04-14-2022 10:00 AM
Hello,
I'm trying to get a old IAD trunked to my PBX to enable some skinny phones to call out.
I've got registration working to the PBX but not calling.
I can call between my two phones (extensions 100& 101) just fine but when I try to call the PSTN I get a fast busy.
I've tried debugging ccsip messages when I place a call and do not see any output.
I'm lost here, I'm the PBX guy not the cisco guy (we don't have one of those).
Config below:
Test2431-16FXS#show run Building configuration... Current configuration : 5086 bytes ! version 12.4 service timestamps debug datetime msec localtime service timestamps log datetime msec localtime service password-encryption ! hostname Test2431-16FXS ! boot-start-marker boot-end-marker ! card type t1 1 logging message-counter syslog logging buffered 4096 ! no aaa new-model clock timezone MST -7 clock summer-time MDT recurring network-clock-participate T1 1/0 network-clock-select 1 T1 1/0 ! ! ip source-route ! ! ip cef ! ip dhcp pool PHONES network 192.168.100.0 255.255.255.0 default-router 192.168.100.1 option 150 ip 192.168.100.1 ! ! ! no ipv6 cef multilink bundle-name authenticated ! ! voice hunt user-busy voice call convert-discpi-to-prog voice call carrier capacity active voice rtp send-recv ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip no supplementary-service h450.2 no supplementary-service h450.3 no supplementary-service sip moved-temporarily no supplementary-service sip refer redirect ip2ip fax protocol pass-through g711ulaw h323 modem passthrough nse codec g711ulaw redundancy sip bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 no call service stop ! voice-card 0 ! ! ! archive log config hidekeys ! ! controller T1 1/0 channel-group 0 timeslots 1-24 ! interface Loopback0 no ip address ! interface FastEthernet0/0 ip address dhcp duplex auto speed auto ! interface FastEthernet0/1 ip address 192.168.100.1 255.255.255.0 duplex auto speed auto ! interface Serial1/0:0 no ip address ! no ip http server no ip http secure-server ! ip forward-protocol nd ! ! logging source-interface Loopback0 ! tftp-server flash:apps45.9-4-2ES26.sbn tftp-server flash:cnu45.9-4-2ES26.sbn tftp-server flash:cvm45sccp.9-4-2ES26.sbn tftp-server flash:dsp45.9-4-2ES26.sbn tftp-server flash:jar45sccp.9-4-2ES26.sbn tftp-server flash:SCCP45.9-4-2SR3-1S.loads tftp-server flash:term65.default.loads tftp-server flash:B016-1-0-4-2.SBN ! ! control-plane ! ! ! voice-port 2/0 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/1 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/2 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/3 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/4 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/5 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/6 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/7 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/8 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/9 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/10 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/11 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/12 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/13 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/14 no supervisory disconnect lcfo no battery-reversal ! voice-port 2/15 no supervisory disconnect lcfo no battery-reversal ! ! ! ! dial-peer voice 1 voip description 10 Digit Dial destination-pattern ..........T session protocol sipv2 session target dns:f0fcdf.net codec g711ulaw ! dial-peer voice 30 voip description 11 Digit Dial destination-pattern ........... session protocol sipv2 session target sip-server codec g711ulaw ! dial-peer voice 100 voip description Catch All Dial destination-pattern .T session protocol sipv2 session target sip-server codec g711ulaw ! ! sip-ua credentials username user_XXXX password 7 XXXXXXX realm f0fcdf.net keepalive target dns:f0fcdf.net authentication username user_XXXXXX password 7 xxxxxxxx registrar dns:f0fcdf.net expires 3600 sip-server dns:f0fcdf.net ! ! ! telephony-service max-ephones 10 max-dn 30 ip source-address 192.168.100.1 port 2000 service phone webAccess 0 service dnis overlay service dnis dir-lookup service dss timeouts interdigit 4 system message CedarNetworks cnf-file perphone load 7916-24 B016-1-0-4-2 load 7965 SCCP45.9-4-2SR3-1S.loads time-zone 6 max-conferences 8 gain -6 call-forward pattern .T moh flash:music-on-hold.au transfer-system full-consult transfer-pattern .T create cnf-files version-stamp 7960 Apr 04 2022 12:38:28 ! ! ephone-dn 10 number 100 label x100 ! ! ephone-dn 11 number 101 ! ! ephone 1 device-security-mode none mac-address C025.5C43.0CDD type 7965 button 1:10 ! ! ! ephone 2 device-security-mode none mac-address 2893.FE13.4C0A type 7965 button 1:11 ! ! ! line con 0 line aux 0 line vty 0 4 login ! end
Solved! Go to Solution.
04-14-2022 10:49 PM - edited 04-14-2022 10:51 PM
You can do that with this.
voice translation-rule 101 rule 1 /\(....\)$/ /\1/
This rule will keep the last four digits and carry that over to the replace side where it is referenced with the \1 “command”.
For more information on voice translation rule please see this document.
04-14-2022 11:21 AM
Update, I deleted the dial-peers except for peer 1 and I can now call out.
How can I route my incoming DID's to the phones?
04-14-2022 12:20 PM - edited 04-14-2022 01:54 PM
OK, so with a translation rule I have incoming calls to 5202764508 and to 5203857734.
voice translation-rule 101 rule 1 /5202764508/ /101/ rule 2 /5203857734/ /100/ ! ! voice translation-profile INCOMING translate called 101
Is there a better way to do this? I'd like to get a range of DID's and rewrite to the last 4 digits.
04-14-2022 10:49 PM - edited 04-14-2022 10:51 PM
You can do that with this.
voice translation-rule 101 rule 1 /\(....\)$/ /\1/
This rule will keep the last four digits and carry that over to the replace side where it is referenced with the \1 “command”.
For more information on voice translation rule please see this document.
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