[Asterisk] Connecting Asterisk PBXs to a SmartNode Patton 4960 Voice Gateway with Full Failover Features
The better way to interconnect your Asterisk based PBXs to the PSTN infrastructure is by a dedicated Voice Gateway.
Thanks to, your telephony infrastructure, will benefit of a rich bunch of features:
- Dedicated DSP equipment.
- Powerful transcoding capabilities.
- Advanced call-routing.
- Failover & Load Balancing.
Many of above were unable to deploy by a PCI Card!!!
With the following configuration you’re able to build a completely fault tolerant architecture with a single point of failure (represented by patton itself):
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 38 39 40 41 42 43 44 45 46 47 48 49 50 51 52 53 54 55 56 57 58 59 60 61 62 63 64 65 66 67 68 69 70 71 72 73 74 75 76 77 78 79 80 81 82 83 84 85 86 87 88 89 90 91 92 93 94 95 96 97 98 99 100 101 102 103 104 105 106 107 108 109 110 111 112 113 114 115 116 117 118 119 120 121 122 123 124 125 126 127 128 129 130 131 132 133 134 135 136 137 138 139 140 141 142 143 144 145 146 147 148 149 150 151 152 153 154 155 156 157 158 159 160 161 162 163 164 165 166 167 168 169 170 171 172 173 174 175 176 177 178 179 180 181 182 183 184 185 186 187 188 189 190 191 192 193 194 195 196 197 198 199 200 201 202 203 204 205 206 207 208 209 210 211 212 213 214 215 216 217 218 219 220 221 222 223 224 225 226 227 228 229 230 231 232 233 234 235 236 237 238 239 240 241 242 243 244 245 246 247 248 249 250 251 252 253 254 255 256 257 258 259 260 261 262 263 264 265 266 267 268 269 270 271 | cli version 3.20 clock local default-offset +00:00 dns-relay webserver port 80 language en sntp-client sntp-client server primary <NTP SERVER ip> port 123 version 4 system hostname tectton-primario system ic voice 0 system clock-source 1 e1t1 0 0 profile r2 default profile napt NAPT_WAN profile ppp default profile call-progress-tone IT_Dialtone play 1 200 425 -12 pause 2 200 play 3 600 425 -12 pause 4 1000 profile call-progress-tone IT_Alertingtone play 1 1000 425 -12 pause 2 4000 profile call-progress-tone IT_Busytone play 1 500 425 -12 pause 2 500 profile tone-set default profile tone-set IT map call-progress-tone dial-tone IT_Dialtone map call-progress-tone ringback-tone IT_Alertingtone map call-progress-tone busy-tone IT_Busytone map call-progress-tone release-tone IT_Busytone map call-progress-tone congestion-tone IT_Busytone profile voip default codec 1 g711alaw64k rx-length 20 tx-length 20 codec 2 g711ulaw64k rx-length 20 tx-length 20 dtmf-relay rtp rtp traffic-class local-default dejitter-max-delay 200 fax transmission 1 bypass g711alaw64k fax transmission 2 bypass g711ulaw64k fax transmission 3 relay t38-udp fax redundancy low-speed 2 high-speed 1 fax volume -10.0 fax dejitter-max-delay 300 fax bypass-method nse modem transmission 1 bypass g711alaw64k modem transmission 2 bypass g711ulaw64k no modem detection on-remote-fax-request modem bypass-method v150-vbd profile pstn default no echo-canceler-nlp no echo-canceler profile sip default no autonomous-transitioning profile dhcp-server DHCPS_LAN network <NETWORK IP> <NETWORK MASK> include 1 <DHCP START> <DHCP END> lease 2 hours default-router 1 <GATEWAY IP> domain-name-server 1 <DNS SERVER> profile aaa default method 1 local method 2 none context ip router interface WAN ipaddress <WAN IP ADDRESS> <WAN NETMASK> use profile napt NAPT_WAN tcp adjust-mss rx mtu tcp adjust-mss tx mtu interface LAN ipaddress <LAN IP ADDRESS> <LAN NETMASK> tcp adjust-mss rx mtu tcp adjust-mss tx mtu context ip router route 0.0.0.0 0.0.0.0 <GATEWAY IP> 0 context cs switch no digit-collection timeout no digit-collection terminating-char national-prefix 0 international-prefix 00 routing-table called-e164 RT_TO_SIP route default dest-service INBOUND routing-table called-e164 SIP_TO_RT route default dest-service OUTBOUND interface isdn IF_ISDN route call dest-table RT_TO_SIP use profile tone-set IT caller-name send-information-following inband-info accept force call-setup setup interface isdn IF_ISDN_BACKUP route call dest-table RT_TO_SIP use profile tone-set IT caller-name send-information-following inband-info accept force call-setup setup interface sip IF_SIP bind context sip-gateway GW_SIP route call dest-table SIP_TO_RT remote <ASTERISK IP> 5060 early-disconnect interface sip IF_SIP_BACKUP bind context sip-gateway GW_SIP route call dest-interface IF_ISDN remote <ASTERISK BACKUP IP> 5061 early-disconnect service hunt-group OUTBOUND drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_ISDN route call 2 dest-interface IF_ISDN_BACKUP service hunt-group INBOUND drop-cause normal-unspecified drop-cause no-circuit-channel-available drop-cause network-out-of-order drop-cause temporary-failure drop-cause switching-equipment-congestion drop-cause access-info-discarded drop-cause circuit-channel-not-available drop-cause resources-unavailable route call 1 dest-interface IF_SIP route call 2 dest-interface IF_SIP_BACKUP context cs switch no shutdown authentication-service AUTH_OB realm 1 asterisk username <ASTERISK USERID> password <ASTERISK PASSWORD> encrypted location-service ASTERISK domain 1 <ASTERISK IP> identity <ASTERISK USER> authentication outbound authenticate 1 authentication-service AUTH_OB username <ASTERISK USER> registration outbound registrar <ASTERISK IP> 5060 lifetime 3600 register auto retry-timeout on-system-error 10 retry-timeout on-client-error 10 retry-timeout on-server-error 10 call outbound location-service ASTERISK_BACKUP domain 1 <ASTERISK BACKUP IP> identity <ASTERISK USER> authentication outbound authenticate 1 authentication-service AUTH_OB username <ASTERISK USER> registration outbound registrar <ASTERISK BACKUP IP> 5060 lifetime 3600 register auto retry-timeout on-system-error 10 retry-timeout on-client-error 10 retry-timeout on-server-error 10 call outbound context sip-gateway GW_SIP interface IF_GW_SIP bind interface LAN context router port 5060 context sip-gateway GW_SIP bind location-service ASTERISK no shutdown context sip-gateway GW_SIP_BACKUP interface IF_GW_SIP_BACKUP bind interface LAN context router port 5061 context sip-gateway GW_SIP_BACKUP bind location-service ASTERISK_BACKUP no shutdown port ethernet 0 0 medium auto encapsulation ip bind interface WAN router no shutdown port ethernet 0 1 medium auto encapsulation ip bind interface LAN router no shutdown port e1t1 0 0 port-type e1 clock slave framing crc4 application long-haul encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side net bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN switch port e1t1 0 0 no shutdown port e1t1 0 1 port-type e1 clock slave framing crc4 application long-haul encapsulation q921 q921 permanent-layer2 uni-side auto encapsulation q931 q931 protocol dss1 uni-side net bchan-number-order ascending encapsulation cc-isdn bind interface IF_ISDN_BACKUP switch port e1t1 0 1 no shutdown |
The big picture:
Saving...










