Hi all. Can anyone post a full sample config using a T1 PRI? I need something to compare with as I'm working on my first and I've got outbound calls working, but inbound is hosed. I've only been able to find snipets of a config with my google searches. Just looking for some direction on how others configure these things. |
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I suggest running 'debug
I suggest running 'debug isdn q931' as this will show you exactly how mant digits telco is sending.
ISDN debugs
So what do you mean, exactly, by "inbound is hosed?" When you dial in do you get a busy signal, telco message, etc?
I am happy to help as I've set up may T1 PRIs but I need more information. Have you run a 'debug isdn q931'?
I get a busy on inbound
I get a busy on inbound calls. I think the problem is that I was specifying 7 digits in my translation rule when I just found out that my telco is only passing 4. I'm guessing that would cause a problem although I am unable to test again until COB. Also, does a translation rule such as:
voice translation-rule 21
rule 1 /5551417/ /500/
accomplish the same thing as:
ephone-dn 12 dual-line
number 500 secondary 5551417
I am attempting to establish my DIDs as I may have a problem with that as well.
I will run that debug
I will run that debug tonight when I attempt the cut again. Did my other question about the commands make sense?
Post your configuration here
Post your configuration here so that we can see what may be missing.
If you want a configuration example, I suggesting reading these:
http://www.cisco.com/en/US/docs/ios/12_3t/dial/command/reference/dia_i2g...
http://www.cisco.com/en/US/tech/tk713/tk628/technologies_tech_note09186a...
http://www.cisco.com/en/US/docs/ios/12_0t/12_0t7/feature/guide/t1_vo_t6....
Thanks for helping me out.
Thanks for helping me out.
Alright, this is what I've been able to coble together so far:
version 12.4
parser config cache interface
parser config interface
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
service internal
service compress-config
service sequence-numbers
!
boot-start-marker
boot system flash uc500-advipservicesk9-mz.124-11.XW6
boot-end-marker
!
card type t1 0 2
logging buffered 4096
!
no aaa new-model
clock timezone CST -6
clock summer-time GMT recurring
network-clock-participate wic 2
network-clock-select 1 T1 0/2/0
!
ip cef
!
!
ip dhcp relay information trust-all
ip dhcp use vrf connected
ip dhcp excluded-address 10.1.2.1 10.1.2.99
!
ip dhcp pool phone
network 10.1.2.0 255.255.255.0
default-router 10.1.2.1
option 150 ip 10.1.2.1
!
!
stcapp ccm-group 1
stcapp
!
stcapp feature access-code
!
multilink bundle-name authenticated
isdn switch-type primary-ni
!
!
voice call send-alert
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
supplementary-service h450.12
sip
no update-callerid
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
!
!
!
!
!
!
!
!
!
voice register global
max-dn 160
max-pool 40
!
!
!
voice translation-rule 25
rule 1 /5554387/ /387/
!
voice translation-rule 1111
rule 15 /.*/ /5551417/
!
voice translation-rule 1112
rule 1 /^9/ //
!
voice translation-rule 2001
rule 1 /5551417/ /500/
!
voice translation-rule 2222
!
!
voice translation-profile AA_Profile
translate called 2001
!
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
!
voice translation-profile CallBlocking
translate called 2222
!
!
voice translation-profile DIDjs_Called_25
translate called 25
!
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate calling 1111
translate called 1112
!
!
voice-card 0
no dspfarm
!
!
!
!
controller T1 0/2/0
framing esf
linecode b8zs
pri-group timeslots 1-24
!
interface Serial0/2/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn sending-complete
!
voice-port 0/2/0:23
bearer-cap Speech
!
sccp local Loopback0
sccp ccm 10.1.2.1 identifier 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
!
dial-peer voice 2001 voip
description ** cue auto attendant number **
translation-profile outgoing PSTN_CallForwarding
destination-pattern 500
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 2003 voip
description ** cue auto attendant PSTN number **
translation-profile outgoing AA_Profile
destination-pattern 5551417$
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 100 pots
description ** incoming dial peer **
destination-pattern 9T
incoming called-number .%
direct-inward-dial
port 0/2/0:23
!
dial-peer voice 101 pots
corlist outgoing call-local
description ** T1 pots dial-peer **
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 5
destination-pattern 9[2-9]......
port 0/2/0:23
forward-digits 7
no sip-register
!
dial-peer voice 102 pots
corlist outgoing call-domestic
description ** T1 pots dial-peer **
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 5
destination-pattern 91[2-9]..[2-9]......
port 0/2/0:23
prefix 1
no sip-register
!
dial-peer voice 103 pots
corlist outgoing call-international
description ** T1 pots dial-peer **
translation-profile outgoing CALLER_ID_TRANSLATION_PROFILE
preference 5
destination-pattern 9011T
port 0/2/0:23
prefix 011
no sip-register
!
dial-peer voice 2005 pots
translation-profile incoming AA_Profile
incoming called-number 5551417
direct-inward-dial
!
dial-peer voice 3021 pots
description DIDjs
translation-profile incoming DIDjs_Called_25
incoming called-number 5554387
direct-inward-dial
port 0/2/0:23
!
!
no dial-peer outbound status-check pots
!
telephony-service
video
load 7971 SCCP70.8-2-2SR2S
max-ephones 40
max-dn 160
ip source-address 10.1.2.1 port 2000
max-redirect 20
auto assign 10 to 43
auto assign 5 to 8 type anl
calling-number initiator
service phone videoCapability 1
timeouts interdigit 3
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.1/voiceview/authentication/authenticate.do
time-zone 8
voicemail 600
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
moh music-on-hold.au
multicast moh 239.10.16.16 port 2000
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
transfer-pattern .T
secondary-dialtone 9
fac custom ephone-hunt join *3
!
ephone-dn 10 dual-line
number 387 secondary 5554387 no-reg primary
label 387
description John Smith
name John Smith
call-forward busy 600
call-forward noan 600 timeout 10
ephone-hunt login
!
!
!
ephone-hunt 1 longest-idle
pilot 421
list *, *, *, *, *, *, *, *, *, *
no-reg pilot
statistics collect
!
!
!
Delete dial-peer 3021 since
Delete dial-peer 3021 since all incoming calls are handled by dial-peer 100. Change dial-peer 100 to:
dial-peer voice 100 pots
description ** incoming dial peer **
incoming called-number .
direct-inward-dial
port 0/2/0:23
voice translation-rule 25 translates 5554387 to 387. you have 387 set on ephone-dn 10. Apply this translation-rule to dial-peer 100. Remove the secondary number from ephone-dn 10 and you should be set.
So I should write one
So I should write one translation rule with statements for all of my DIDs and apply it to dial peer 100?
Also, since my telco is only passing me 4 digits, should I change my translation rules to 4 digits and leave off the leading numbers like this:
voice translation-rule 25
rule 1 /4387/ /387/
If telco is passing four
If telco is passing four digits, and if each user has their own DID, then I would suggest setting the user's extension as their DID. i.e. telco sends 4387, end user's extension is 4387. That way inbound translations are not necessary.
Otherwise, stick with applying the translations to dial-peer 100.
Thank you for all of your
Thank you for all of your help.