TA908e SBC, SIP calls failing

[philhutchison]

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

[Anonymous]

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