17 Replies Latest reply on May 31, 2019 12:40 PM by yredovich

    Busy signal when dialing out - New set up

    nicklarose New Member

      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

        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#

      ADTRAN908E#

      ADTRAN908E#

      ADTRAN908E#

      ADTRAN908E#

      ADTRAN908E#

      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

        • Re: Busy signal when dialing out - New set up
          nicklarose New Member

          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.

            • Re: Busy signal when dialing out - New set up
              markfreeman Employee

              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

                • Re: Busy signal when dialing out - New set up
                  nicklarose New Member

                  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

              • Re: Busy signal when dialing out - New set up
                acoolov New Member

                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.

                • Re: Busy signal when dialing out - New set up
                  yredovich New Member

                  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.

                    • Re: Busy signal when dialing out - New set up
                      acoolov New Member

                      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"

                        • Re: Busy signal when dialing out - New set up
                          jwable Frequent Visitor

                          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

                      • Re: Busy signal when dialing out - New set up
                        acoolov New Member

                        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.