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Cisco IAD 2431 SIP Trunking help needed.

SBjohnk
Level 1
Level 1

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
1 Accepted Solution

Accepted Solutions

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.



Response Signature


View solution in original post

3 Replies 3

SBjohnk
Level 1
Level 1

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?

SBjohnk
Level 1
Level 1

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.

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.



Response Signature