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The following is an example config for using a Cisco Integrated Services Router - in this case, a 3945 with IOS15.1 - serving as a gateway to the PSTN. The following configuration:

  1. Is US-centric regarding dial plan;
  2. Has 3 ISDN-PRI's in an NFAS group for the PSTN connectivity;
  3. Includes config to utilize Cisco's Survivable Remote Site Telephony for sipX phones when this router is located in a Branch;
  4. Includes config to interoperate with 3 sipX servers composing a single cluster (tweak to your needs);
  5. Will generate OPTIONS pings during calls to ensure media path is alive;
  6. Includes examples for ipv4 address trust lists, which must be configured to enable your sipX servers and phones to route calls through the gateway;
  7. Does NOT include routing, authentication, QoS, or other config - this is the sipX interoperation only.

In the example below, values offset with <value> will be specific to your implementation. You also have the option of enabling TCP or UDP for signaling, as noted by <tcp|udp> in the config - select one.

In sipX, the router should be set up as an unmanaged gateway; and the phones set with the router's FQDN or IP address as their secondary registration server in order for SRST to work. You will have to test registration timers for both Primary and Secondary in Phone Groups to make sure SRST works to your liking.

version 15.1

card type t1 0 0
card type t1 0 1
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-select 1 T1 0/0/0

ip dhcp pool VOIP
 network <voip.subnet> <netmask>
 option 66 ascii "tftp://<sipx.ip>"
 default-router <router.ip>
 dns-server <dns1.ip> <dns2.ip>
 domain-name <>
 lease 0 8


ip domain retry 0
ip domain timeout 2
ip domain name <>
ip host <sipx1.fqdn> <sipx1.ip.addr>
ip host <sipx2.fqdn> <sipx2.ip.addr>
ip host <sipx3.fqdn> <sipx3.ip.addr>
ip host _sip._udp.<sipx1.fqdn> srv 1 0 5060 <sipx1.fqdn>
ip host _sip._udp.<sipx2.fqdn> srv 2 0 5060 <sipx2.fqdn>
ip host _sip._udp.<sipx3.fqdn> srv 3 0 5060 <sipx3.fqdn>
ip host _sip._tcp.<sipx1.fqdn> srv 1 0 5060 <sipx1.fqdn>
ip host _sip._tcp.<sipx2.fqdn> srv 2 0 5060 <sipx2.fqdn>
ip host _sip._tcp.<sipx3.fqdn> srv 3 0 5060 <sipx3.fqdn>
isdn switch-type primary-ni
trunk group pri
 description Hunt group for pri ports
 hunt-scheme longest-idle
 translation-profile incoming FROM-PRI
voice-card 0
 dsp services dspfarm
voice service pots
voice service voip
 ip address trusted list
  ipv4 <sipx1.ip.addr> <netmask>
  ipv4 <sipx2.ip.addr> <netmask>
  ipv4 <sipx3.ip.addr> <netmask>
  ipv4 <phone.vlan.subnet>
 clid substitute name
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 fax protocol none
  no call service stop
 modem passthrough nse codec g711ulaw
  bind control source-interface <interface>
  bind media source-interface <interface>
  session transport <tcp|udp>
  min-se 120 session-expires 120
  registrar server expires max 600 min 60
  options-ping 300
voice class codec 4
 codec preference 1 g711alaw
 codec preference 2 g711ulaw
 codec preference 3 g729r8

voice register global

 max-dn 80

 max-pool 100


voice translation-rule 1
 rule 1 /<telco.digits>/ /<extn.range>\1/
voice translation-rule 3
 rule 1 /\(^.*\)/ /1\1/ type any unknown plan any unknown
 rule 2 /\(^.*\)/ /011\1/ type international international
voice translation-rule 10
 rule 1 /^1\(.*\)/ /\1/
voice translation-rule 666
 rule 1 reject /<some.rule>/
voice translation-profile CLID_PREPEND
 translate calling 3
voice translation-profile FROM-PRI
 translate called 1
voice translation-profile TO_PSTN
 translate calling 10
voice translation-profile call_block
 translate calling 666
controller T1 0/0/0
 cablelength long 0db
 pri-group timeslots 1-24 nfas_d primary nfas_int 0 nfas_group 1
controller T1 0/0/1
 cablelength long 0db
 pri-group timeslots 1-24 nfas_d backup nfas_int 1 nfas_group 1
controller T1 0/1/0
 cablelength long 0db
 pri-group timeslots 1-24 nfas_d none nfas_int 2 nfas_group 1
interface Serial0/0/0:23
 no ip address
 encapsulation hdlc
 isdn switch-type primary-ni
 isdn incoming-voice voice
 isdn supp-service name calling
 trunk-group pri
 no cdp enable
ip dns view default
 domain timeout 2
 domain retry 0
ip rtcp report interval 5003
voice-port 0/0/0:23
voice-port 0/1/0:23
voice-port 0/0/1:23
no ccm-manager fax protocol cisco
dial-peer voice 911 pots
 trunkgroup pri
 description Emergency Services
 preference 1
 destination-pattern 9911
 progress_ind alert enable 8
 progress_ind progress enable 8
 forward-digits 3
dial-peer voice 9000 voip
 description sipx
 call-block translation-profile incoming call_block
 preference 3
 destination-pattern <extension.range>
 progress_ind setup enable 3
 modem passthrough nse codec g711ulaw
 session protocol sipv2
 session target sip-server
 session transport <tcp|udp>
 incoming called-number .
 dtmf-relay rtp-nte sip-notify sip-kpml
 codec g711ulaw
 fax rate 14400
 fax protocol t38 version 0 ls-redundancy 5 hs-redundancy 2 fallback none
 no vad
dial-peer voice 91 pots
 trunkgroup pri
 description Outgoing Calls
 translation-profile incoming CLID_PREPEND
 translation-profile outgoing TO_PSTN
 preference 1
 destination-pattern 9T
 progress_ind alert enable 8
 progress_ind progress enable 8
 incoming called-number .
 timer receive-rtp 1200
 retry invite 2
 retry bye 2
 retry cancel 2
 retry notify 6
 retry options 4
 timers trying 100
 timers notify 100
 timers buffer-invite 500
 timers options 1000
 sip-server dns:<sipx.domain>
 refer-ood enable

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