Follow the configs and some more information:
CME-A
- All incoming/outcoming calls to PSTN are made by a SIP Trunk (working fine)
- The problem is only with calls to dial-peer 102 (172.30.0.1)
- I tried to get only the relevant information
voice service voip
ip address trusted list
ipv4 172.40.0.0 255.255.255.0
ipv4 192.168.254.0 255.255.255.0
ipv4 172.20.0.0 255.255.255.0
ipv4 192.168.254.1
ipv4 172.30.0.0 255.255.255.0
ipv4 192.168.151.0 255.255.255.0
ipv4 201.33.210.0 255.255.255.0
ipv4 201.33.209.0 255.255.255.0
ipv4 10.10.51.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
h225 h245-address on-connect
call preserve
sip
rel1xx disable
min-se 86400 session-expires 86400
header-passing
early-offer forced
midcall-signaling passthru
g729 annexb-all
!
voice class codec 1
codec preference 1 g729br8
codec preference 2 g729r8
codec preference 3 g711ulaw
voice translation-rule 1
rule 1 /^.*/ /xxxxxxx/
!
voice translation-rule 2
rule 1 /0/ /011/
!
voice translation-rule 3
rule 1 /00/ /0/
!
!
voice translation-profile PSTN_Outgoing
translate calling 1
translate called 2
!
voice translation-profile SIP_DDD
translate calling 1
translate called 3
!
!
interface GigabitEthernet0/0.172
description ## Voice Lan ##
encapsulation dot1Q 172
ip address 172.40.0.1 255.255.255.0
dial-peer voice 1 voip
description ## Incoming Call from SIP Trunk ##
session protocol sipv2
session target sip-server
incoming called-number .
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
dial-peer voice 102 voip
description $$ CME-B ###
destination-pattern 64..
session protocol sipv2
session target ipv4:172.30.0.1
dtmf-relay rtp-nte
no vad
sip-ua
credentials username xxxxxx password bbbbbbbb realm sip.provider.net
authentication username xxxxxx password bbbbbbb
no remote-party-id
disable-early-media 180
retry invite 2
retry register 10
timers connect 100
registrar dns:sip.provider.net expires 300
sip-server dns:sip.provider.net
no transport tcp
!
telephony-service
no auto-reg-ephone
max-ephones 15
max-dn 15
ip source-address 172.40.0.1 port 2000
auto assign 1 to 30
time-zone 17
time-format 24
date-format dd-mm-yy
max-conferences 8 gain -6
call-forward pattern 20..
call-forward pattern 64..
transfer-system full-consult
transfer-pattern 64..
transfer-pattern 20..
secondary-dialtone 0
create cnf-files version-stamp 7960 Jun 27 2013 09:40:49
!
!
ephone-dn 7
number 3007
label 3007
name Teste
!
ephone 7
mac-address 588D.0972.6274
max-calls-per-button 2
type 6961
button 1:7
CME-B
- Connected to PSTN by an E1 card (working fine incoming/outcoming calls)
- Calls to dial-peer 130 (CME-A) goes fine;
CME-B#show run
card type e1 0 3
network-clock-participate wic 3
voice-card 0
dspfarm
dsp services dspfarm
voice service voip
ip address trusted list
ipv4 172.20.0.0 255.255.255.0
ipv4 172.30.0.0 255.255.255.0
ipv4 192.168.252.0 255.255.255.0
ipv4 10.10.51.0 255.255.255.0
ipv4 192.168.252.1
ipv4 172.40.0.0 255.255.255.0
ipv4 192.168.254.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface GigabitEthernet0/0.172
bind media source-interface GigabitEthernet0/0.172
registrar server expires max 1200 min 300
dial-peer voice 1 pots
description # Inbound PSTN E1 #
incoming called-number .
direct-inward-dial
dial-peer voice 2 voip
description ## Inbound SIP Trunk ##
incoming called-number .
direct-inward-dial
dial-peer voice 130 voip
description $$ CME-A ###
destination-pattern 30..
session protocol sipv2
session target ipv4:172.40.0.1 (CME-A)
dtmf-relay rtp-nte
no vad
telephony-service
no auto-reg-ephone
max-ephones 20
max-dn 15
ip source-address 172.30.0.1 port 2000
auto assign 1 to 30
url services http://172.30.0.2/voiceview/common/login.do
url authentication http://172.30.0.1/CCMCIP/authenticate.asp
cnf-file location flash:
cnf-file perphone
time-zone 17
time-format 24
date-format dd-mm-yy
dialplan-pattern 1 11xxxxxx.. extension-length 4 extension-pattern 64..
voicemail 8200
max-conferences 8 gain -6
call-park system application
call-forward pattern 89..
call-forward pattern 20..
call-forward pattern 30..
moh "music-on-hold.au"
dn-webedit
time-webedit
transfer-system full-consult
transfer-pattern 64..
transfer-pattern 82..
transfer-pattern 89..
transfer-pattern 20..
transfer-pattern 0T
transfer-pattern 30..
secondary-dialtone 0
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 7
number 6420
name TESTE
ephone 4
device-security-mode none
mac-address 0026.C763.CBE6
username "user4" password 1234
type CIPC
button 1:7
I'll get the debugs and will share with you in a while.
