Cisco UC – Gateways and SRST

By | 04/11/2014

Gateway

Gateways are Cisco routers that have Voice Ports to connect to PSTN network and PVDM cards to transcode.

Voice Cards

Pri E1 -> VWIC-1MFT-E1
Bri ports -> VIC2-2FXO
Analog devices (fax, mobile track) -> VIC-4FXS
Data ISDN -> WIC-2BRI

VWIC – Voice and WAN Interface Card (Voice + Data)
VIC – Voice Interface Card (Only Voice)
WIC – WAN Interface Card (Only Data)
VIC2 – The 2 means that it’s the second generation of that card.
VIC2-2FXO – The number after the first – defines the number of ports that the card have

PVDM

PVDM2-64 – 64 defines the capacity of DSPs

Each voice channel in the gateway will require one available DSP. If there are no available DSPs, the voice channel won’t work.

Dial-peer considerations

Dial-Peers on the gateways don’t remove digits of the destination patterns as default.

When calling a number that fits in several dial peer destination patterns, it will use the most specific one.

MGCP Configuration

@CUCM

1) Configure the gateway in Device > Gateway section. Define the protocol MGCP, domain name (including domain) and the card slots. Each card slot will need the Device Pool that will define the codecs that will use and several parameters that will be provided by the ISP.

@GW

2) Configure the gateway to get the configuration automatically from the CUCM server.

ccm-manager config server [CUCM-TFTP IP]
ccm-manager config

Configuration automatically generated from the CUCM:

controller e1 0/1/0
 pro-group timeslots 1-31 service mgcp

voice-port 0/1/0:15

ccm-manager redundant-host [SUBSCRIBER IP]
ccm-manager mgcp
mgcp
mgcp call-agent [PUBLISHER IP] 2427 service-type mgcp version 0.1

interface serial 0/1/0:15
 isdn switch-type primary-net5 //depending on the configured parameters provided by the ISP
 isdn bind-l3 ccm-manager

dial-peer voice 999020 pots
 service mgcpapp
 port 0/2/0

3) When reseting the CUCM gateway, the configuration will be downloaded again and all the changes done on it will be overriden. For this reason, it’s recommended to remove the two initial commands after the configuration is downloaded.

no ccm-manager config server [CUCM-TFTP IP]
no ccm-manager config

4) We should also modify the PRI channels in use under the controller E1 to define the actual quantity of channels that the ISP is providing, but there are other commands that needs to be removed disabled before doing this change.

voice-port 0/1/0:15
 shutdown
interface serial 0/1/0:15
 no isdn bind-l3 ccm-manager
controller e1 0/1/0
 pro-group timeslots 1-4 service mgcp
voice-port 0/1/0:15
 no shutdown
interface serial 0/1/0:15
 isdn bind-l3 ccm-manager

Troubleshooting commands

show ccm manager 
show mgcp
show mgcp endpoints
debug mgcp all

H.323 Configuration

@GW

1) Voice ports should be configured using the ISP provided parameters, time-slots, clock,…

2) Dial-Peers should be configured in the GW to define where the calls should be redirected to voice ports

Forward land line calls through the PRI first or FX0 second

voice translation-rule 922551
 rule 1 /^1/ /922552/
voice translation-profile toPri
 translate calling 922551
dial-peer voice 8900 pots
 destination-pattern 0[89]........
 port 0/1/0:15
 translation-profile outgoing topri

dial-peer voice 8901 pots
 destination-pattern 0[89]……..
 port 0/2/0
 preference 5

Forward mobile calls through the FX0 first and second through the PRI

dial-peer voice 6700 pots
 destination-pattern 0[67]……..
 port 0/1/0:15
 preference 5
dial-peer voice 6701 pots
 destination-pattern 0[67]……..
 port 0/2/0

Forward emergency calls through the PRI

dial-peer voice 112 pots
 destination-patter 112
 port 0/1/0:15
 forward-digits all
dial-peer voice 113 pots
 destination-pattern 0112
 port 0/1/0:15
 forward-digits 3

3) Configuration of the dial-peer to forward calls to the CUCM for devices connected to it

dial-peer voice 2000 voip
 destination-pattern DN..
 session target ipv4:[PUBLSIHER IP]

dial-peer voice 2001 voip
 destination-pattern DN..
 session target ipv4:[SUBSCRIBER IP]
 preference 5

4) Configuration to make the Gateway connect to CUCM using the loopback interface

interface loopback.0
 h323-gateway voip interface
 h323-gateway voip bind srcaddr [LOOPBACK IP]

Gateway dial-peers will use H.323 and G.729 as default.

@CUCM

5) Configure the H323 gateway in Device > Gateway section. The name of the device should be the loopback IP of the gateway.

6) Configure a route pattern in Call routing > Route Hunt > Route Patterns so the CUCM knows which calls should be forwarded to the H323 gateway.

Troubleshooting commands

debug h225 q931

SIP Configuration

@CUCM

1) Create a Device > Trunk setting up SIP as device protocol, the name, device pool, the GW loopback IP, SIP Profile (Standard) and SIP trunk security profile (Non Secure)

2) Create a route pattern in Call routing > Route Hount > Route Patterns to define the calls that will be forwarded to the SIP gateway.

@GW

3) Create one dial peer for each CUCM box defining the destination pattern and the protocol. Subscriber will have a preference to be penalized.

dial-peer voice 2000 voip
 destination-pattern 20..
 session target ipv4:10.2.1.1
 session protocol sip
dial-peer voice 2001 voip
 destination-pattern 20..
 session target ipv4:10.2.1.2
 preference 5
 session protocol sipv2

4) Configure the loopback interface as a source interface for the SIP protocol.

voice service voip
 sip
  bind all source-interface loopback0

Troubleshooting commands

debug cssip messages

Router general commands

Voice ports and hardware information

show inventory
show diag

Troubleshooting commands

show dial-peer voice summary
show dialplan number XXXXX
show voice port summary 
show controller e1
show isdn status 
debug vpm signal (for fxo)
debug isdn q931 (for PRI)
csim start EXTENSION // command to start a call from the gateway

SRST

SRST (Survivable Remote Site Telephony) configuration allows the phones to register to the local gateway in case of lose of connectivity to the CUCM. Gateway perform a simple Call Manager Express functions.

@SRST GW

1) Configure SRST options

call-manager-fallback
 max-ephones 5
 max-dn 10
 ip source-address [SRST GW IP for the phones]
 system message primary [Message to shown in the phone screen]

“max-ephones ?” and “max-dn ?” will provide information about the maximum number of phones and DNs that can be configured with the current license.

When in Survivable mode, phones will kept the current extensions.

@CUCM

2) Definition of the SRST system in the CUCM in System > SRST. Setup a name, the GW IP and the port 2000.

3) Configuration of the SRST availability on the Device Pool that the remote sites are using. It’s possible to specify the SRST created in the previous point or it can be setup so the phones will always use their gateway as a SRST. In that case, point 2) is not needed.

Usually the number of allowed SRST phones is limited and it’s not possible to cover all the setup phones. It’s required to create two different Device Pools for the remote site to determine which phones will be capable to connect to the SRST and which phones won’t.