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:
- Is US-centric regarding dial plan;
- Has 3 ISDN-PRI's in an NFAS group for the PSTN connectivity;
- Includes config to utilize Cisco's Survivable Remote Site Telephony for sipX phones when this router is located in a Branch;
- Includes config to interoperate with 3 sipX servers composing a single cluster (tweak to your needs);
- Will generate OPTIONS pings during calls to ensure media path is alive;
- Includes examples for ipv4 address trust lists, which must be configured to enable your sipX servers and phones to route calls through the gateway;
- 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 15.1card type t1 0 0card 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 <domain.name>
lease 0 8
!
ip domain retry 0ip domain timeout 2ip domain name <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 dspfarm 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 h323 no call service stop modem passthrough nse codec g711ulaw sip 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 0ip 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 direct-inward-dial 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 . direct-inward-dial!!gateway timer receive-rtp 1200!sip-ua 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 handle-replaces!!!gatekeeper shutdown