25 January 2010

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

13 January 2010

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

08 December 2009

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

01 July 2009

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

25 June 2009

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

22 June 2009

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

19 June 2009

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