both incoming and outgoing call doesn't work

Discussion in 'General' started by tsmvision, Mar 6, 2009.

  1. tsmvision

    Joined:
    Mar 6, 2009
    Messages:
    2
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    0
    Hello

    I'm newbie w/ Elastix.
    I've tried to let my box work for incoming and outgoing call, but unfortunately it didn't.

    first here is my debugging log for outgoing

    thank you

    <outgoing>

    PBXTest*CLI>
    <--- SIP read from 10.10.10.190:5060 --->
    INVITE sip:919283016340@10.10.10.132 SIP/2.0
    Via: SIP/2.0/UDP 10.10.10.190:5060;branch=z9hG4bK1daf0564d08c3f3e
    From: "namjoong01" <sip:1001@10.10.10.132>;tag=fa8246c96eb9ac45
    To: <sip:919283016340@10.10.10.132>
    Contact: <sip:1001@10.10.10.190:5060;transport=udp>
    Supported: replaces, timer, path
    Call-ID: c1c645793a4ce95c@10.10.10.190
    CSeq: 5199 INVITE
    User-Agent: Grandstream GXP1200 1.1.6.44
    Proxy-Require: 10.10.10.132
    Max-Forwards: 70
    Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    Content-Type: application/sdp
    Content-Length: 348

    v=0
    o=1001 8000 8000 IN IP4 10.10.10.190
    s=SIP Call
    c=IN IP4 10.10.10.190
    t=0 0
    m=audio 5014 RTP/AVP 0 8 4 18 2 97 9 3
    a=sendrecv
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:97 iLBC/8000
    a=fmtp:97 mode=20
    a=rtpmap:9 G722/8000
    a=rtpmap:3 GSM/8000
    a=ptime:20

    <------------->
    --- (14 headers 17 lines) ---
    Sending to 10.10.10.190 : 5060 (no NAT)
    Using INVITE request as basis request - c1c645793a4ce95c@10.10.10.190

    <--- Reliably Transmitting (NAT) to 10.10.10.190:5060 --->
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 10.10.10.190:5060;branch=z9hG4bK1daf0564d08c3f3e;received=10.10.10.190
    From: "namjoong01" <sip:1001@10.10.10.132>;tag=fa8246c96eb9ac45
    To: <sip:919283016340@10.10.10.132>;tag=as1bb82a62
    Call-ID: c1c645793a4ce95c@10.10.10.190
    CSeq: 5199 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="06e09b9b"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'c1c645793a4ce95c@10.10.10.190' in 32000 ms (Method: INVITE)
    Found user '1001'

    <--- SIP read from 10.10.10.190:5060 --->
    ACK sip:919283016340@10.10.10.132 SIP/2.0
    Via: SIP/2.0/UDP 10.10.10.190:5060;branch=z9hG4bK1daf0564d08c3f3e
    From: "namjoong01" <sip:1001@10.10.10.132>;tag=fa8246c96eb9ac45
    To: <sip:919283016340@10.10.10.132>;tag=as1bb82a62
    Contact: <sip:1001@10.10.10.190:5060;transport=udp>
    Supported: path
    Call-ID: c1c645793a4ce95c@10.10.10.190
    CSeq: 5199 ACK
    User-Agent: Grandstream GXP1200 1.1.6.44
    Max-Forwards: 70
    Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    Content-Length: 0


    <------------->
    --- (12 headers 0 lines) ---
    PBXTest*CLI>
    <--- SIP read from 10.10.10.190:5060 --->
    INVITE sip:919283016340@10.10.10.132 SIP/2.0
    Via: SIP/2.0/UDP 10.10.10.190:5060;branch=z9hG4bK9e0aa2943a6c49bc
    From: "namjoong01" <sip:1001@10.10.10.132>;tag=fa8246c96eb9ac45
    To: <sip:919283016340@10.10.10.132>
    Contact: <sip:1001@10.10.10.190:5060;transport=udp>
    Supported: replaces, timer, path
    Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:919283016340@10.10.10.132", nonce="06e09b9b", response="a695a37a411dbb4429c658bed9d4576b"
    Call-ID: c1c645793a4ce95c@10.10.10.190
    CSeq: 5200 INVITE
    User-Agent: Grandstream GXP1200 1.1.6.44
    Proxy-Require: 10.10.10.132
    Max-Forwards: 70
    Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    Content-Type: application/sdp
    Content-Length: 348

    v=0
    o=1001 8000 8001 IN IP4 10.10.10.190
    s=SIP Call
    c=IN IP4 10.10.10.190
    t=0 0
    m=audio 5014 RTP/AVP 0 8 4 18 2 97 9 3
    a=sendrecv
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:4 G723/8000
    a=rtpmap:18 G729/8000
    a=rtpmap:2 G726-32/8000
    a=rtpmap:97 iLBC/8000
    a=fmtp:97 mode=20
    a=rtpmap:9 G722/8000
    a=rtpmap:3 GSM/8000
    a=ptime:20

    <------------->
    --- (15 headers 17 lines) ---
    Sending to 10.10.10.190 : 5060 (NAT)
    Using INVITE request as basis request - c1c645793a4ce95c@10.10.10.190
    Found user '1001'
    Found RTP audio format 0
    Found RTP audio format 8
    Found RTP audio format 4
    Found RTP audio format 18
    Found RTP audio format 2
    Found RTP audio format 97
    Found RTP audio format 9
    Found RTP audio format 3
    Peer audio RTP is at port 10.10.10.190:5014
    Found audio description format PCMU for ID 0
    Found audio description format PCMA for ID 8
    Found audio description format G723 for ID 4
    Found audio description format G729 for ID 18
    Found audio description format G726-32 for ID 2
    Found audio description format iLBC for ID 97
    Found audio description format G722 for ID 9
    Found audio description format GSM for ID 3
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x1d0f (g723|gsm|ulaw|alaw|g726|g729|ilbc|g722)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
    Peer audio RTP is at port 10.10.10.190:5014
    Looking for 919283016340 in from-internal (domain 10.10.10.132)
    list_route: hop: <sip:1001@10.10.10.190:5060;transport=udp>

    <--- Transmitting (NAT) to 10.10.10.190:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.10.10.190:5060;branch=z9hG4bK9e0aa2943a6c49bc;received=10.10.10.190
    From: "namjoong01" <sip:1001@10.10.10.132>;tag=fa8246c96eb9ac45
    To: <sip:919283016340@10.10.10.132>
    Call-ID: c1c645793a4ce95c@10.10.10.190
    CSeq: 5200 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: <sip:919283016340@10.10.10.132>
    Content-Length: 0


    <------------>
    -- Executing [919283016340@from-internal:1] Macro("SIP/1001-082797e0", "user-callerid|SKIPTTL|") in new stack
    -- Executing [s@macro-user-callerid:1] NoOp("SIP/1001-082797e0", "user-callerid: device 1001") in new stack
    -- Executing [s@macro-user-callerid:2] Set("SIP/1001-082797e0", "AMPUSER=1001") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("SIP/1001-082797e0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("SIP/1001-082797e0", "1|Set|REALCALLERIDNUM=1001") in new stack
    -- Executing [s@macro-user-callerid:5] NoOp("SIP/1001-082797e0", "REALCALLERIDNUM is 1001") in new stack
    -- Executing [s@macro-user-callerid:6] Set("SIP/1001-082797e0", "AMPUSER=1001") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/1001-082797e0", "AMPUSERCIDNAME=GXP-1200") in new stack
    -- Executing [s@macro-user-callerid:8] GotoIf("SIP/1001-082797e0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:9] Set("SIP/1001-082797e0", "AMPUSERCID=1001") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/1001-082797e0", "CALLERID(all)="GXP-1200" <1001>") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/1001-082797e0", "REALCALLERIDNUM=1001") in new stack
    -- Executing [s@macro-user-callerid:12] ExecIf("SIP/1001-082797e0", "0|Set|CHANNEL(language)=") in new stack
    -- Executing [s@macro-user-callerid:13] NoOp("SIP/1001-082797e0", "TTL: ARG1: SKIPTTL") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("SIP/1001-082797e0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,23)
    -- Executing [s@macro-user-callerid:23] NoOp("SIP/1001-082797e0", "Using CallerID "GXP-1200" <1001>") in new stack
    -- Executing [919283016340@from-internal:2] Set("SIP/1001-082797e0", "_NODEST=") in new stack
    -- Executing [919283016340@from-internal:3] Macro("SIP/1001-082797e0", "record-enable|1001|OUT|") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/1001-082797e0", "0?2:4") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/1001-082797e0", "recordingcheck|20090306-021538|1236330938.4") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    recordingcheck|20090306-021538|1236330938.4: Outbound recording not enabled
    -- AGI Script recordingcheck completed, returning 0
    -- Executing [s@macro-record-enable:5] NoOp("SIP/1001-082797e0", "No recording needed") in new stack
    -- Executing [919283016340@from-internal:4] Macro("SIP/1001-082797e0", "dialout-trunk|1|19283016340|") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("SIP/1001-082797e0", "DIAL_TRUNK=1") in new stack
    -- Executing [s@macro-dialout-trunk:2] ExecIf("SIP/1001-082797e0", "0|Authenticate|") in new stack
    -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1001-082797e0", "0?disabletrunk|1") in new stack
    -- Executing [s@macro-dialout-trunk:4] Set("SIP/1001-082797e0", "DIAL_NUMBER=19283016340") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("SIP/1001-082797e0", "DIAL_TRUNK_OPTIONS=tr") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("SIP/1001-082797e0", "GROUP()=OUT_1") in new stack
    -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1001-082797e0", "1?nomax") in new stack
    -- Goto (macro-dialout-trunk,s,9)
    -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1001-082797e0", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:10] Set("SIP/1001-082797e0", "DIAL_TRUNK_OPTIONS=") in new stack
    -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1001-082797e0", "outbound-callerid|1") in new stack
    -- Executing [s@macro-outbound-callerid:1] GotoIf("SIP/1001-082797e0", "1?start") in new stack
    -- Goto (macro-outbound-callerid,s,3)
    -- Executing [s@macro-outbound-callerid:3] NoOp("SIP/1001-082797e0", "REALCALLERIDNUM is 1001") in new stack
    -- Executing [s@macro-outbound-callerid:4] GotoIf("SIP/1001-082797e0", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,9)
    -- Executing [s@macro-outbound-callerid:9] Set("SIP/1001-082797e0", "USEROUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:10] Set("SIP/1001-082797e0", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:11] Set("SIP/1001-082797e0", "TRUNKOUTCID=") in new stack
    -- Executing [s@macro-outbound-callerid:12] GotoIf("SIP/1001-082797e0", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,16)
    -- Executing [s@macro-outbound-callerid:16] GotoIf("SIP/1001-082797e0", "1?usercid") in new stack
    -- Goto (macro-outbound-callerid,s,18)
    -- Executing [s@macro-outbound-callerid:18] GotoIf("SIP/1001-082797e0", "1?report") in new stack
    -- Goto (macro-outbound-callerid,s,22)
    -- Executing [s@macro-outbound-callerid:22] NoOp("SIP/1001-082797e0", "CallerID set to "GXP-1200" <1001>") in new stack
    -- Executing [s@macro-dialout-trunk:12] AGI("SIP/1001-082797e0", "fixlocalprefix") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
    -- AGI Script fixlocalprefix completed, returning 0
    -- Executing [s@macro-dialout-trunk:13] Set("SIP/1001-082797e0", "OUTNUM=19283016340") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("SIP/1001-082797e0", "custom=ZAP/g0") in new stack
    -- Executing [s@macro-dialout-trunk:15] GotoIf("SIP/1001-082797e0", "1?gocall") in new stack
    -- Goto (macro-dialout-trunk,s,17)
    -- Executing [s@macro-dialout-trunk:17] Macro("SIP/1001-082797e0", "dialout-trunk-predial-hook|") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1001-082797e0", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:19] GotoIf("SIP/1001-082797e0", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:20] Dial("SIP/1001-082797e0", "ZAP/g0/19283016340|300|") in new stack
    == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:21] Goto("SIP/1001-082797e0", "s-CHANUNAVAIL|1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] GotoIf("SIP/1001-082797e0", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:3] NoOp("SIP/1001-082797e0", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack
    -- Executing [919283016340@from-internal:5] Macro("SIP/1001-082797e0", "outisbusy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/1001-082797e0", "all-circuits-busy-now|noanswer") in new stack
    Audio is at 10.10.10.132 port 12264
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP

    <--- Transmitting (NAT) to 10.10.10.190:5060 --->
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 10.10.10.190:5060;branch=z9hG4bK9e0aa2943a6c49bc;received=10.10.10.190
    From: "namjoong01" <sip:1001@10.10.10.132>;tag=fa8246c96eb9ac45
    To: <sip:919283016340@10.10.10.132>;tag=as2c2e8469
    Call-ID: c1c645793a4ce95c@10.10.10.190
    CSeq: 5200 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: <sip:919283016340@10.10.10.132>
    Content-Type: application/sdp
    Content-Length: 206

    v=0
    o=root 2907 2907 IN IP4 10.10.10.132
    s=session
    c=IN IP4 10.10.10.132
    t=0 0
    m=audio 12264 RTP/AVP 0 8
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    <------------>
    -- <SIP/1001-082797e0> Playing 'all-circuits-busy-now' (language 'en')
    PBXTest*CLI>
    <--- SIP read from 10.10.10.53:22486 --->



    <------------->
    -- Executing [s@macro-outisbusy:2] Playback("SIP/1001-082797e0", "pls-try-call-later|noanswer") in new stack
    -- <SIP/1001-082797e0> Playing 'pls-try-call-later' (language 'en')
    -- Executing [s@macro-outisbusy:3] Macro("SIP/1001-082797e0", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1001-082797e0", "w") in new stack
    -- Executing [s@macro-hangupcall:2] NoCDR("SIP/1001-082797e0", "") in new stack
    -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1001-082797e0", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] GotoIf("SIP/1001-082797e0", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] GotoIf("SIP/1001-082797e0", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,11)
    -- Executing [s@macro-hangupcall:11] Hangup("SIP/1001-082797e0", "") in new stack
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1001-082797e0' in macro 'hangupcall'
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1001-082797e0' in macro 'outisbusy'
    == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1001-082797e0'
    Scheduling destruction of SIP dialog 'c1c645793a4ce95c@10.10.10.190' in 32000 ms (Method: INVITE)

    <--- Reliably Transmitting (NAT) to 10.10.10.190:5060 --->
    SIP/2.0 603 Declined
    Via: SIP/2.0/UDP 10.10.10.190:5060;branch=z9hG4bK9e0aa2943a6c49bc;received=10.10.10.190
    From: "namjoong01" <sip:1001@10.10.10.132>;tag=fa8246c96eb9ac45
    To: <sip:919283016340@10.10.10.132>;tag=as2c2e8469
    Call-ID: c1c645793a4ce95c@10.10.10.190
    CSeq: 5200 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: <sip:919283016340@10.10.10.132>
    Content-Length: 0


    <------------>
    PBXTest*CLI>
    <--- SIP read from 10.10.10.190:5060 --->
    ACK sip:919283016340@10.10.10.132 SIP/2.0
    Via: SIP/2.0/UDP 10.10.10.190:5060;branch=z9hG4bK9e0aa2943a6c49bc
    From: "namjoong01" <sip:1001@10.10.10.132>;tag=fa8246c96eb9ac45
    To: <sip:919283016340@10.10.10.132>;tag=as2c2e8469
    Contact: <sip:1001@10.10.10.190:5060;transport=udp>
    Supported: path
    Proxy-Authorization: Digest username="1001", realm="asterisk", algorithm=MD5, uri="sip:919283016340@10.10.10.132", nonce="06e09b9b", response="a695a37a411dbb4429c658bed9d4576b"
    Call-ID: c1c645793a4ce95c@10.10.10.190
    CSeq: 5200 ACK
    User-Agent: Grandstream GXP1200 1.1.6.44
    Max-Forwards: 70
    Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK,MESSAGE
    Content-Length: 0


    <------------->
    --- (13 headers 0 lines) ---
    PBXTest*CLI>
    <--- SIP read from 10.10.10.190:5060 --->



    <------------->
    Really destroying SIP dialog '5aa8ea381bfd49c5154a86066c2a2769@127.0.0.1' Method: REGISTER
    PBXTest*CLI>
     
  2. tsmvision

    Joined:
    Mar 6, 2009
    Messages:
    2
    Likes Received:
    0
    sorry I solved, I just didn't setup Sip trunking provider for outbound : )
    thank you
     

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