-frame" defects; a detected AIS defect. For E1-CRC signals, a second with one of the following errors: 832 or more "path code violation" errors; one or more "out-of-frame" defects. For E1-noCRC signals, a second with 2048 or more line code violations. For D4 signals, a count of 1-second intervals with "framing" errors, or an "out-of-frame" defect, or 1544 line code violations. |
Unavail Secs |
Count of the total number of seconds where interface has been unavailable to pass traffic. |
Configuring ISDN PRI Trunks
To configure a PRI trunk, you need to gather the following information from the PSTN service provider:
ISDN PRI Switch Type |
Description |
---|---|
primary-4ess |
Lucent 4ESS switch type for the United States |
primary-5ess |
Lucent 5ESS switch type for the United States |
primary-dms100 |
Northern Telecom DMS-100 switch type for the United States |
primary-dpnss |
Digital Private Networking Signaling System (DPNSS) switch type for Europe |
primary-net5 |
NET5 switch type for the United Kingdom, Europe, Asia, and Australia |
primary-ni |
National ISDN switch type 2 for the United States |
primary-ntt |
NTT switch type for Japan |
primary-qsig |
Q.SIG switch type |
primary-ts014 |
TS014 switch type for Australia (now obsolete) |
Basic configuration of the interface for a PRI voice trunk begins by configuring the gateway for an ISDN switch type. You do this in global configuration mode by using the isdn switch-type switch type command, where switch type matches the device that the PSTN service provider is using; see Table 6-2. Defining a switch type globally for the gateway is required.
Note
You can override the ISDN switch type at the individual interface level. This is useful when you have interfaces in the gateway that are connected to different switches.
The second step is to configure the PRI voice timeslots on the controller. Enter controller configuration and use the command pri-group timeslots (1-23) for T1 or pri-group timeslots (1-31) for E1. This command associates the specified timeslots as ISDN channels. If you do not specify timeslots, all B-channels are associated to the PRI group. It is important to understand that the D-channel is always added to the pri-group, even if it is not explicitly specified, because it is required for signaling. Further, you can specify only one PRI group on the controller because each circuit has a single D-channel available.
After you specify the PRI group, the gateway automatically creates an associated voice port and serial interface. You configure the voice port in a similar way to the FXO voice port, as shown earlier.
The serial interface actually controls the D-channel (signaling channel). You do all configuration that affects processing of the signaling on the D-channel. It is on the D-channel that you can set the B-channel selection order if necessary. You do this by entering interface configuration mode on the D-channel serial interface and using the isdn bchan-number-order {ascending | descending} command.
After you have completed these steps, your service should be operational. Remember that during call setup, both the calling and called numbers are signaled on the D-channel. This allows call control to match on either field to route the call. See Chapter 9, "Dial Plans," and Chapter 10, "Digit Manipulation," for more details. Example 6-12 shows a typical E1 ISDN PRI configuration. In this example, the first four B-channels are used for voice traffic.
Timeslot 16 is the E1 D-channel that was added to the PRI group automatically. The B-channel selection order was also changed to begin with B-channel 1. Interface Serial1/0:15 and voice port 1/0:15 were automatically created when the PRI group was added to the E1 controller.
[View full width] Shanghai#show running-config Building configuration... ! ! Unnecessary output removed ! ! ISDN switch type set globally ! isdn switch-type primary-net5 ! ! First 4 B-channels included in the pri-group, D-channel 16 added automatically ! controller E1 1/0 pri-group timeslots 1-4, 16 ! ! D-channel interface ! interface Serial1/0:15 no ip address isdn switch-type primary-net5 isdn incoming-voice voice isdn bchan-number-order ascending no cdp enable ! ! Voice port created for pri-group ! voice-port 1/0:15 ! end |
MGCP
If MGCP is going to control the ISDN PRI, a few configuration changes are required. With MGCP, the Layer 3 Q.931 messages are backhauled to the CallManager and not processed by the gateway.
When setting up the PRI group on the controller, you need to specify that MGCP will control the signaling. You do this from controller configuration mode by using the pri-group timeslots number service mgcp command.
Under the serial interface created for the D-channel, you also must code the isdn bind-l3 ccm-manager command. This attaches the D-channel Q.931 protocol to the CallManager backhaul. Example 6-13 shows the differences in configuration for an ISDN PRI under MGCP control.
Shanghai#show running-config Building configuration... ! ! Unnecessary output removed ! ! ISDN switch type set globally ! isdn switch-type primary-net5 ! ! Added 'service mgcp' for MGCP-controlled interface ! controller E1 1/0 pri-group timeslots 1-4, 16 service mgcp ! ! D-channel interface ! interface Serial1/0:15 no ip address isdn switch-type primary-net5 isdn incoming-voice voice isdn bind-l3 ccm-manager no cdp enable ! ! Voice port created for pri-group ! voice-port 1/0:15 ! end |
The isdn bind-l3 ccm-manager command shown under interface Serial1/0:15 is available only after you add the service mgcp keywords to the PRI group definition. If the command is unknown when you try to configure it, look at the controller to ensure that the PRI group is defined correctly.
Verifying and Troubleshooting
To verify that the circuit is operating properly, you can use the show isdn status command. Example 6-14 shows the output of this command.
Shanghai#show isdn status Global ISDN Switchtype = primary-net5 ISDN Serial1/0:15 interface dsl 0, interface ISDN Switchtype = primary-net5 Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x8000000F Number of L2 Discards = 0, L2 Session ID = 2 Total Allocated ISDN CCBs = 0 |
The example shows the status of the circuit at Layers 1, 2, and 3. Layer 1 should show ACTIVE, with the Layer 2 state as MULTIPLE FRAME ESTABLISHED. If that is not the case, check the physical interface for errors, as shown in the previous section. Also verify that the switch type is correct and that it matches the PSTN specification.
If Layer 2 is showing an incorrect state, you can use the debug isdn q921 trace to help determine the cause. Example 6-15 shows a problem with Layer 2, where both sides of the connection are configured as network, and the Layer 2 connection cannot be established.
[View full width] Miami#show isdn status Global ISDN Switchtype = primary-ni ISDN Serial1/0:23 interface ******* Network side configuration ******* dsl 0, interface ISDN Switchtype = primary-ni Layer 1 Status: ACTIVE Layer 2 Status: TEI = 0, Ces = 1, SAPI = 0, State = AWAITING_ESTABLISHMENT Layer 3 Status: 0 Active Layer 3 Call(s) Active dsl 0 CCBs = 0 The Free Channel Mask: 0x80000000 Number of L2 Discards = 0, L2 Session ID = 11 Total Allocated ISDN CCBs = 0 Miami#debug isdn q921 debug isdn q921 is ON. Miami# *Jul 5 19:31:01.477: ISDN Se1/0:23 Q921: Net TX -> SABMEp sapi=0 tei=0 *Jul 5 19:31:01.933: ISDN Se1/0:23 Q921: Net RX <- BAD FRAME( 0x02017F) *Jul 5 19:31:02.477: ISDN Se1/0:23 Q921: Net TX -> SABMEp sapi=0 tei=0 *Jul 5 19:31:02.933: ISDN Se1/0:23 Q921: Net RX <- BAD FRAME(0x02017F) *Jul 5 19:31:03.477: ISDN Se1/0:23 Q921: L2_EstablishDataLink: sending SABME |
If the status of Layer 1 and 2 are good and you are still having trouble placing calls, make a test call with debug isdn q931 active. This debug shows activity on the PRI D-channel and is useful in troubleshooting. First, if your test call produces trace output, it confirms that the request is actually being sent out the PRI interface. Second, as you can see in Example 6-16, it shows a great deal of information about the call. You can see the calling and called numbers, which B-channel is requested, and failure cause codes with reasonably clear descriptive text.
[View full width] Shanghai#debug isdn q931 debug isdn q931 is ON. Shanghai# *May 13 13:46:32.662: ISDN Se1/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x0, Calling num 2001 *May 13 13:46:32.662: ISDN Se1/0:15 Q931: Applying typeplan for sw-type 0x12 is 0x0 0x0, Called num 19192099832 *May 13 13:46:32.662: ISDN Se1/0:15 Q931: TX -> SETUP pd = 8 callref = 0x0011 Bearer Capability i = 0x8090A3 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x0081, '2001' Plan:Unknown, Type:Unknown Called Party Number i = 0x80, '1212' Plan:Unknown, Type:Unknown *May 13 13:46:32.694: ISDN Se1/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x80 11 Channel ID i = 0xA98381 Exclusive, Channel 1 *May 13 13:46:32.710: ISDN Se1/0:15 Q931: RX <- DISCONNECT pd = 8 callref = 0x8 011 Cause i = 0x8281 - Unallocated/unassigned number *May 13 13:46:32.710: ISDN Se1/0:15 Q931: TX -> RELEASE pd = 8 callref = 0x0011 *May 13 13:46:32.726: ISDN Se1/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0 x8011 |
This example shows that extension 2001 called 1212 on B-channel 1. The call failed because 1212 is an unassigned numberthe number does not exist. The PSTN expects numbers in E.164 format. You can quickly see that the gateway is not passing numbers in the correct format to the PSTN, which is a dial plan configuration issue. You can also tell from this example that the B-channels are being selected in ascending order as was configured.
An additional command can provide assistance when you are troubleshooting CallManager MGCP-controlled trunks. It is the show ccm-manager command that, among other things, shows the status of the Q.931 backhaul. You can see this information at the end of Example 6-17. Other useful information provided by this command includes the destination IP address for the backhaul link, packet and error counts, and the slot and port information for the interfaces that are being backhauled.
[View full width] Shanghai#show ccm-manager MGCP Domain Name: Shanghai Priority Status Host ============================================================ Primary Registered 10.1.5.2 First Backup None Second Backup None Current active Call Manager: 10.1.5.2 Backhaul/Redundant link port: 2428 Failover Interval: 30 seconds Keepalive Interval: 15 seconds Last keepalive sent: 21:50:32 UTC May 13 2005 (elapsed time : 00:00:04) Last MGCP traffic time: 21:50:35 UTC May 13 2005 (elapsed time : 00:00:020 Last failover time: None Last switchback time: None Switchback mode: Graceful MGCP Fallback mode: Not Selected Last MGCP Fallback start time: None Last MGCP Fallback end time: None MGCP Download Tones: Disabled Backhaul Link info: Link Protocol: TCP Remote Port Number: 2428 Remote IP Address: 10.1.5.2 Current Link State: OPEN Statistics: Packets recvd: 1 Recv failures: 0 Packets xmitted: 1 Xmit failures: 0 PRI Ports being backhauled: Slot 1, port 0 Configuration Error History: FAX mode: cisco |
Q.931 information is also written to the CallManager trace files and can be viewed with the Q.931 Translator utility provided with CallManager. This helps in troubleshooting by allowing you to look at historical call data. You can also use the Voice Log Translator (VLT) tool to parse and translate CallManager trace file data. It is available for download at Cisco.com.
Configuring E1 R2 Trunks
R2 signaling is a CAS method that is defined in International Telecommunications Union Telecommunication Standardization Sector (ITU-T) recommendations Q.400 through Q.490. This signaling method is widely used in Europe, Latin America, Australia, and Asia. You need to beware of many country-specific variants before you configure R2. These variants are described in the Consultative Committee for International Telegraph and Telephone (CCITT) R2 specification.
R2 signaling has two components: line signaling, which provides supervisory control signals, and interregister signaling, which provides call setup control.
Line signaling uses the ABCD bits in timeslot 16 for supervisory purposes, such as indicating line seizure or line clearing. Only the A and B bits are used in the CCITT-R2 format. Line signaling is supported in Cisco Voice Gateways in the following formats:
Interregister signaling uses in-band multifrequency signals in each timeslot to send the calling number, called number, and call category. Three types of interregister signaling are supported:
To configure an R2 signaling on an E1 trunk, you need to gather the following information from the PSTN service provider:
To begin, you need to configure the physical port, as shown previously in the section "Configuring E1/T1 Physical Layer Connections."
The next step is to set up the DS0 groups. A DS0 group is a logical grouping of individual 64-Kbps voice channels on the E1. You can configure the entire E1 as a single DS0 group, or you can set up several DS0 groups, depending on requirements. For a PSTN connection, the DS0 logical configuration must match what the PSTN provides.
You configure the DS0 groups by using the following command:
ds0-group group number timeslots 1-31 type line-signaling-type interregister- signaling-type ani
The group number assigns a logical grouping to the channels defined in this DS0 group. You can specify up to 31 separate groups on an E1 trunk. Each logical DS0 group will have a separate voice port.
The timeslots keyword identifies the channels on the E1 that are being grouped under this DS0 group.
The line-signaling-type and interregister-signaling-type must match those that the PSTN service provider requires.
The ani optional keyword is added if the PSTN is providing ANI (calling number) information. The DNIS (called number) information is always provided.
Example 6-18 shows a typical E1 R2 configuration for a voice trunk in the United Kingdom. In this example, the first four channels of the E1 are used for voice. ANI information is being provided. Voice port 1/0:0 was created for the logical DS0 group 0. The cptone GB command under the voice port sets the locale to the United Kingdom.
Leeds#show running-config Building configuration... ! ! Unnecessary output deleted ! version 12.3 ! controller E1 1/0 ds0-group 0 timeslots 1-4 type r2-digital r2-compelled ani ! ! Voice port 1/0:0 created for logical ds0-group 0 ! voice-port 1/0:0 cptone GB ! end |
Many country-specific variations of R2 signaling exist. To customize the voice gateway for the country where it is deployed, use the cas-custom channel number subcommand in controller configuration mode. The channel number parameter must match the group number of the DS0 group that you are customizing. Whenever you are setting up customized R2, Cisco recommends that you begin by using the country country use-defaults subcommand of the cas-custom command. This subcommand automatically sets up typical parameters for the chosen country. You can customize further from that starting point. Example 6-19 shows how you can use the cas-custom command.
Leeds#show running-config Building configuration... ! ! Unnecessary output deleted ! version 12.3 ! controller E1 1/0 ds0-group 0 timeslots 1-4 type r2-digital r2-compelled ani cas-custom 0 unused-abcd 0 1 1 1 country hongkong-china answer-signal group-b 1 ! ! Voice-port 1/0:0 created for logical ds0-group 0 ! voice-port 1/0:0 cptone HK ! end |
In Example 6-19, the router generated the three subcommands following cas-custom 0. The actual command entered under cas-custom 0 was country hongkong-china use-defaults.
Note
You can find further information on country-specific R2 protocol customization on Cisco.com:
E1 R2 customization with the cas-custom commandwww.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800942f2.shtml
E1 R2 signaling configuration and troubleshootingwww.cisco.com/en/US/partner/tech/tk652/tk653/technologies_configuration_example09186a00800ad389.shtml
MGCP
CallManager MGCP does not support E1 CAS (E1 R2) trunks.
Verifying and Troubleshooting
You can view the status and signaling of the voice ports by individual timeslot (channel) using the show voice port summary command, as shown in Example 6-20. This example reflects the configuration shown previously in Example 6-18. Voice port 1/0:0 is controlling associated timeslots 1 through 4 with r2-digital signaling. You can also see that channel 1 is active (seized).
Leeds#show voice port summary IN OUT PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC ========= == ============ ===== ==== ======== ======== == 1/0:0 01 r2-digital up dorm idle seized y 1/0:0 02 r2-digital up dorm idle idle y 1/0:0 03 r2-digital up dorm idle idle y 1/0:0 04 r2-digital up dorm idle idle y |
You can find information about the progress of a call on the voice port with the show voice call summary command. In Example 6-21, you can see that a call is in progress on voice port 1/0:0 channel 4.
Leeds#show voice call summary PORT CODEC VAD VTSP STATE VPM STATE ============== ======== === ==================== ====================== 1/0:0.1 - - - R2_Q421_IDLE 1/0:0.2 - - - R2_Q421_IDLE 1/0:0.3 - - - R2_Q421_IDLE 1/0:0.4 None y S_R2_DIALING R2_Q421_OG_SEIZE_ACK |
If you are having problems completing calls on the E1 trunk, the debug vpm signal and debug vtsp all commands can provide useful information to assist in troubleshooting.
Configuring T1 CAS Trunks
T1 CAS circuits use in-band signaling, "robbing" the least significant bits from channels that carry voice to handle framing and to pass state information. The framing and state information is carried along with the voice information in each channel. This is in contrast to ISDN, which has one channel dedicated to carrying signaling for all the other voice channels.
To configure a T1 CAS trunk, you need to gather the following information from the PSTN service provider:
- For FXO or FXSLoop-start or ground-start.
- For E&MImmediate-start, delay-dial, wink-start (also known as Feature Group B (FGB)), double-wink-start (also known as Feature Group D [FGD]), or Feature Group D Exchange Access North American (FGD-EANA).
Signaling information is passed over T1 circuits by emulating the methods used for analog trunks: FXO, FXS, and E&M. This information is passed using the A and B bits in SF circuits or the ABCD bits in ESF circuits. For more details on these signaling methods, see Chapter 5.
Loop-start signaling is the simplest CAS signaling method. Unfortunately, it has several disadvantages that make it an undesirable choice. It has no far-end answer or disconnect supervision. That is, you cannot relay when the remote side of the call answers or hangs up. Loop-start also provides no seizure of the channel on incoming calls. This can lead to a condition known as glare when both sides of the connection try to place a call at the same time.
Ground-start signaling has some advantages over loop-start. Ground-start can recognize far-end disconnect. It also can seize the channel on an inbound call, preventing the occurrence of glare. For this reason, ground-start is often used between PBXs or from a PBX to a voice gateway.
E&M signaling has many advantages over the other types shown so far and is the preferred choice for CAS trunks. E&M provides both answer and disconnect supervision and glare avoidance, can receive ANI information (FGD only), and can send ANI information (FGD-EANA only).
To begin, it is necessary to configure the physical port, as shown previously in the section, "Configuring E1/T1 Physical Layer Connections."
The next step is to configure DS0 groups. You configure DS0 groups in controller configuration mode using the ds0-group keyword. The syntax is ds0-group group-number timeslots 023 type signaling-type. The group-number parameter identifies the DS0 group and is a number in the range of 023. You can configure up to 24 DS0 groups, and you must assign each a different number. The timeslots 023 parameter establishes which channels in the T1 belong to this logical voice port. Table 6-3 lists the signaling types supported in current Cisco IOS versions that you might need for a PSTN connection.
T1 CAS Signaling Type |
Description |
---|---|
e&m-delay-dial |
E&M delay-dial signaling. |
e&m-fgd |
E&M Type II Feature Group D. |
e&m-immediate-start |
E&M immediate start. |
e&m-wink-start |
E&M wink-start signaling. Also known as Feature Group B (FGB) signaling. |
fgd-eana |
FGD Exchange Access North American. |
fxo-ground-start |
FXO ground-start signaling. |
fxo-loop-start |
FXO loop-start signaling. |
fxs-ground-start |
FXS ground-start signaling. |
fxs-loop-start |
FXS loop-start signaling. |
none |
External call control. |
It is important to note that if CallManager MGCP call control is used with the T1 CAS port, only E&M wink-start and E&M delay-dial are supported.
After you define the DS0 group, the gateway creates a voice port in the form of voice-port slot/port:ds0-group-number, as shown in Example 6-22. In this example, two DS0 groups were created. The first is associated with voice port 1/0:0, which controls timeslots 1 through 4 on the T1 and uses E&M wink-start signaling. The second is associated with voice port 1/0:1, which controls timeslots 5 through 8 on the T1 using E&M FGD signaling. Call control references the voice ports individually to route calls in and out of the associated timeslots on the T1.
Miami#show running-config Building configuration... ! ! Unnecessary output removed ! version 12.3 ! controller T1 1/0 framing esf linecode b8zs ds0-group 0 timeslots 1-4 type e&m-wink-start ds0-group 1 timeslots 5-8 type e&m-fgd ! ! Voice ports created by DS0 groups defined above voice-port 1/0:0 ! voice-port 1/0:1 ! end |
MGCP
When you configure a T1 CAS connection that CallManager MGCP will control, you set all the call control parameters on the CallManager, rather than on the gateway. On the gateway, you associate the voice port created to the MGCP application with a dial peer.
CallManager MGCP supports only E&M wink-start and E&M delay dial signaling types for T1 CAS circuits.
You can control whether a port is used for inbound calls, outbound calls, or both.
You can also set the number of significant digits that CallManager should collect for an inbound call. You might need to do this if you are using DID and the last digits of the dialed number represent the extension where the call is to be routed. By setting the number of significant digits to the extension length, you can remove the need to further manipulate the incoming dialed number to deliver the call.
You can individually apply parameters to voice timeslots that are defined on the T1 controller.
Verifying and Troubleshooting
You can view the status and signaling of the voice ports by individual timeslot (channel) using the show voice port summary command, as demonstrated in Example 6-23. This example reflects the configuration shown previously in Example 6-22. Voice port 1/0:0 is controlling associated timeslots 1 through 4 with E&M wink-start signaling. You can also see that channel 1 is active (seized). Voice port 1/0:1 is controlling associated timeslots 5 through 8 with E&M FGD signaling.
Miami#show voice port summary IN OUT PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC ========= == ============ ===== ==== ======== ======== == 1/0:0 01 e&m-wnk up up seized idle y 1/0:0 02 e&m-wnk up dorm idle idle y 1/0:0 03 e&m-wnk up dorm idle idle y 1/0:0 04 e&m-wnk up dorm idle idle y 1/0:1 05 e&m-fgd up dorm none none y 1/0:1 06 e&m-fgd up dorm none none y 1/0:1 07 e&m-fgd up dorm none none y 1/0:1 08 e&m-fgd up dorm none none y |
You can also find information about the progress of a call on the voice port with the show voice call summary command. In Example 6-24, you can see that a call is in progress on voice port 1/0:0 channel 3.
Miami#show voice call summary PORT CODEC VAD VTSP STATE VPM STATE ============== ======== === ==================== ====================== 1/0:0.1 - - - EM_ONHOOK 1/0:0.2 - - - EM_ONHOOK 1/0:0.3 None y S_DS_DIALING EM_WAIT_DIAL_DONE 1/0:0.4 - - - EM_ONHOOK 1/0:1.5 - - - FGD_ONHOOK 1/0:1.6 - - - FGD_ONHOOK 1/0:1.7 - - - FGD_ONHOOK 1/0:1.8 - - - FGD_ONHOOK |
If you are having a problem with calls on the port, you can use the debug voip vtsp all command, which shows the progress of the call control signaling for the call in progress on the port.
Configuring ISDN BRI Trunks
ISDN BRI the only service described that is not delivered on a T1 or E1 circuit. The information and tasks described in the section "Configuring E1/T1 Physical Layer Connections" do not apply to a BRI. BRI voice trunks are not used often in the United States; however, they are popular in other areas of the world.
An ISDN BRI is similar to a PRI in that the signaling information is carried separately from the voice-bearer traffic in a dedicated data channel. A BRI circuit has just two voice-bearer channels and one data channel (2B+D). Only two voice calls can occur simultaneously on a BRI.
BRI trunks operate in a master-slave mode. The master side is called the network termination (NT) side and is responsible for providing clocking and power. The slave side is called the terminal equipment (TE) side. This master-slave relationship extends to Layer 2 and Layer 3 protocols, as defined by ITU-T specifications Q.921 and Q.931, respectively. The master side initiates the Layer 2 and Layer 3 communications. When connecting to the PSTN, the voice gateway is always configured as the slave side (TE). An NT1 device is typically required to connect the voice gateway TE interface to the PSTN.
To configure an ISDN BRI voice trunk, you need to gather the following information:
Begin configuration by defining the ISDN switch type in global configuration mode using the isdn switch-type switch-type command. Table 6-4 lists ISDN switch types supported by current versions of Cisco IOS that you might encounter when setting up a PSTN BRI trunk. You can override this setting at the port level if necessary to support multiple connections to different switch types.
Locale |
ISDN BRI Switch Type |
Description |
---|---|---|
Australia, Europe, United Kingdom |
basic-1tr6 |
German 1TR6 ISDN switch |
basic-net3 |
NET3 ISDN BRI for Norway NET3, Australia NET3, and New Zealand NET3 switch types; ETSI-compliant switch types for Euro-ISDN E-DSS1 signaling system |
|
vn3 |
French ISDN BRI switch |
|
Japan |
ntt |
Japanese NTT ISDN switch |
North America |
basic-5ess |
Lucent basic rate 5ESS switch |
basic-dms100 |
Northern Telecom DMS-100 BRI switch |
|
basic-ni |
National ISDN switch |
Continue the configuration in interface configuration mode using the interface bri slot/port command. If necessary, configure the first SPID using the isdn spid1 spid [ldn] command, as well as using the PSTN-assigned SPID and LDN numbers. You can configure the second SPID in the same manner using the isdn spid2 spid [ldn] command.
For voice calls to be properly routed to digital signal processor (DSP) resources for voice processing, you must also code the isdn incoming-voice voice command on the interface.
Example 6-25 shows the configuration of a typical ISDN BRI voice trunk.
Lima#show running-config Building configuration... ! ! Unnecessary output removed ! version 12.3 ! isdn switch-type basic-5ess ! interface BRI1/0 no ip address no ip directed-broadcast isdn switch-type basic-5ess isdn twait-disable isdn incoming-voice voice ! voice-port 1/0 ! end |
MGCP
CallManager versions 4.1.3SR1 and later allow you to control BRI voice ports with MGCP. With MGCP, the Layer 3 Q.931 messages are backhauled to the CallManager and are not processed by the gateway.
Under the BRI interface, you must code isdn bind-l3 ccm-manager service mgcp. This attaches the D-channel Q.931 protocol to the CallManager backhaul. Example 6-26 shows the configuration for an ISDN BRI under MGCP control.
Lima#show running-config Building configuration... ! ! Unnecessary output removed ! version 12.3 ! isdn switch-type basic-net3 ! interface BRI1/0/0 no ip address no ip directed-broadcast isdn switch-type basic-net3 isdn bind-l3 ccm-manager service mgcp isdn incoming-voice voice ! voice-port 1/0/0 ! end |
In Example 6-26, the ISDN switch type is basic-net3. This is the only switch type supported by CallManager for MGCP control of a BRI.
Verifying and Troubleshooting
To verify that the circuit is operating properly, you can use the show isdn status command. Example 6-27 shows the output of this command. The example shows that Layers 1 and 2 are active and that no Layer 3 calls are currently in progress. You can also see the ISDN switch type configured globally and for the specific port.
Lima#show isdn status Global ISDN Switchtype = basic-5ess ISDN BRI1/0 interface dsl 0, interface ISDN Switchtype = basic-5ess Layer 1 Status: ACTIVE Layer 2 Status: TEI = 64, Ces = 1, SAPI = 0, State = MULTIPLE_FRAME_ESTABLISHED Layer 3 Status: 0 Active Layer 3 Call(s) Activated dsl 0 CCBs = 0 |
If you have not properly established the Layer 2 connection, use the debug isdn q921 trace to help identify the problem. If the Layer 1 and Layer 2 output looks correct and you are still having trouble placing or receiving calls on the port, use the debug isdn q931 trace command. Both of these are described in the troubleshooting portion of the "Configuring ISDN PRI Trunks" section.
Case Study Add an E1 R2 Connection to the Leeds Gateway |
Part I: Voice Gateways and Gatekeepers
Gateways and Gatekeepers
Part II: Gateways
Media Gateway Control Protocol
H.323
Session Initiation Protocol
Circuit Options
Connecting to the PSTN
Connecting to PBXs
Connecting to an IP WAN
Dial Plans
Digit Manipulation
Influencing Path Selection
Configuring Class of Restrictions
SRST and MGCP Gateway Fallback
DSP Resources
Using Tcl Scripts and VoiceXML
Part III: Gatekeepers
Deploying Gatekeepers
Gatekeeper Configuration
Part IV: IP-to-IP Gateways
Cisco Multiservice IP-to-IP Gateway
Appendix A. Answers to Chapter-Ending Review Questions
Index