Configuring a Gateway of Last Resort Using IP Commands

Commands ip default-gateway, ip default-network and ip route 0.0.0.0 0.0.0.0 explained:


http://www.cisco.com/en/US/tech/tk365/technologies_tech_note09186a0080094374.shtml

NX-OS and IOS Comparision

Cisco UCS servers

Название	Размер	CPU sockets	CPU	Память	Диски	I/O
UCS B200 M2	Половина	2	Intel Xeon 5600	12 DIMM 96 GB	2 SFF SAS	1 Mezz
UCS B230 M1	Половина	2	Intel Xeon 7500	32 DIMM 256 GB	2 SFF SAS	1 Mezz
UCS B250 M2	Целый	2	Intel Xeon 5600	48 DIMM 384 GB	2 SFF SAS	2 Mezz
UCS B440 M1	Целый	4	Intel Xeon 7500	32 DIMM 256GB	4 SFF SAS/SATA	2 Mezz
UCS C200 M2	1 RU	2	Intel Xeon 5600	12 DIMM 96 GB	4 SFF SAS/SATA	-
UCS C210 M2	2 RU	2	Intel Xeon 5600	12 DIMM 96 GB	16 SFF SAS/SATA	-
UCS C250 M2	2 RU	2	Intel Xeon 5600	48 DIMM 384 GB	4 SFF SAS/SATA	-
UCS C460 M1	4 RU	4	Intel Xeon 7500	64 DIMM 512 GB	12 SFF SAS/SATA	-

Cisco Expo-2010 Ukraine

Digit Manipulation

Digit Manipulation Techniques summary table:

Digit Stripping
POTS dial peers by default remove, or strip, any outbound digits that explicitly match their destination pattern.
To disable digit stripping use command:
 no digit-strip

Forward Digits
You can specify the exact number of digits to be forwarded. If the number of digits presented exceeds the number allowed, the rightmost
digits are sent.
forward-digits [number | all | extra]
where:
- number gives the number of digits to be forwarded.
- all means to forward all digits.
- extra tells the gateway to forward any digits that are longer than the length of the destination pattern.

Prefix Digits
You can transmit more than the dialed digits of a called number. Only for POTS dial peers.
prefix string

Number Expansion

Number expansion is another way to add digits to an outgoing called number; however, number expansion is applied to the gateway as a whole and manipulates only the called number. Number expansion manipulation occurs before any outbound dial peer is matched. Outbound dial peers match the expanded numbers, not the original ones.
num-exp original-number expanded-number

Caller ID
During an outgoing call, the CLID is sent as part of the call information. CLID information includes at least one calling party number. The CLID might also include a name, a second number, and redirecting number information.
CLID commands:

- clid network-number number Specifies the network number to be sent in the IE. It sets the presentation indicator to "Y" and the screening indicator to "network provided." Available in both dial-peer and voice service voip configuration modes.

- clid second-number strip Removes the original calling number from the H.225 source address field. You can also give this command on the same line as the clid network-number command. It is valid only if you have configured a network number.

- clid restrict Transmits the calling party information but sets the presentation indicator to "N" so that it is not displayed to the called party.

- clid strip [name | pi-restrict [all]] Removes the CLID number if just the clid strip command is given and sets the presentation indicator to "N." It removes the CLID name if the name option is added. To remove both the name and number, you must enter both commands separately. The pi-restrict option causes the CLID number to be stripped only when you set the progress indicator to "restricted." Adding the all keyword strips both the CLID number and name. The pi-restrict all option is available in both dial-peer and voice service voip configuration modes.

- clid substitute name Substitutes the calling number for the display name. Available in both dial-peer and voice service voip configuration modes.

Voice Translation Rules and Profiles

Using voice translation profiles for digit manipulation requires three steps:

Step 1.	Create one or more voice translation rules and a prioritized list of translations associated with each rule. A maximum of 128 rules is supported, with 15 translations per rule.
Step 2.	Create one or more voice translation profiles and associate the translation rules to the profile. You can define up to 1000 profiles, each with its own unique name. Within the profile, you can apply one voice translation rule to calling numbers, one to called numbers, and one to redirected called numbers.
Step 3.	Apply the voice translation profile to all VoIP calls globally, a dial peer, a voice port, a trunk group, a source IP group, or an interface.

Creating Voice Translation Rules

To create a voice translation rule, use the command voice translation-rule tag in global configuration mode. Then create an ordered list of one or more rules with the following command:

rule precedence /match pattern/ /replace pattern/ [type {match-type replace-type} [plan {match-type replace-type}]]

You can enter rules in any order; the precedence value determines the order in which the rules are executed. You can configure up to 15 rules.

Creating Voice Translation Profiles

(config)#voice translation-profile name
(cfg-translation-profile)#translate ?
  called           Translation rule for the called-number
  calling          Translation rule for the calling-number
  redirect-called  Translation rule for the redirect-number
(cfg-translation-profile)#translate called 1
(cfg-translation-profile)#translate calling 2
(cfg-translation-profile)#translate redirect-called 3

Applying Translation Profiles

After you create a voice translation profile, you can assign it to:

- Dial-Peer

- Voice Port

- Trunk Group

- VoIP Calls Globaly

- NFAS

- SRST

Blocking Calls

!
(config)#voice translation-rule 410
(cfg-translation-rule)#rule 1 reject /expression/
!
(config)#voice translation-profile BLOCK
(cfg-translation-profile)#translate calling 410
!
(config)#dial-peer voice 1 pots
(config-dial-peer)#call-block translation-profile incoming BLOCK
(config-dial-peer)#call-block disconnect-cause incoming call-reject
!

Order of Operation in Digit Manipulation

Translation Profile Order

 Troubleshooting Digit Manipulation

- test voice translation-rule rule-number phone-number Shows the results of a translation rule, enabling you to test it to ensure that it does what you planned
- debug voice translation Shows the translations happening
- show dialplan numbernumber Verifies number expansion and which dial peers a phone number matches
- debug voip ccapi inout Shows inbound and outbound dial peers being matched
- show num-exp[number] Displays the number expansion rules configured
- show dial-peer voice [tag] Displays any CLID, translation profiles, call blocking, disconnect cause, digit stripping, forwarding, or prefixing that is configured on the dial peer
- show voice translation-rule [number | sort [ascending|descending]] Lists the translation rules that are configured on the router and all translation patterns configured for each one
- show voice translation-profile [name | sort [ascending|descending]] Lists the translation profiles configured on the router and all translation rules associated with each one
- debug isdn q931 Shows the called and calling numbers sent out a PRI link for troubleshooting CLID commands
- csim start phone-number Simulates a phone call from the router; can be used with debugs

Cisco Unified Mobility

Consists:
- Mobile Connect: Allows an incoming call to a user's enterprise phone number to be offered to the user's office phone, as well as to up to ten configurable remote destinations. Typically such remote destinations are mobile phones and home office phones.
- Mobile Voice Access (MVA): similar to those of Mobile Connect for outgoing calls. Users who are outside the enterprise can make calls as if they were directly connected to CUCM. This functionality is commonly called
Direct Inward System Access (DISA) in traditional telephony environments.

Dial-peers Configuration

Dial-peer wildcard symbols:

 Examples of how these wildcard symbols are applied to the destination pattern and the dial string that results when dial string 4085551234 is matched to an outbound POTS dial peer.

 Matching Inbound Dial Peers

The router uses the full digit string received in the setup request for matching against the configured dial peers. The router or gateway matches call setup element parameters in the following order:
1. The router or gateway attempts to match the called number of the call setup request with the configured incoming called-number of each dial peer.
2. If a match is not found, the router or gateway attempts to match the calling number of the call setup request with the answer-address of each dial peer.
3. If a match is not found, the router or gateway attempts to match the calling number of the call setup request to the destination-pattern of each dial peer.
4. The voice port uses the voice port number associated with the incoming call setup request to match the inbound call leg to the configured dial peer port parameter.
5. If multiple dial peers have the same port configured, the router or gateway matches the first dial peer added to the configuration.
6. If a match is not found in the previous steps, dial peer 0 is matched.

Characteristics of the Default Dial Peer

Default dial peers are used for inbound matches only. Dial peer 0 has the following characteristics:
- Any codec
- IP precedence 0
- VAD enabled
- No RSVP support
- fax-rate service
- no ivr application

Matching Outbound Dial Peers

Outbound dial-peer matching is completed on a digit-by-digit basis. Therefore, the router or gateway checks for dial-peer matches after receiving each digit and then routes the call when a full match is made.
The router or gateway uses the dial peer destination-pattern command to determine how to route the call.

Device Mobility

When phones move between different CUCM sites, inaccurate phone settings may occur.
The phone configuration parameters that can be dynamically applied to the device configuration are grouped in two categories:
- Roaming-Sensitive Settings
 Date/Time Group (dp)
 Region (dp)
 Location (dp)
 Connection Monitor Duration (dp)
 Network Locale (dp, phone)
 SRST Reference (dp, phone)
 Media Resource Group List (dp, phone)
 Physical Location (dp)
 Device Mobility Group (dp)
- Device Mobility-Related Settings (impact on call routing)
 Device Mobility Calling Search Space (dp, phone)
 AAR Calling Search Space (dp, phone) (Calling Search Space only in the Phone Configuration not Line!)
 AAR Group (dp, phone)

Overlapping parameters configured at the phone have higher priority than settings at the home device pool and lower priority than settings at the roaming device pool.

Algorithm:
The current device pool is chosen as follows:
 - If the DMI is associated with the phone's home device pool, the phone is considered to be in its home location. Therefore, Device Mobility does not reconfigure the phone.
 - If the DMI is associated with one or more device pools other than the phone's home device pool, one of the associated device pools is chosen based on a round-robin load-sharing algorithm.
If the current device pool is different from the home device pool, the following checks are performed:
 - If the physical locations are not different, the phone's configuration is not modified.
 - If the physical locations are different, the roaming-sensitive parameters of the current roaming device pool are applied.
 - If the Device Mobility Groups are the same, in addition to different physical locations, the Device Mobility-related settings are also applied, along with the roaming-sensitive parameters.
When a user roams with a device from Germany to the U.S., all the roaming-sensitive settings are applied, but the Device Mobility-related settings are not applied. The phone now uses the PSTN gateway and dial rules of its home location even though the user moved to another site. The user does not have to adapt to the dial rules of the local site to which the phone was moved.


Configuration:
1. Configure physical locations.
2. Configure Device Mobility Groups.
3. Configure device pools.
4. Configure DMIs (that is, IP subnets).
5. Set the Device Mobility mode using the following:
 - A Cisco CallManager service parameter to set the default for all phones
 - The Phone Configuration window for an individual configuration for each phone.

Call Admission Control

CAC limits the number of calls between certain parts of the network to avoid bandwidth oversubscription, with too many voice calls over WAN links.

Locations:
The configured bandwidth limit is independent of the call's destination location. Unlike region configuration, in which the maximum permitted codec is configured for each pair of regions, the bandwidth limit of a location applies to all interlocation calls, regardless of the other location.

CUCM CAC is based on a hub-and-spoke topology, in which all calls over the WAN are monitored for CAC as if they go through HQ. Because the configured bandwidth limit does not consider the destination location, the 24-kbps limit of BR1 allows any call to go out or in, regardless of where it goes to or comes from. The headquarters limit is unaffected by such a call. Only locations BR1 and BR2 subtract 24 kbps from their limits. Because locations-based CAC does not provide topology awareness, CUCM does not even know that the call physically flows through headquarters. 

1. Add locations and configure the CAC bandwidth limit.
2. Assign locations to devices.

RSVP-Enabled Locations
RSVP can be enabled selectively between pairs of locations.
RSVP makes this CAC mechanism WAN topology-aware, because RSVP will communicate over the WAN. Standard locations do not contain in their configurations details of the WAN topology. RSVP-enabled locations work well with all topologies (full-mesh, partial-mesh, and hub-and-spoke) and adapt to network changes by considering the actual topology.
An RSVP agent is a device called a Media Termination Point (MTP) through which the call has to flow. RSVP is used only between the two RSVP agents. The Real-Time Transport Protocol (RTP) stream from the IP Phone to the RSVP agent does not use RSVP.
There are three separate RTP streams: Phonel talks to RSVP Agentl, RSVP Agentl talks to RSVP Agent2, and RSVP Agent2 talks to Phone2.
Media resource group lists are used to identify the RSVP agent to be used by an IP Phone.
Confiiguration:
1. Configure RSVP service parameters.
2. Configure RSVP agents in Cisco IOS.
!
dspfarm profile 1 mtp
 codec pass-through
 rsvp
 maximum sessions software 20
asswciate application SCCP
!
interface Serial0/0
 ip rsvp bandwidth 40
!
3. Add RSVP agents to CUCM.
 Cisco IOS Enhanced Software Media Termination Point
4. Enable RSVP between location pairs.
5. Configure media resource groups.
6. Configure media resource group lists.
7. Assign media resource group lists to devices.

AAR
AAR does not work with Survivable Remote Site Telephony (SRST). AAR is activated only after a call is denied by CAC, not by WAN failures.
AAR works only for calls placed to internal directory numbers. It does not apply to calls placed to route patterns or feature patterns such as Meet-Me or Call Park. However, it does work for hunt pilots and computer telephony interface (CTI) ports.
The alternative number used for the PSTN call is composed of the dialed directory number, a prefix configured per AAR source and destination group, and the external phone number mask of the called device.

Configuration
1. Configure AAR service parameters (Cisco CallManager service).
 - Automated Alternate Routing Enable
 - Out-of-Bandwidth Text
 - AAR Network Congestion Rerouting Text
2. Configure partitions and CCS.
3. Configure AAR groups. Assign a dial prefixes.
4. Configure phones for AAR:
 Phone Configuration:
 - AAR Calling Search Space
 - AAR Group (or directory number Group is used)
 Directory Number Configuration:
 - AAR Settings: Voice Mail checkbox
 - AAR Destination Mask (CFNB)
 - AAR Group
 - External Phone Number Mask

Multicast MOH from Branch Router Flash Configuration

All IP Phones must be able to access the main-site CUCM MOH server from their media resource group list.
Use Multicast for MOH Audio check box has to be checked at the media resource group that includes the multicast-enabled MOH server.
G.711 codec is used between the MOH server and the branch phones, because SRST multicast MOH supports only G.711.
Recommended practice dictates that you increment multicast on IP address instead of port number.

1. Enable multicast routing in the network.
 (config)#ip multicast-routing
 (config-if)#ip pim sparse-dense-mode
2. Configure multicast MOH in CUCM:
 a. Configure MOH audio sources for multicast MOH.
 b. Configure MOH audio server for multicast MOH.
 c. Configure the maximum hop value to prevent multicast MOH streams from being sent over the IP WAN.
3. Enable multicast MOH from branch router flash at the branch router.
call-manager-fallback
 ip source-address
 moh moh-file.au
 multicast moh 239.0.0.1 port 20120
4. Optional: Use alternative methods to prevent multicast MOH streams from being sent over the IP WAN:
 a. Use IP ACL at the IP WAN router interface.
 b. Disable multicast routing on the IP WAN router interface.

Transcoding Configuration

1. Add a Transcoder Resource in CUCM
- Cisco IOS Enhanced Media Termination Point for the Cisco NM-HDV2, NM-HD-1V/2V/2VE, 2800 series, and 3800 series routers;

- Cisco IOS Media Termination Point in the Cisco NM-HDV and 1700 series routers;
- Cisco Media Termination Point Hardware for Cisco Catalyst WS-X6608-T1 and WS-X6608-E1;
- Cisco Media Termination Point WS-SVC-CMM.

2. Configure the Transcoder Resource in Cisco IOS Software
!
interface Loopback0
 ip address
!
voice-card 0
 dspfarm
 dsp service s dspfarm
!
sccp local Loopback0
 sccp ccm identifier 1 version 6.0
 sccp
!
sccp ccm group 1
 associate ccm 1 priority 1
 associate profile 1 register HQ-1 XCODER
!
dspfarm profile 1 transcode
 codec g711ulaw
 codec g711alaw
 codec g729ar8
 codec g729abr8
 maximum sessions 2
 associat e application SCCP
 no shutdown
!
#show sccp
#show sccp ccm group
#show dspfarm profile
#show sccp connections

SRST and MGCP Fallback Configuration

Availability Options:
- PSTN backup: Use the PSTN as a backup for on-net intersite calls.
- MGCP fallback: Configure an MGCP gateway to fall back, and use the locally configured (POTS), H.323 or (SIP) dial peers when the connection to its call agent is lost.
- SRST: Cisco IP Phones using either SIP or SCCP must register to a call-processing device for the phones to work.
- Call Forward Unregistered (CFUR): This is a call-forwarding configuration for CUCM that becomes effective when the IP Phone is not registered.
- Automated Alternate Routing (AAR) and Call Forward on No Bandwidth (CFNB): AAR allows calls to be rerouted over the PSTN when calls over the IP WAN are not admitted by CAC. CFNB is a call-forwarding configuration of IP Phones, which becomes effective when AAR is used.
- Mobility solutions: CUCM Extension Mobility and the Device Mobility features.

SRST:

1. Define the SRST references for phones in CUCM Administration.
2. Configure the Call Forward Unregistered (CFUR) feature, and set the CFUR destination of lines on remote-site phones to the correct public switched telephone network (PSTN) number in CUCM Administration to enable reachable remote sites in SRST mode.
3. Apply to the Device Pool.
4. Configure SRST on gateway:
(config)#call-manager-fallback
(config-cm-fallback )#i p source-address ip-address [port port] [any-match | strict-match]
(config-cm-fallback )#max-dn max-directory-numbers [dual-line] [preference preference-order]
(config-cm-fallback)#max-ephones max-phones
Optional:
(config-cm-fallback)#limit-dn {7910 | 7935 | 7940 | 7960} max-lines
(config-cm-fallback)#keepalive seconds
5. Implement a simplified SRST dial plan on the remote-site gateways to ensure connectivity for remote-site phones in SRST mode.

MGCP Fallback:

(config)#ccm-manager fallback-mgcp
(config)#application
(config-app)#global
(config-app-global)#service alternate Default
(config)#voice service voip
(conf-voi-serv )#h323
--To provide call preservation during switchback
(conf-serv-h323)#no h225 timeout keepalive

CCME in SRST Mode:
(config)#telephony-service
(config-telephony )#srst mode auto-provision all
(config-telephony )#srst dn line-mode dual
(config-telephony )#srst ephone template 1
(config-telephony )#srst dn template 3
(config-telephony )#srst ephone description CCME SRST
(config-telephony)#max-ephone 10
(config-telephony)#max-dn 30
(config)#ephone-template 1
(config-ephone-template)#keep-conference local-only
(config)#ephone-dn-template 3
(config-ephone-template)#hold-alert 25 idle

MS OCS and Avaya PBX interconnection with Cisco ISR

Configuration:
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
!
isdn switch-type primary-qsig
!
voice-card 0
no dspfarm
!
voice rtp send-recv
!
voice service voip
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
session transport tcp
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
controller E1 0/0/0
pri-group timeslots 1-31
clock source line
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn incoming-voice voice
no cdp enable
!
voice-port 0/0/0:15
compand-type u-law
cptone RU
!
dial-peer voice 7100 voip
destination-pattern 75..
voice-class codec 1
session protocol sipv2
session target ipv4:192.168.1.1
session transport tcp
dtmf-relay rtp-nte
no vad
!
dial-peer voice 1 pots
incoming called-number .T
direct-inward-dial
!
dial-peer voice 2 pots
destination-pattern ....
port 0/0/0:15
forward-digits all
!
dial-peer voice 3 pots
destination-pattern .....
port 0/0/0:15
forward-digits all
!
dial-peer voice 4 pots
destination-pattern 90.........
port 0/0/0:15
forward-digits all
!
dial-peer voice 5 pots
destination-pattern 9.......
port 0/0/0:15
forward-digits all
!

CallManager Express Configuration

Coping files to router:
Router(config)# archive tar /xtract tftp://192.168.1.1/cme-full-4.3.0.0.tar flash:

TFTP server activation on router:
Router(config)# tftp-server flash:/P00308000500.bin alias P00308000500.bin

Config example:
!
telephony-service
 max-ephones 5
 max-dn 5
 ip source-address 192.168.1.1 port 2000
 auto assign 1 to 5
 system message Cisco
 load ata ATA030203SCCP051201A.zup
 time-format 24
 date-format dd-mm-yy
 moh music-on-hold.au
 web admin system name admin secret cisco
 dn-webedit
 time-webedit
!
ephone-dn 1 dual-line
 number 1001 secondary 4000
!
ephone 1
 mac-addres 0013.2442.6743
 button [line number][separator][ephone-dn]
!

Create configuration files:
Router(config-telephony)# create cnf-files

Diagnostic commands:
# show telephony-service tftp-bindings
# show ephone attempted-registrations
# show ephone
# show ephone summary
# debug tftp events
# debug ephone register

SIP Configuration

Sip Basic Call Flow:Configuration:

Router(config)#voice service voip
Router(conf-voi-serv)#sip
Router(conf-serv-sip)#session transport {udp|tcp}
Router(conf-serv-sip)#bind control source-interface Loopback 0
Router(conf-serv-sip)#bind media source-interface Loopback 0
Router(conf-serv-sip)#exit
Router(conf-voi-serv)#no shutdown
Router(config)#sip-ua
Router(config-sip-ua)#authentication username password
Router(config-sip-ua)#registrar dns:sip2.cisco.com expires 3600
Router(config-sip-ua)#sip-server dns:sip2.cisco.com
Router(config-sip-ua)#retry invite 2
Router(config-sip-ua)#retry response 2
Router(config-sip-ua)#retry bye 2
Router(config-sip-ua)#retry cancel 2

Dial-peer:
Router(config)#dial-peer voice 50 pots
Router(config-dial-peer)#destination-pattern 5.
Router(config-dial-peer)#session protocol sipv2
Router(config-dial-peer)#session target sip-server
Router(config-dial-peer)#dtmf-relay {rtp-nte|sip-notify}

Verifying Configuration:
# show sip service
# show sip-ua status
# show sip-ua register status
# show sip-ua timers
# show sip-ua connections
# show sip-ua calls
# show sip-ua statistics
# debug asnl events
# debug voip ccapi inout
# debug voip ccapi protoheaders
# debug ccsip all

MGCP Configuration

MGCP Basic Call Flow:Comands:
AUEP - Audit Endpoint (status of endpoint)
AUCX - Audit Connection (status)
CRCX - Create Connection (by CA)
DLCX - Delete Connection (be CA,GW)
MDCX - Modify Connection (by CA)
RQNT - Notification Reuest (by CA)
NTFY - Notification about Event (by GW)
RSIP - Restart in progress (by GW)

Configuration:

Automatic configuration:
! Call agent - ccm
Router(config)#ccm-manager config {ip address}
Router(config)#ccm-manager config

Residential Gateway:
! Call agent - ccm
Router(config)#ccm-manager mgcp
Router(config)#mgcp
Router(config-mgcp)#mgcp call-agent 10.10.10.50 service-type mgcp
Router(config)#dial-peer voice 1 pots
Router(config-dialpeer)#application mgcpapp
Router(config-dialpeer)#port 1/0/0
Router(config)#dial-peer voice 2 pots
Router(config-dialpeer)#application mgcpapp
Router(config-dialpeer)#port 1/0/1
Router(config-dialpeer)#exit
Router(config)#mgcp package-capability dtmf-package
Router(config)#mgcp package-capability gm-package
Router(config)#mgcp package-capability line-package
Router(config)#mgcp package-capability rtp-package
Router(config)#mgcp default-package line-package

Trunk Gateway:
Router(config)#ccm-manager mgcp
Router(config)#mgcp
Router(config)#mgcp call-agent 10.10.10.50 service-type mgcp
Router(config)#controller E1 0/0/0
Router(config-controller)#ds0-group 1 timeslots 1-31 type none service mgcp

Fax Relay and Pass-Through:
Router(config)#no ccm-manager fax protocol cisco
Router(config)#mgcp pakage-capability fxr-pakage
Router(config)#mgcp pakage-capability rtp-pakage
Router(config)#mgcp fax rate 14400
Router(config)#mgcp timer 300
Router(config)#mgcp fax-relay sg3-to-g3

Verifying Configuration:
# show mgcp
# show ccm-manager
# show mgcp endpoint
# show mgcp statistics
# debug voip ccapi inout
# debug mgcp [ all | errors | events | packets | parser ]

H.323 Configuration

H.323 Elements:
- H.245 - codecs and features negotiation
- H.225 - call setup
- H.225 RAS - gatekeeper (Registration, Admition, Status)

H.323 Basic Call Flow:
Configuration:

! Enable h323 VOIP servises
Router(config)#voice service voip
Router(conf-voi-serv)#h323
! specifies the time to maintain an established H.225 TCP connection when there are no calls on that connection
Router(conf-serv-h323)#h225 timeout tcp call-idle never
!
Router(config)#interface loopback 0
Router(config-if)#ip address 10.10.1.1 255.255.255.0
Router(config-if)#h323-gateway voip interface
Router(config-if)#h323-gateway voip h323-id gw1
Router(config-if)#h323-gateway voip bind srcaddr 10.10.1.1
! Codecs configuration
Router(config)#voice class codec 100
Router(config-class)#codec preference 1 g711alaw
Router(config-class)#codec preference 2 g729br8
!
Router(config)#voice class h323 600
! TCP establish timeout value in seconds. Default: 15
Router(config-class)#h225 timeout tcp establish 10
! sets the timeout value, in seconds, for the response of the outgoing SETUP message. Default: 15
Router(config-class)#h225 timeout setup 10
!
Router(config)#dial-peer voice 500 voip
Router(config-dial-peer)#voice-class codec 100
Router(config-dial-peer)#voice-class h323 600
Router(config-dial-peer)#dtmf-relay {[cisco-rtp] [h245-alphanumeric] [h245-signal] [rtp-nte [digit-drop]] [sip-notify]}

Fax Pass-Through (G.711, No VAD, No Echo Cancelation):
Router(config)voice service voip
Router(config-serv)#fax protocol pass-through
or
Router(config)#dial-peer voice 50 voip
Router(config-dial-peer)#destination-pattern 50
Router(config-dial-peer)#session target ipv4:10.10.10.50
Router(config-dial-peer)fax protocol pass-through g711alaw
Router(config-dial-peer)#fax rate 14400

Fax T.38 Relay:
Router(config)#voice service voip
Router(conf-voi-serv)#fax protocol {t38|cisco}
!
Router(config)#dial-peer voice 51 voip
Router(config-dial-peer)#destination-pattern 51
Router(config-dial-peer)#session target ipv4:10.10.10.50
! Disable fax-relay Error Correction Mode
Router(config-dial-peer)#fax-relay ecm disable
! Enable the fax stream between two SG3 fax machines to negotiate down to G3 speeds
Router(config-dial-peer)#fax-relay sg3-to-g3
Router(config-dial-peer)#fax rate 14400

DTMF Relay:
Router(config-dial-peer)#dtmf-relay h245-alphanumeric

Verifying Configuration:
#show gateway
#debug cch323 h225
#debug cch323 h245
#debug h225
#debug h245
#debug voip ccapi inout

Digital Voice Ports Configuration

QSIG:
card type e1 0 0
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
!
isdn switch-type primary-qsig
!
controller E1 0/0/0
framing CRC4
pri-group timeslots 1-31
clock source line
linecode hdb3
!
interface Serial0/0/0:15
no ip address
isdn switch-type primary-qsig
! incoming calls to DSP
isdn incoming-voice voice
! for countries with variabe length numbering plan
isdn overlap-receiving
!user - TE (default); network - NT
isdn protocol-emulete user | network
isdn bchan-number-order descending round-robin
no cdp enable
!
voice-port 0/0/0:15
cptone ru
compand-type a-law
!
end

PSTN NET5:

card type e1 0 0
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
!
isdn switch-type primary-net5
!
controller E1 0/0/0
pri-group timeslots 1-31
clock source line
!
interface Serial0/0/0:15
no ip address
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
!
voice-port 0/0/0:15
!
end

MGCP PRI:

ccm-manager mgcp
!
card type e1 0 0
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
!
controller E1 0/0/0
pri-group timeslots 1-31 type none service mgcp
clock source line
!
interface Serial0/0/0:15
no ip address
isdn switch-type primary-ni
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
!
voice-port 0/0/0:15
!
mgcp
mgcp call-agent 10.1.5.2 service-type mgcp version 0.1
!
mgcp profile default
!
dial-peer voice 1 pots
application mgcpapp
port 1/0:23
!
end

Verifying Voice Ports:

! Voice ports
# show voice port summary
# show voice port
# show voice dsp
! E1
# show controllers e1 0/0/0
# show voice call summary
# show call active voice
# show call history voice
# show network clocks
! ISDN
# show isdn status
# show isdn active
# show dialer
! Debug
# debug isdn q921
# debug isdn q931
# debug isdn event
# debug isdn error

Analog Voice Ports Configuration

FXS ports:

Router(config)# voice-port slot/port
!port signaling
Router(config-voiceport)# signal {loopstart | groundstart}
Router(config-voiceport)# no shutdown
Router(config-voiceport)# description
!call progress tones
Router(config-voiceport)# cptone locale
!ring interval
Router(config-voiceport)# ring cadence {pattern-number | define pulse interval}
!frequency on the ring
Router(config-voiceport)# ring frequency {25 | 50}
Router(config-voiceport)# station-id name
Router(config-voiceport)# station-id number
Router(config-voiceport)# caller-id enable
!
 dial-peer voice pots
 destination-pattern
 port
 forward-digits all
!

FXO ports:

!FXO trunks does not support DID, because of DNIS only for outgoing calls to PSTN not incoming. Incoming ANI is supported.

Router(config)# voice-port 0/0/0
Router(config-voiceport)# signal {loop-start | ground-start}
Router(config-voiceport)# dial-type {dtmf | pulse}
!number of calls before hang-up
Router(config-voiceport)# ring number


Timers:

Router(config)# voice-port 0/1/0
!how long the dial tone is presented before the first digit
Router(config-voiceport)# timeouts initial
!wait before sending digits; default is 10 seconds
Router(config-voiceport)#timeouts interdigit
!default is 180 seconds
Router(config-voiceport)#timeouts ringing
!DTMF digit signal duration
Router(config-voiceport)#timing digit
!DTMF interdigit signal duration
Router(config-voiceport)#timing interdigit
!maximum duration of a hookflash indication
Router(config-voiceport)#timing {hookflash-in | hookflash-out}

Verifying Voice Ports:

# show voice port summary
# show voice port [slot/subunit/port]
# show voice busyout
# show voice dsp
# show voice call summary
# show call active voice
# show call history voice

# test voice port
# csim start

DSP

1 DSP C5510 = PVDM2-16

DSP functions:

- Transcoding
- Voice Termination
- MTP
- Conferencing

Codec Compleity:

- Medium: G.728, G.723, G.729, G.729B, fax relay
C549: 4, C5510: 8 voice/fax relay calls per DSP.
- High: G.711, G.726, G.729A, G.729AB, fax relay
C549: 2, C5510: 6 voice/fax relay calls per DSP.
- Flex: optimal mode.
- Secure: support of secure RTP.

Router(config)# voice-card 1
Router(config-voicecard)# codec complexity flex
! Verifying dsp
Router# show voice dsp

Media Resources Configuration:

! DSP farm config
Router(config)# voice-card
Router(config-voicecard)# dsp services dspfarm
! DSP profile config
Router(config)# dspfarm profile {conference | mtp | transcode}
Router(config-dspfarm-profile)# codec { | pass-through}
Router(config-dspfarm-profile)# maximum sessions
Router(config-dspfarm-profile)# associate profile sccp
! SCCP config
Router(config)# sccp local FastEthernet 0/1
Router(config)# sccp ccm identifier 1 priority 1 version
Router(config)# sccp
! SCCP group config
Router(config)#sccp ccm group 1
Router(config-sccp-ccm)#bind interface FastEthernet0/1
Router(config-sccp-ccm)#associate ccm 1 priority 1
Router(config-sccp-ccm)#associate profile 1
Router(config-sccp-ccm)#register
!
!Verifying config
Router# show dspfarm dsp all
Router# show dspfarm profile