We have a TA 904 connected to an Avaya PBX with a PRI interface to the Adtran 904. To the PSTN is SIP using registration. Everything looks good and incoming calls are working and reaching the Avaya PBX with success, so the SIP part to the carrier is ok and the PRI interface sends the calls to the Avaya also. The issue is with outbound calls. The TA 904 doesn't even try to make the SIP call to the PSTN. Everything looks good in the isdn messaging but we see the TA 904 sending a disconnect to the Avaya with message A9 Cause:41 (TEMPORARY_FAILURE). In the show voice history detail I see caller id and caled id ok and the destination ports ok. Here is the config:
hostname "XXXX"
clock timezone -4-Caracus
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ip subnet-zero
ip classless
ip routing
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!
!
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no auto-config
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event-history on
no logging forwarding
no logging console
no logging email
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service password-encryption
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no ip firewall alg msn
no ip firewall alg mszone
no ip firewall alg h323
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!
!
!
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no dot11ap access-point-control
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!
!
!
!
!
!
!
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interface eth 0/1
ip address x.x.x.x 255.255.255.252
media-gateway ip primary
no shutdown
!
!
!
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interface t1 0/1
lbo short 3
tdm-group 1 timeslots 1-24 speed 64
shutdown
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interface t1 0/2
description T1 to Client PBX
tdm-group 1 timeslots 1-24 speed 64
no shutdown
!
!
interface pri 1
description T1 PRI to Client PBX
role network b-channel-restarts enable
isdn name-delivery setup
connect t1 0/2 tdm-group 1
no shutdown
!
!
interface fxs 0/1
no shutdown
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interface fxs 0/2
no shutdown
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interface fxs 0/3
no shutdown
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interface fxs 0/4
no shutdown
!
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isdn-group 1
max-channels 24
connect pri 1
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!
!
!
!
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!
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ip route 0.0.0.0 0.0.0.0 y.y.y.y 20
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no tftp server
no tftp server overwrite
http server
no http secure-server
no snmp agent
no ip ftp server
no ip scp server
no ip sntp server
!
!
!
!
!
!
!
!
sip
sip udp 5060
no sip tcp
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!
!
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voice feature-mode network
voice flashhook mode transparent
voice forward-mode network
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!
!
!
!
!
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voice dial-plan 1 local NXX-NXX-XXXX
voice dial-plan 3 long-distance 1-NXX-NXX-XXXX
voice dial-plan 4 international 011$
!
!
!
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voice codec-list Trunk
codec g711ulaw
codec g711alaw
codec g729
!
!
!
voice trunk T01 type sip
description "SIP"
sip-server primary test.net
registrar primary test.net
outbound-proxy primary z.z.z.z
conferencing-uri "t"
domain "test.net"
register sipuser auth-name "sipuser" password encrypted "pwd"
register 7872222222 auth-name "sipuser" password encrypted "pwd"
register 7873333333 auth-name "sipuser" password encrypted "pwd"
register 7874444444 auth-name "sipuser" password encrypted "pwd"
register 7875555555 auth-name "sipuser" password encrypted "pwd"
codec-list Trunk both
authentication username "sipuser" password encrypted "pwd"
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voice trunk T02 type isdn
description "PRI-1 to PBX Customer_1"
resource-selection circular descending
connect isdn-group 1
no early-cut-through
t38
rtp delay-mode adaptive
codec-list Trunk
!
!
voice grouped-trunk SIP
trunk T01
accept NXX-NXX-XXXX cost 0
accept 1-NXX-NXX-XXXX cost 200
accept 1-800-NXX-XXXX cost 200
accept 1-888-NXX-XXXX cost 200
accept 1-877-NXX-XXXX cost 200
accept 1-866-NXX-XXXX cost 200
accept 1-855-NXX-XXXX cost 200
accept 1-900-NXX-XXXX cost 200
accept 1-976-NXX-XXXX cost 200
accept 976-XXXX cost 200
accept 011-$ cost 200
accept 411 cost 200
accept 611 cost 200
accept 911 cost 200
accept 0-NXX-NXX-XXXX cost 200
accept 10-10-XXX-$ cost 200
accept NXX-976-XXXX cost 200
accept $ cost 0
!
!
voice grouped-trunk "PRI TO PBX CUST_1"
trunk T02
accept 7872222222 cost 0
accept 7873333333 cost 0
accept 7874444444 cost 0
accept 7875555555 cost 0
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!
!
!
!
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sip registrar
sip registrar default-expires 120
sip registrar realm nnpr.net
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!
!
!
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sip timer registration-failure-retry 10
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sip grammar request-uri host domain
sip grammar from host domain
sip grammar to host domain
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!
!
!
!
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line con 0
no login
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line telnet 0 4
login local-userlist
no shutdown
line ssh 0 4
login local-userlist
no shutdown
!
!
!
!
!
end
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