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nicklarose · ‎08-23-2018

Hello,

I would like to to use an Adtran 908E as the gateway device so that I can peel off a few analog lines as well. This is my first experience with Adtran and its going pretty well, but I am unable to get outbound calls to work as I immediately get a busy signal after the last digit is input. Calling inbound works fine.

Here is my config, as well as debug output for "debug sip stack messages", and "debug voice switchboard" while I dialed outbound to destination: 3143212222

My DID is 3145001048

A.B.C.D is what I used to replace the public IP on my ADTRAN

______________________________________________________________________________________________________________

ADTRAN908E#show run

Building configuration...

!

! ADTRAN, Inc. OS version R13.2.0.E

! Boot ROM version R10.9.3.B1

! Platform: Total Access 908e (3rd Gen)

!

hostname "ADTRAN908E"

enable password md5 encrypted omitted

!

clock timezone -6-Central-Time

!

ip subnet-zero

ip classless

ip routing

ipv6 unicast-routing

!

name-server 208.67.220.220

!

auto-config

auto-config authname cstltech encrypted password omitted

!

event-history on

no logging forwarding

no logging email

!

service password-encryption

!

username "cstltech" password encrypted "omitted"

!

ip firewall

no ip firewall alg msn

no ip firewall alg mszone

no ip firewall alg h323

!

no dot11ap access-point-control

!

interface eth 0/1

no shutdown

!

interface eth 0/2

no shutdown

!

interface gigabit-eth 0/1

ip address A.B.C.D 255.255.255.224

no ip proxy-arp

ip access-policy Public

no shutdown

media-gateway ip primary

!

interface t1 0/1

no shutdown

!

interface t1 0/2

no shutdown

!

interface t1 0/3

no shutdown

!

interface t1 0/4

no shutdown

!

interface fxs 0/1

no shutdown

!

interface fxs 0/2

no shutdown

!

interface fxs 0/3

no shutdown

!

interface fxs 0/4

no shutdown

!

interface fxs 0/5

no shutdown

!

interface fxs 0/6

no shutdown

!

interface fxs 0/7

no shutdown

!

interface fxs 0/8

no shutdown

!

ip access-list standard ics

remark Internet Connection Sharing

permit any

!

ip access-list standard self

remark Traffic to Adtran

permit any

!

ip policy-class Private

allow list self self

nat source list ics interface gigabit-ethernet 0/1 overload

!

ip policy-class Public

allow list self self

!

ip route 0.0.0.0 0.0.0.0 D.F.G.W

!

no tftp server

no tftp server overwrite

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 forward-mode network

!

voice dial-plan 1 local NXX-NXX-XXXX

voice dial-plan 2 long-distance 1-NXX-NXX-XXXX

voice dial-plan 3 operator-assisted 1-411

voice dial-plan 4 user1 N11

voice dial-plan 5 user2 XXX-NXX-XXXX

!

voice codec-list 711

default

codec g711ulaw

!

voice trunk T01 type sip

description "To FlowRoute"

sip-server primary sip.flowroute.com

registrar primary sip.flowroute.com

register SipActUserName auth-name "SipActUserName" password encrypted "omitted"

codec-list 711 both

!

voice grouped-trunk FLOWROUTE

trunk T01

accept 2XXX cost 0

accept 13145001048 cost 1

accept $ cost 0

!

voice user 333

connect fxs 0/3

no cos

password encrypted "omitted"

caller-id-override external-name User333

did "13145001048"

sip-authentication password encrypted "omitted"

modem-passthrough

codec-list 711

!

line con 0

login

password encrypted "omitted"

line-timeout 0

!

line telnet 0 4

login

password encrypted "omitted"

no shutdown

line ssh 0 4

login local-userlist

no shutdown

!

ntp source ethernet 0/1

!

end

ADTRAN908E#

______________________________________________________________________________________________________________

ADTRAN908E#debug voice switchboard

ADTRAN908E#

04:55:02.946 SB.CALL 4 Idle Called the call routine with 3143212222

04:55:02 SB.TGMgr For dialed number 3143212222, against template $, on TrunkGroup FLOWROUTE, the score is 500

04:55:02.946 SB.CCM isMappable:

04:55:02.947 SB.CCM : Call Struct 0x0x502b5410 : Call-ID = 4

04:55:02.947 SB.CCM : Org Acct = 333 Dst Acct = T01

04:55:02.947 SB.CCM : Org Port ID = FxsPhone 0/3 Dst Port ID = unknown 0/0

04:55:02.947 SB.CCM isMappable: Call Connection Type is TDM_TO_RTP

04:55:02.947 SB.CCM isMappable: Reserving RTP Channel 0/1.1

04:55:02.948 SB.CCM isMappable: Creating SDP Offer

04:55:02.949 SB.CCM updateOfferWithEndpointConfig: DTMF(NTE 101), VAD(off), ptime(0)

04:55:02.949 SB.CCM translateOffer: offer codec list: PCMU

04:55:02.950 SB.CCM translateOffer: revised offer codec list: PCMU

04:55:02.950 SB.CCM translateOffer: codec list after answerer: PCMU

04:55:02.950 SB.CCM translateOffer: DTMF signaling: answerer has no restrictions configured, passing offer(NTE 101) through

04:55:02.951 SB.CCM translateOffer: success

04:55:02.952 SB.CALL 4 Idle Call sent from 333 to T01 (3143212222)

04:55:02.952 SB.CALL 4 State change >> Idle->Delivering

04:55:02.957 SB.CALL 4 Delivering Called the deliverResponse routine from Delivering

04:55:02.957 SB.CALL 4 Delivering DeliverResponse(accept) sent from T01 to 333

04:55:02 SB.CallStructObserver 4 Created

04:55:03.169 SB.CALL 4 Delivering Called the clearCall routine

04:55:03.169 SB.CALL 4 Delivering SIP Proxy rejected call to 3143212222 for survivability - no matching Proxy user

04:55:03.169 SB.CALL 4 Delivering No available resources on call from 333 to T01 (last attempt)

04:55:03.169 SB.CALL 4 State change >> Delivering->Clearing

04:55:03.170 SB.CALL 4 Clearing Called the clearResponse routine

04:55:03.170 SB.CALL 4 State change >> Clearing->CallIdlePending

04:55:03.170 SB.CCM release:

04:55:03.170 SB.CCM : Call Struct 0x0x502b5410 : Call-ID = 4

04:55:03.171 SB.CCM : Org Acct = 333 Dst Acct = T01

04:55:03.171 SB.CCM : Org Port ID = FxsPhone 0/3 Dst Port ID = SipTrunk 0/0.97

04:55:03.171 SB.CCM : SDP Transaction = CallID: 4

04:55:03.171 SB.CCM : SDP Offer = 0x502b0d10, (127.0.0.3:10008)

04:55:03.171 SB.CCM : Offer side SRTP session details

04:55:03.171 SB.CCM : None

04:55:03.172 SB.CCM : Answer side SRTP session details

04:55:03.172 SB.CCM : None

04:55:03.172 SB.CCM : RTP Channel = 0/1.1

04:55:03.172 SB.CCM release: Call Connection Type is TDM_TO_RTP

04:55:03.172 SB.CCM release: Releasing RTP Channel 0/1.1

04:55:03.172 SB.CALL 4 CallIdlePending ClearResponse sent from 333 to T01

04:55:03 SB.CallStructObserver 4 Finalized

ADTRAN908E#

ADTRAN908E#no debug voice switchboard

ADTRAN908E#

ADTRAN908E#debug sip stack messages

ADTRAN908E#

04:57:29.472 SIP.STACK MSG Tx: UDP src=A.B.C.D:5060 dst=216.115.69.144:5060

04:57:29.472 SIP.STACK MSG INVITE sip:3143212222@sip.flowroute.com:5060 SIP/2.0

04:57:29.473 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bb68-7f000001-13c4-d37-4bb34a1f-d37

04:57:29.473 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>

04:57:29.473 SIP.STACK MSG Call-ID: 4fad4b28-7f000001-13c4-d37-6ab56556-d37@sip.flowroute.com

04:57:29.473 SIP.STACK MSG CSeq: 1 INVITE

04:57:29.473 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d37-33a1e0-101898a

04:57:29.473 SIP.STACK MSG Max-Forwards: 70

04:57:29.474 SIP.STACK MSG Supported: 100rel,replaces

04:57:29.474 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

04:57:29.474 SIP.STACK MSG User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R13.2.0.E

04:57:29.474 SIP.STACK MSG Contact: <sip:13145001048@A.B.C.D:5060;transport=UDP>

04:57:29.474 SIP.STACK MSG Content-Type: application/sdp

04:57:29.474 SIP.STACK MSG Content-Length: 210

04:57:29.474 SIP.STACK MSG

04:57:29.475 SIP.STACK MSG v=0

04:57:29.475 SIP.STACK MSG o=- 1535018249 1 IN IP4 A.B.C.D

04:57:29.475 SIP.STACK MSG s=-

04:57:29.475 SIP.STACK MSG c=IN IP4 A.B.C.D

04:57:29.475 SIP.STACK MSG t=0 0

04:57:29.475 SIP.STACK MSG m=audio 10010 RTP/AVP 0 101

04:57:29.476 SIP.STACK MSG a=silenceSupp:off - - - -

04:57:29.476 SIP.STACK MSG a=rtpmap:0 PCMU/8000

04:57:29.476 SIP.STACK MSG a=rtpmap:101 telephone-event/8000

04:57:29.476 SIP.STACK MSG a=fmtp:101 0-15

04:57:29.476 SIP.STACK MSG

04:57:29.537 SIP.STACK MSG Rx: UDP src=216.115.69.144:5060 dst=A.B.C.D:5060

04:57:29.537 SIP.STACK MSG SIP/2.0 100 Trying

04:57:29.537 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bb68-7f000001-13c4-d37-4bb34a1f-d37

04:57:29.537 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>

04:57:29.538 SIP.STACK MSG Call-ID: 4fad4b28-7f000001-13c4-d37-6ab56556-d37@sip.flowroute.com

04:57:29.538 SIP.STACK MSG CSeq: 1 INVITE

04:57:29.538 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d37-33a1e0-101898a

04:57:29.538 SIP.STACK MSG Content-Length: 0

04:57:29.538 SIP.STACK MSG

04:57:29.580 SIP.STACK MSG Rx: UDP src=216.115.69.144:5060 dst=A.B.C.D:5060

04:57:29.580 SIP.STACK MSG SIP/2.0 407 Proxy Authentication Required

04:57:29.580 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bb68-7f000001-13c4-d37-4bb34a1f-d37

04:57:29.580 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>;tag=870a5b262384e4f9f82f59836d699db5.970d

04:57:29.581 SIP.STACK MSG Call-ID: 4fad4b28-7f000001-13c4-d37-6ab56556-d37@sip.flowroute.com

04:57:29.581 SIP.STACK MSG CSeq: 1 INVITE

04:57:29.581 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d37-33a1e0-101898a

04:57:29.581 SIP.STACK MSG Proxy-Authenticate: Digest realm="sip.flowroute.com", nonce="W38vaFt/LjwXLaE40Kk9/41EAHk/i7gV", qop="auth"

04:57:29.581 SIP.STACK MSG Content-Length: 0

04:57:29.581 SIP.STACK MSG

04:57:29.583 SIP.STACK MSG Tx: UDP src=A.B.C.D:5060 dst=216.115.69.144:5060

04:57:29.583 SIP.STACK MSG ACK sip:3143212222@sip.flowroute.com:5060;transport=UDP SIP/2.0

04:57:29.583 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bb68-7f000001-13c4-d37-4bb34a1f-d37

04:57:29.584 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>;tag=870a5b262384e4f9f82f59836d699db5.970d

04:57:29.584 SIP.STACK MSG Call-ID: 4fad4b28-7f000001-13c4-d37-6ab56556-d37@sip.flowroute.com

04:57:29.584 SIP.STACK MSG CSeq: 1 ACK

04:57:29.584 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d37-33a1e0-101898a

04:57:29.584 SIP.STACK MSG Max-Forwards: 70

04:57:29.584 SIP.STACK MSG Supported: 100rel,replaces

04:57:29.585 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

04:57:29.585 SIP.STACK MSG User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R13.2.0.E

04:57:29.585 SIP.STACK MSG Contact: <sip:13145001048@A.B.C.D:5060;transport=UDP>

04:57:29.585 SIP.STACK MSG Content-Length: 0

04:57:29.585 SIP.STACK MSG

04:57:29.588 SIP.STACK MSG Tx: UDP src=A.B.C.D:5060 dst=216.115.69.144:5060

04:57:29.588 SIP.STACK MSG INVITE sip:3143212222@sip.flowroute.com:5060 SIP/2.0

04:57:29.589 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bb68-7f000001-13c4-d37-4bb34a1f-d37

04:57:29.589 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>

04:57:29.589 SIP.STACK MSG Call-ID: 4fad4b28-7f000001-13c4-d37-6ab56556-d37@sip.flowroute.com

04:57:29.589 SIP.STACK MSG CSeq: 2 INVITE

04:57:29.589 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d37-33a254-133a88db

04:57:29.589 SIP.STACK MSG Max-Forwards: 70

04:57:29.590 SIP.STACK MSG Supported: 100rel,replaces

04:57:29.590 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

04:57:29.590 SIP.STACK MSG User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R13.2.0.E

04:57:29.590 SIP.STACK MSG Contact: <sip:13145001048@A.B.C.D:5060;transport=UDP>

04:57:29.590 SIP.STACK MSG Proxy-Authorization: Digest username="",realm="sip.flowroute.com",nonce="W38vaFt/LjwXLaE40Kk9/41EAHk/i7gV",uri="sip:3143212222@sip.flowroute.com:5060",response="75444c746a3a6c3139a3f02325c2c883",algorithm=MD5,cnonce="33a254",qop=auth,nc=00000001

04:57:29.590 SIP.STACK MSG Content-Type: application/sdp

04:57:29.590 SIP.STACK MSG Content-Length: 210

04:57:29.591 SIP.STACK MSG

04:57:29.591 SIP.STACK MSG v=0

04:57:29.591 SIP.STACK MSG o=- 1535018249 1 IN IP4 A.B.C.D

04:57:29.591 SIP.STACK MSG s=-

04:57:29.591 SIP.STACK MSG c=IN IP4 A.B.C.D

04:57:29.591 SIP.STACK MSG t=0 0

04:57:29.591 SIP.STACK MSG m=audio 10010 RTP/AVP 0 101

04:57:29.592 SIP.STACK MSG a=silenceSupp:off - - - -

04:57:29.592 SIP.STACK MSG a=rtpmap:0 PCMU/8000

04:57:29.592 SIP.STACK MSG a=rtpmap:101 telephone-event/8000

04:57:29.592 SIP.STACK MSG a=fmtp:101 0-15

04:57:29.592 SIP.STACK MSG

04:57:29.653 SIP.STACK MSG Rx: UDP src=216.115.69.144:5060 dst=A.B.C.D:5060

04:57:29.653 SIP.STACK MSG SIP/2.0 100 Trying

04:57:29.653 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bb68-7f000001-13c4-d37-4bb34a1f-d37

04:57:29.654 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>

04:57:29.654 SIP.STACK MSG Call-ID: 4fad4b28-7f000001-13c4-d37-6ab56556-d37@sip.flowroute.com

04:57:29.654 SIP.STACK MSG CSeq: 2 INVITE

04:57:29.654 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d37-33a254-133a88db

04:57:29.654 SIP.STACK MSG Content-Length: 0

04:57:29.654 SIP.STACK MSG

04:57:29.696 SIP.STACK MSG Rx: UDP src=216.115.69.144:5060 dst=A.B.C.D:5060

04:57:29.697 SIP.STACK MSG SIP/2.0 403 Bad au - support@flowroute.com

04:57:29.697 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bb68-7f000001-13c4-d37-4bb34a1f-d37

04:57:29.697 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>;tag=870a5b262384e4f9f82f59836d699db5.4a2c

04:57:29.697 SIP.STACK MSG Call-ID: 4fad4b28-7f000001-13c4-d37-6ab56556-d37@sip.flowroute.com

04:57:29.697 SIP.STACK MSG CSeq: 2 INVITE

04:57:29.697 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d37-33a254-133a88db

04:57:29.698 SIP.STACK MSG Content-Length: 0

04:57:29.698 SIP.STACK MSG

04:57:29.700 SIP.STACK MSG Tx: UDP src=A.B.C.D:5060 dst=216.115.69.144:5060

04:57:29.700 SIP.STACK MSG ACK sip:3143212222@sip.flowroute.com:5060;transport=UDP SIP/2.0

04:57:29.700 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bb68-7f000001-13c4-d37-4bb34a1f-d37

04:57:29.700 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>;tag=870a5b262384e4f9f82f59836d699db5.4a2c

04:57:29.701 SIP.STACK MSG Call-ID: 4fad4b28-7f000001-13c4-d37-6ab56556-d37@sip.flowroute.com

04:57:29.701 SIP.STACK MSG CSeq: 2 ACK

04:57:29.701 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d37-33a254-133a88db

04:57:29.701 SIP.STACK MSG Max-Forwards: 70

04:57:29.701 SIP.STACK MSG Supported: 100rel,replaces

04:57:29.701 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

04:57:29.701 SIP.STACK MSG User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R13.2.0.E

04:57:29.702 SIP.STACK MSG Contact: <sip:13145001048@A.B.C.D:5060;transport=UDP>

04:57:29.702 SIP.STACK MSG Content-Length: 0

04:57:29.702 SIP.STACK MSG

04:57:29.933 SIP.STACK MSG Tx: UDP src=A.B.C.D:5060 dst=216.115.69.144:5060

04:57:29.933 SIP.STACK MSG REGISTER sip:sip.flowroute.com:5060 SIP/2.0

04:57:29.934 SIP.STACK MSG From: <sip:SipAcct#@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bf08-7f000001-13c4-d38-47be38cb-d38

04:57:29.934 SIP.STACK MSG To: <sip:SipAcct#@sip.flowroute.com:5060;transport=UDP>

04:57:29.934 SIP.STACK MSG Call-ID: 4fb18010-7f000001-13c4-71b-49fdb6f0-71b

04:57:29.934 SIP.STACK MSG CSeq: 601 REGISTER

04:57:29.934 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d38-33a3ad-18a1c7e6

04:57:29.934 SIP.STACK MSG Max-Forwards: 70

04:57:29.934 SIP.STACK MSG Supported: 100rel,replaces

04:57:29.935 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

04:57:29.935 SIP.STACK MSG User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R13.2.0.E

04:57:29.935 SIP.STACK MSG Contact: <sip:SipAcct#@A.B.C.D:5060;transport=UDP>

04:57:29.935 SIP.STACK MSG Expires: 3600

04:57:29.935 SIP.STACK MSG Content-Length: 0

04:57:29.935 SIP.STACK MSG

04:57:30.038 SIP.STACK MSG Rx: UDP src=216.115.69.144:5060 dst=A.B.C.D:5060

04:57:30.039 SIP.STACK MSG SIP/2.0 401 Unauthorized

04:57:30.039 SIP.STACK MSG From: <sip:SipAcct#@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bf08-7f000001-13c4-d38-47be38cb-d38

04:57:30.039 SIP.STACK MSG To: <sip:SipAcct#@sip.flowroute.com:5060;transport=UDP>;tag=870a5b262384e4f9f82f59836d699db5.0a38

04:57:30.039 SIP.STACK MSG Call-ID: 4fb18010-7f000001-13c4-71b-49fdb6f0-71b

04:57:30.039 SIP.STACK MSG CSeq: 601 REGISTER

04:57:30.039 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d38-33a3ad-18a1c7e6

04:57:30.039 SIP.STACK MSG WWW-Authenticate: Digest realm="sip.flowroute.com", nonce="W38vaFt/LjzjUM4DliOlqc/3dzSKh5yx", qop="auth"

04:57:30.040 SIP.STACK MSG Content-Length: 0

04:57:30.040 SIP.STACK MSG

04:57:30.042 SIP.STACK MSG Tx: UDP src=A.B.C.D:5060 dst=216.115.69.144:5060

04:57:30.043 SIP.STACK MSG REGISTER sip:sip.flowroute.com:5060 SIP/2.0

04:57:30.043 SIP.STACK MSG From: <sip:SipAcct#@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bf08-7f000001-13c4-d38-47be38cb-d38

04:57:30.043 SIP.STACK MSG To: <sip:SipAcct#@sip.flowroute.com:5060;transport=UDP>

04:57:30.043 SIP.STACK MSG Call-ID: 4fb18010-7f000001-13c4-71b-49fdb6f0-71b

04:57:30.043 SIP.STACK MSG CSeq: 602 REGISTER

04:57:30.043 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d38-33a41a-683e2fe7

04:57:30.044 SIP.STACK MSG Max-Forwards: 70

04:57:30.044 SIP.STACK MSG Supported: 100rel,replaces

04:57:30.044 SIP.STACK MSG Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER, REGISTER

04:57:30.044 SIP.STACK MSG User-Agent: ADTRAN_Total_Access_908e_3rd_Gen/R13.2.0.E

04:57:30.044 SIP.STACK MSG Contact: <sip:SipAcct#@A.B.C.D:5060;transport=UDP>

04:57:30.044 SIP.STACK MSG Expires: 3600

04:57:30.045 SIP.STACK MSG Authorization: Digest username="SipAcct#",realm="sip.flowroute.com",nonce="W38vaFt/LjzjUM4DliOlqc/3dzSKh5yx",uri="sip:sip.flowroute.com:5060",response="779bc0b96efa1a6c5d657ed6c7014cd4",algorithm=MD5,cnonce="33a41a",qop=auth,nc=00000001

04:57:30.045 SIP.STACK MSG Content-Length: 0

04:57:30.045 SIP.STACK MSG

04:57:30.148 SIP.STACK MSG Rx: UDP src=216.115.69.144:5060 dst=A.B.C.D:5060

04:57:30.149 SIP.STACK MSG SIP/2.0 200 OK

04:57:30.149 SIP.STACK MSG From: <sip:SipAcct#@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bf08-7f000001-13c4-d38-47be38cb-d38

04:57:30.149 SIP.STACK MSG To: <sip:SipAcct#@sip.flowroute.com:5060;transport=UDP>;tag=870a5b262384e4f9f82f59836d699db5.d3b5

04:57:30.149 SIP.STACK MSG Call-ID: 4fb18010-7f000001-13c4-71b-49fdb6f0-71b

04:57:30.149 SIP.STACK MSG CSeq: 602 REGISTER

04:57:30.149 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d38-33a41a-683e2fe7

04:57:30.150 SIP.STACK MSG Contact: <sip:SipAcct#@199.193.198.107:5060;rinstance=225ec16d5e73883a>;q=1;expires=15;received="sip:199.193.198.107:5060", <sip:SipAcct#@199.193.199.228:5060;rinstance=516c7cf985659a40>;q=1;expires=42;received="sip:199.193.199.228:5060", <sip:SipAcct#@199.193.198.116:5060;rinstance=5ab2f358a3e73217>;q=1;expires=32;received="sip:199.193.198.116:5060", <sip:SipAcct#@A.B.C.D:5060;transport=UDP>;q=1;expires=2725;received="sip:A.B.C.D:5060"

04:57:30.150 SIP.STACK MSG Content-Length: 0

04:57:30.150 SIP.STACK MSG

Thanks in advance for your assistance!

Nick

nicklarose · ‎08-30-2018

Any thoughts?

I assume the following line is a clue, but haven't been able to get it working:

04:55:03.169 SB.CALL 4 Delivering

SIP Proxy rejected call to 3143212222 for survivability - no matching Proxy user

Quick recap:

Adtran 908E. I have the trunk set up and registered and I am able to call inbound to the DID from my cell phone to this analog phone. 2 way audio

I am unable to call outbound from the analog phone.

Anonymous · ‎08-30-2018

Nick,

Sorry for late reply.

Looks like your SIP authenication is bad:

we send invite with auth but then the sip server replies back with 403 error message:

04:57:29.696 SIP.STACK MSG Rx: UDP src=216.115.69.144:5060 dst=A.B.C.D:5060

04:57:29.697 SIP.STACK MSG SIP/2.0 403 Bad au - support@flowroute.com

04:57:29.697 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa4bb68-7f000001-13c4-d37-4bb34a1f-d37

04:57:29.697 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>;tag=870a5b262384e4f9f82f59836d699db5.4a2c

04:57:29.697 SIP.STACK MSG Call-ID: 4fad4b28-7f000001-13c4-d37-6ab56556-d37@sip.flowroute.com

04:57:29.697 SIP.STACK MSG CSeq: 2 INVITE

04:57:29.697 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-d37-33a254-133a88db

04:57:29.698 SIP.STACK MSG Content-Length: 0

Do the following:

show sip trunk-registration

like this:

TA900e#show sip trunk-registration

Trk Identity Reg'd Grant Expires Success Failed Requests Chal Roll

--- -------------------- ----- ------- ------- ------- ------ -------- ---- ----

T01 4045162100 Yes 3600 2365 439 0 878 439 0

T01 4045162001 Yes 3600 2365 439 0 878 439 0

T01 4045162002 Yes 3600 2365 439 0 878 439 0

T01 4045162200 Yes 3600 2365 439 0 878 439 0

You want to make sure it says yes after the registration number

if not then you need to verify your sip username and password in Voice trunk t01 with what your provider has.

Let me know what you find out.

-Mark

nicklarose · ‎08-31-2018

Does this mean the auth on the user (voice user 333) is wrong? What is the user supposed to be authenticating against? I believe when I was setting it up (following a guide) I was confused about what to use for the password of the "voice user 333" line so that definitely could be the issue.

I went ahead and removed the following two bolded lines to test, but no change in behavior:

voice user 333

connect fxs 0/3

no cos

password encrypted "omitted"

caller-id-override external-name User333

did "13145001048"

sip-authentication password encrypted "omitted"

modem-passthrough

codec-list 711

Here is the output requested

______________________________________________________________________________________________

ADTRAN908E#show sip trunk-registration

Trk Identity Reg'd Grant Expires Success Failed Requests Chal Roll

--- -------------------- ----- ------- ------- ------- ------ -------- ---- ----

T01 omitted Yes 50 4 873 6373 1747 873 6559

Total Displayed: 1

______________________________________________________________________________________________

Thanks!

Nick

Anonymous · ‎08-31-2018

copy and paste this into your config:

voice trunk T01

no register SipActUserName auth-name "SipActUserName" password encrypted "omitted"

register SipActUserName

authentication username "SipActUserName" password encrypted "omitted"

end

then make sure your buffer is big on your telnet/ssh client

turn on the following debug

debug sip stack messages

debug sip cldu

debug voice verbose

debug interface fxs

then enter

clear sip trunk-registration

sip trunk-registration force-register

then

place an inbound call

then place an outbound call.

then copy all debug and put in a text document and attach it to your reply.

-Mark

Anonymous · ‎08-31-2018

make sure you change and update the info in the commands to have your actual data!

nicklarose · ‎08-31-2018

Mark,

Done and attached. Lines 114-116 is the clear / force registration.

I called from 314-874-6851 into the analog line @ 314-500-1048 - success

then call from the analog line out to 314-321-2222 - busy tone

Thank you!

Nick

Anonymous · ‎08-31-2018

Nick,

Looks like you have been added to a blacklist by your provider. you need to contact their support as the SIP message indicates at line 1206:

01:25:32.809 SIP.STACK MSG Tx: UDP src=A.B.C.D:5060 dst=216.115.69.144:5060

01:25:32.810 SIP.STACK MSG INVITE sip:3143212222@sip.flowroute.com:5060 SIP/2.0

01:25:32.874 SIP.STACK MSG Rx: UDP src=216.115.69.144:5060 dst=A.B.C.D:5060

01:25:32.874 SIP.STACK MSG SIP/2.0 100 Trying

01:25:32.917 SIP.STACK MSG Rx: UDP src=216.115.69.144:5060 dst=A.B.C.D:5060

01:25:32.917 SIP.STACK MSG SIP/2.0 403 Destination Blacklist - support@flowroute.com

01:25:32.917 SIP.STACK MSG From: "User333" <sip:13145001048@sip.flowroute.com:5060;transport=UDP>;tag=4fa26c88-7f000001-13c4-a678a-6d049a35-a678a

01:25:32.917 SIP.STACK MSG To: <sip:3143212222@sip.flowroute.com:5060>;tag=870a5b262384e4f9f82f59836d699db5.032a

01:25:32.917 SIP.STACK MSG Call-ID: 4fad79e0-7f000001-13c4-a678a-69dba8a5-a678a@sip.flowroute.com

01:25:32.918 SIP.STACK MSG CSeq: 2 INVITE

01:25:32.918 SIP.STACK MSG Via: SIP/2.0/UDP A.B.C.D:5060;branch=z9hG4bK-a678b-28a47767-7bec0ddd

01:25:32.918 SIP.STACK MSG Content-Length: 0

Hope that helps.

-Mark

nicklarose · ‎09-06-2018

Mark,

I opened a ticket with FlowRoute and it's resolved. Turns out I just needed to prepend the area code when dialing and all outbound calls are working! Thanks so much for taking the time to troubleshoot. Much appreciated!

Nick

Anonymous · ‎09-06-2018

Nick,

Thanks for update. that is weird because the outbound call you made had the area code in it (314) , i think they must have made some other changes.

04:57:29.588 SIP.STACK MSG Tx: UDP src=A.B.C.D:5060 dst=216.115.69.144:5060

04:57:29.588 SIP.STACK MSG INVITE sip:3143212222@sip.flowroute.com:5060 SIP/2.0

Glad everything is working now.

-Mark

Anonymous · ‎05-22-2019

Hello,

I am having the same exact issue with TA924e connected to a flowroute trunk as well.

Except, mine works fine, but as soon as I configure ANI Substitution in the trunk with the CID i need all my FXS ports to show as, I get this error:

SIP/2.0 403 Bad au - support@flowroute.com

What i'm trying to do is override a caller ID for all FSX stations so they all show the same number. I tried all other options but nothing works. It always just shows the sip identity number as the caller id.

I need all stations to show the same phone number.

I learned that I can use a ANI Substitution to replace the caller id with the desired number in the trunk, so that any extension using that trunk to call out would show as that phone number.

But as soon as I configure it like this for example:

match $ substitute 1234567891 (i configure it in the gui)

Outbound calling stops working, I get the busy signal and that error - SIP/2.0 403 Bad au - support@flowroute.com

Any idea why it's doing it?

ANI Substitution is supposed to replace the caller ID, not the dialed number.

On the other hand, the DNIS substitution replaces the dialed number (which i don't need at this point)

Please help.

Anonymous · ‎05-23-2019

@acoolov

Instead of using ANI substitution, try putting an override command on your voice trunk

voice trunk T01 type sip

caller-id-override number-inbound 1002003000

Even though it says "inbound" the command will make the CPE send that DID on all outbound calls sent from that trunk

Anonymous · ‎05-24-2019

Thanks for your response,

I just tried that and it does not override the ext number 200 that I have put in the sip identity field. It always shows the sip identity that I put in there.

Are there any other ways to override outbound caller id?

Anonymous · ‎05-24-2019

acoolov

Yep, I see what you mean. I just tried it on my FXS setup and wasn't able to get it to work either right away. Guess I was thinking about the caller ID override on PRI when I wrote this earlier. Sorry about that.

I went back to the lab and got it to work in the end - I removed the caller ID override from voice trunk and added it to all of the FXS voice users instead. This puts the override DID in the SIP From and Contact fields for all outbound calls from any of the users. So try that on your side and it should work now.

voice trunk T01 type sip

no caller-id-override number-inbound

voice user 300

connect fxs 0/1

caller-id-override external-number 1002003000

voice user 301

connect fxs 0/2

caller-id-override external-number 1002003000

etc.

Anonymous · ‎05-29-2019

I just tried this on my TA924e

voice user 300

connect fxs 0/1

caller-id-override external-number 1002003000

But I am still getting the SIP Identity number. Here is my exact config. 1111111111 is the number I need all my lines to show when calling out. 2222222222 is the number that is entered in the sip identity for the extension.

When I call out, I still see 2222222222 showing, not 1111111111 that I want it to show.

voice user 200

connect fxs 0/1

password "1234"

caller-id-override external-number 1111111111

sip-identity 2222222222 T01 register

sip-authentication password "1234"

Anonymous · ‎05-30-2019

Normally you would have the SIP Trunk handle the registration not the FXS port. remove the sip-identity and sip-authentication from the FXS ports leave it on the sip trunk. Use caller-id-override to handle number change. If your provider does not allow multiple registrations from the same identity the additional registration will be blocked or discarded. I provided additional information in your original post.

John Wable

Anonymous · ‎05-30-2019

Sure, the problem is, when remove the sip identity on the fxs port it removes the trunk chosen to be used on that port. How do i choose the trunk to use for fxs port without adding sip identity? In the gui I didn't see another way to choose a trunk for the fxs port. And when i chose the trunk, i leave username and password blank so they don't authenticate, and only enter sip identity name field. But with Sipstation it things that field is username and fails. With flowroute it works.

Anonymous · ‎05-31-2019

You should have a "voice grouped-trunk" statement in your config that tells the CPE which trunk to route all outbound calls to.

For example:

voice grouped-trunk SIP

description "To SIP Network"

trunk T01

accept $ cost 0

!

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Re: Busy signal when dialing out - New set up

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Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

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Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

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Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

Re: Busy signal when dialing out - New set up

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Re: Busy signal when dialing out - New set up

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