I have a TA908e that I have had running for about a year now. All of a sudden SIP calls in both directions are failing.
When I run the show voice call summary active I see the calls hit the ADTRAN. When calling from the outside in I hear one ring and then the line drops. Any Ideas on what would cause this? A reboot did not fix it and it has happened two days in a row now. Nothing has changed in the config on this router since it was installed.
Outbound Call
Active Calls Summary
CallID From To Duration Codec SHF Call State
==============================================================================
4 307xxxyyyy 1307xxxyyyy 00:00:07 N/A None Delivering
Voice Mail LoginT02
0/0 0/0
None
Inbound Call
CallID From To Duration Codec SHF Call State
==============================================================================
11 3078404640 307xxxyyyy 00:00:05 N/A None Delivering
WYOMING CALL T02
0/0 0/0
None
12 3072226333 307xxxyyyy 00:00:03 N/A None Delivering
Unavailable T02
0/0 0/0
None
Trunk Registration
GAUCSBC01#sh sip trunk-registration
Trk Identity Reg'd Grant Expires Success Failed Requests Chal Roll
--- -------------------- ----- ------- ------- ------- ------ -------- ---- ----
T01 307xxxyyy1 Yes 3600 2058 1 1 2 1 1
T02 307xxxyyy0 Yes 3600 2058 1 1 2 1 1
Total Displayed: 2
Current Config
Building configuration...
!
!
! ADTRAN, Inc. OS version R12.3.4.E
! Boot ROM version R10.9.3.B3
! Platform: Total Access 908e (3rd Gen), part number 4243908F1
! Serial number CFG1589461
!
!
hostname "GAUCSBC01"
enable password encrypted
!
!
clock timezone -7-Mountain-Time
!
ip subnet-zero
ip classless
ip default-gateway 204.98.x.y
ip routing
ipv6 unicast-routing
!
!
name-server x.x.x.x
!
!
auto-config
auto-config authname adtran encrypted password
!
event-history on
no logging forwarding
no logging email
!
service password-encryption
!
username "admin" password encrypted
!
!
ip firewall
ip firewall stealth
no ip firewall alg msn
no ip firewall alg mszone
no ip firewall alg h323
!
no dot11ap access-point-control
!
qos map ONEFLEX 10
match precedence 5
priority percent 90
qos map ONEFLEX 20
match precedence 3
bandwidth percent 5
!
!
!
!
interface eth 0/1
description GAUC_VOICE_LAN
ip address 172.20.x.x 255.255.255.0
no shutdown
media-gateway ip primary
!
!
interface eth 0/2
ip address dhcp hostname "TA908e"
shutdown
!
!
!
interface gigabit-eth 0/1
description GAUC_PUBLIC_VOICE_INT
ip address 204.98.x.x 255.255.255.248
ip access-policy ONEFLEX
traffic-shape rate 10000000
max-reserved-bandwidth 100
qos-policy out ONEFLEX
no shutdown
media-gateway ip primary
!
!
!
!
interface t1 0/1
shutdown
!
interface t1 0/2
shutdown
!
interface t1 0/3
shutdown
!
interface t1 0/4
shutdown
!
interface fxs 0/1
shutdown
!
interface fxs 0/2
shutdown
!
interface fxs 0/3
shutdown
!
interface fxs 0/4
shutdown
!
interface fxs 0/5
shutdown
!
interface fxs 0/6
shutdown
!
interface fxs 0/7
shutdown
!
interface fxs 0/8
shutdown
!
!
!
!
!
!
!
!
ip access-list standard GAUCADMIN
permit 172.20.x.x 0.0.0.255
permit 172.20.x.x 0 0.0.0.255
permit 172.28.x.x 0.0.255.255
permit 173.242.x.x 0.0.0.31
permit 204.98.x.x 0.0.0.7
!
ip access-list standard SIP_CTL
permit host x.x.x.x
permit host y.y.y.y
!
!
ip access-list extended VOICE-SUBNET
permit ip 172.20.x.x 0.0.0.255 any
!
ip access-list extended VoIP-Signaling
permit udp host 204.98.x.x any eq 5100
!
!
!
!
ip policy-class ONEFLEX
allow list VoIP-Signaling self
allow list GAUCADMIN self
allow list SIP_CTL self
!
ip policy-class VOICE-LAN
allow list VOICE-SUBNET self
allow list LOCAL
!
!
!
ip route 0.0.0.0 0.0.0.0 204.98.x.x
ip route 172.20.x.x 255.255.255.0 172.20.x.x
ip route 172.28.x.x 255.255.0.0 172.20.x.x
!
no tftp server
no tftp server overwrite
no http server
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
no sip tls
!
!
!
voice feature-mode network
voice transfer-mode local
voice forward-mode network
!
!
!
!
!
!
!
!
!
!
!
!
voice codec-list G711-G729
codec g711ulaw
codec g729
!
voice codec-list G729-G711
codec g729
codec g711ulaw
!
voice codec-list G711only
codec g711ulaw
!
!
voice trunk-list PBX_TRUNKS
trunk T10
!
voice trunk-list PROVIDER_TRUNKS
trunk T01
trunk T02
!
!
voice trunk T01 type sip
match dnis "*221" substitute "*21*"
match dnis "*661" substitute "*61*"
match dnis "*663" substitute "*63*"
match dnis "*667" substitute "*67*"
match dnis "*222" substitute "#21"
match dnis "*440" substitute "#40"
match dnis "*441" substitute "#41"
match dnis "*776" substitute "#76"
match dnis "*777" substitute "#77"
match dnis "*443" substitute "#43"
sip-server primary 65.149.23.24 udp 5100
domain "voip.centurylink.com"
hmr Trunk1_Outbound out
register 3073147251 auth-name "284200-307xxxyyyy" password encrypted asfsafsdasadfsadf
trust-domain
codec-list G711-G729 both
grammar from host domain
grammar p-asserted-identity host domain
grammar to host domain
authentication username "284200-307xxxyyyy" password encrypted afdsadfsafsafsdf
!
voice trunk T02 type sip
match dnis "*221" substitute "*21*"
match dnis "*661" substitute "*61*"
match dnis "*663" substitute "*63*"
match dnis "*667" substitute "*67*"
match dnis "*222" substitute "#21"
match dnis "*440" substitute "#40"
match dnis "*441" substitute "#41"
match dnis "*776" substitute "#76"
match dnis "*777" substitute "#77"
match dnis "*443" substitute "#43"
sip-server primary 65.149.25.24 udp 5100
domain "voip.centurylink.com"
hmr Trunk1_Outbound out
register 3073147250 auth-name "284200-307xxxyyyy" password encrypted asdasdfsadfsdafsdff
trust-domain
codec-list G711-G729 both
grammar from host domain
grammar p-asserted-identity host domain
grammar to host domain
authentication username "284200-307xxxyyyy" password encrypted asfdsafdsadfsadfsafsafdsadf
!
voice trunk T10 type sip
sip-server primary 172.20.x.x
trust-domain
codec-list G711-G729 both
grammar from host local
transfer-mode network
!
!
voice grouped-trunk IQ_SIP
trunk T10
accept $ cost 0
permit list PROVIDER_TRUNKS
permit list any
!deny all other trunks
!deny all other ani
!
!
voice grouped-trunk CUSTOMER_SIP
resource-selection circular
trunk T01
trunk T02
accept NXX-NXX-XXXX cost 0
accept 1-NXX-NXX-XXXX cost 0
accept 011-$ cost 0
accept 411 cost 0
accept 611 cost 0
accept 911 cost 0
accept *$ cost 0
accept $ cost 0
!
!
sip privacy
!
sip qos dscp 24
!
ip rtp symmetric-filter
ip rtp media-anchoring
ip rtp media-anchoring qos dscp 40
!
hmr policy Trunk1_Outbound
rule-set Add_PAI_Trunk1 10
!
hmr policy Trunk2_Outbound
rule-set Add_PAI_Trunk2 10
!
!
hmr rule-set Add_PAI_Trunk1
message-rule Initialize_Variables message-type request 10
set private-variable Add_PAI new-value false 10
message-rule Remove_Existing_PAI message-type any 15
remove header p-asserted-identity position all 10
message-rule Check_For_PAI_Candidates message-type request 20
set private-variable Add_PAI header sip-req-uri position first match-value "/^INVITE /" new-value true 10
message-rule Add_PAI_If_Needed message-type request 30
match private-variable Add_PAI match-value true
add header p-asserted-identity position first new-value /<sip:%public.Privacy_Prefix_1%@voip.centurylink.com>/ 10
!
hmr rule-set Add_PAI_Trunk2
message-rule Initialize_Variables message-type request 10
set private-variable Add_PAI new-value false 10
message-rule Remove_Existing_PAI message-type any 15
remove header p-asserted-identity position all 10
message-rule Check_For_PAI_Candidates message-type request 20
set private-variable Add_PAI header sip-req-uri position first match-value "/^INVITE /" new-value true 10
message-rule Add_PAI_If_Needed message-type request 30
match private-variable Add_PAI match-value true
add header p-asserted-identity position first new-value /<sip:%public.Privacy_Prefix_2%@voip.centurylink.com>/ 10
!
!
hmr set public-variable Privacy_Prefix_1 new-value "307xxxyyyy"
hmr set public-variable Privacy_Prefix_2 new-value "307xxxyyyy"
!
!
!
end
something major must have changed for calls to drop in both directions. Have you verified that your SIP service hasn't been turned off?
verify trunk registration
show sip trunk-registration
SSh or telnet into your unit and enter the following debug and then place an outbound call, put that debug in one text file, then place an inbound call and put that debug into another text file and then attach them to your reply.
debug sip stack messages
debug voice verbose
debug sip cldu
debug hmr
Thanks,
Mark