Can't do any calls ...

Discussion in 'General' started by inforemp, Sep 16, 2009.

  1. inforemp

    Joined:
    Sep 8, 2009
    Messages:
    10
    Likes Received:
    0
    You can answer either spanish or english.

    I've been trying with every post and every manual, but i can't manage to get it to work,
    I can't call for example, if I want to dial 100000 (my voip internal numbers), from my extension, i always get "all lines busy"
    Same with incoming calls, if I dial my most likely "miss-configurated" extension, wich is 100123
    from the outside, it never gets in

    I'll paste everything i have until now, hoping someone will help me...

    [​IMG]
    [​IMG]

    Open ATA Linksys PAP2 ports: 5060 = Line 1 5061 Line


    sip.conf

    externip=cuenxxxx.ath.xx
    localnet=192.168.1.0/255.255.255.0

    sip_nat.conf

    nat=yes
    externip=cuentxxx.ath.cx
    localnet=192.168.1.0/255.255.255.0
    externrefresh=5




    trunk:

    username= ? (extension user, or voip?)
    type=peer
    secret=xxxxx
    insecure=very
    host= ? (router ip, elastix, or my voip?)
    dtmfmode=rfc2833
    disallow=all
    allow=alaw&ulaw&gsm
    canredirect=no
    canreinvite=no

    user details

    canreinvite=no
    context=from-trunk
    fromuser= ? (extension or voip?)
    qualify=no
    secret=xxxx
    type=user
    username=xxxx


    Register String


    voipuser:voippassword@200.69.159.33 (ip voip) no?


    outbound routes:

    dial pattern: 8|x.

    trunk sequence: trunk


    my sip debug:
    -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/3000-09f58aa8", "0?bypass|1") in new stack
    -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/3000-09f58aa8", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:19] Dial("SIP/3000-09f58aa8", "SIP/citarella/100000|300|") in new stack
    Audio is at 192.168.1.100 port 13526
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (NAT) to 192.168.1.50:5060:
    INVITE sip:100000@192.168.1.50 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed;rport
    From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
    To: <sip:100000@192.168.1.50>
    Contact: <sip:1003000@192.168.1.100>
    Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
    CSeq: 102 INVITE
    User-Agent: Elastix
    Max-Forwards: 70
    Date: Wed, 16 Sep 2009 12:24:28 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 287

    v=0
    o=root 3125 3125 IN IP4 192.168.1.100
    s=session
    c=IN IP4 192.168.1.100
    t=0 0
    m=audio 13526 RTP/AVP 0 8 3 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    ---
    -- Called citarella/100000
    cuentaip*CLI>
    <--- SIP read from 192.168.1.50:5060 --->
    SIP/2.0 404 Not Found
    To: <sip:100000@192.168.1.50>;tag=b81f97b570e4f770i0
    From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
    Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
    CSeq: 102 INVITE
    Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed
    Server: Linksys/PAP2T-3.1.15(LS)
    Content-Length: 0


    <------------->
    --- (8 headers 0 lines) ---
    Transmitting (NAT) to 192.168.1.50:5060:
    ACK sip:100000@192.168.1.50 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK1f6352ed;rport
    From: "1003000" <sip:1003000@192.168.1.100>;tag=as50c053f6
    To: <sip:100000@192.168.1.50>;tag=b81f97b570e4f770i0
    Contact: <sip:1003000@192.168.1.100>
    Call-ID: 21483b1906c22465184755a9611a97c7@192.168.1.100
    CSeq: 102 ACK
    User-Agent: Elastix
    Max-Forwards: 70
    Content-Length: 0


    ---
    -- SIP/citarella-09f5fa88 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [s@macro-dialout-trunk:20] Goto("SIP/3000-09f58aa8", "s-CONGESTION|1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [s-CONGESTION@macro-dialout-trunk:1] GotoIf("SIP/3000-09f58aa8", "1?noreport") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,3)
    -- Executing [s-CONGESTION@macro-dialout-trunk:3] NoOp("SIP/3000-09f58aa8", "TRUNK Dial failed due to CONGESTION - failing through to other trunks") in new stack
    -- Executing [8100000@from-internal:5] Macro("SIP/3000-09f58aa8", "outisbusy|") in new stack
    -- Executing [s@macro-outisbusy:1] Playback("SIP/3000-09f58aa8", "all-circuits-busy-now|noanswer") in new stack
    Audio is at 192.168.1.100 port 11100
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x100 (g729) to SDP
    Adding codec 0x1 (g723) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    cuentaip*CLI>
    <--- Transmitting (NAT) to 192.168.1.50:5060 --->
    SIP/2.0 183 Session Progress
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>;tag=as6734574c
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 INVITE
    User-Agent: Elastix
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Contact: <sip:8100000@192.168.1.100>
    Content-Type: application/sdp
    Content-Length: 355

    v=0
    o=root 3125 3125 IN IP4 192.168.1.100
    s=session
    c=IN IP4 192.168.1.100
    t=0 0
    m=audio 11100 RTP/AVP 0 8 18 4 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=no
    a=rtpmap:4 G723/8000
    a=fmtp:4 annexa=no
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    <------------>
    -- <SIP/3000-09f58aa8> Playing 'all-circuits-busy-now' (language 'es')
    Really destroying SIP dialog '21483b1906c22465184755a9611a97c7@192.168.1.100' Method: INVITE
    -- Executing [s@macro-outisbusy:2] Playback("SIP/3000-09f58aa8", "pls-try-call-later|noanswer") in new stack
    -- <SIP/3000-09f58aa8> Playing 'pls-try-call-later' (language 'es')
    cuentaip*CLI>
    <--- SIP read from 192.168.1.50:5060 --->
    CANCEL sip:8100000@192.168.1.100 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 CANCEL
    Max-Forwards: 70
    Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="65ea3d5a",uri="sip:8100000@192.168.1.100",algorithm=MD5,response="4afdab4c474409369b72ce7ae707f132"
    User-Agent: Linksys/PAP2T-3.1.15(LS)
    Content-Length: 0


    <------------->
    --- (10 headers 0 lines) ---
    Sending to 192.168.1.50 : 5060 (NAT)

    <--- Reliably Transmitting (NAT) to 192.168.1.50:5060 --->
    SIP/2.0 487 Request Terminated
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>;tag=as6734574c
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 INVITE
    User-Agent: Elastix
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>

    <--- Transmitting (NAT) to 192.168.1.50:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61;received=192.168.1.50
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>;tag=as6734574c
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 CANCEL
    User-Agent: Elastix
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/3000-09f58aa8' in macro 'outisbusy'
    == Spawn extension (from-internal, 8100000, 5) exited non-zero on 'SIP/3000-09f58aa8'
    -- Executing [h@from-internal:1] Macro("SIP/3000-09f58aa8", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/3000-09f58aa8", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/3000-09f58aa8", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/3000-09f58aa8", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/3000-09f58aa8", "") in new stack
    == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/3000-09f58aa8' in macro 'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/3000-09f58aa8'
    cuentaip*CLI>
    <--- SIP read from 192.168.1.50:5060 --->
    ACK sip:8100000@192.168.1.100 SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-dc72af61
    From: Linea 1 <sip:3000@192.168.1.100>;tag=634d08aa8d51d295o0
    To: <sip:8100000@192.168.1.100>;tag=as6734574c
    Call-ID: c4612e0e-bb4b4c81@192.168.1.50
    CSeq: 102 ACK
    Max-Forwards: 70
    Proxy-Authorization: Digest username="3000",realm="asterisk",nonce="65ea3d5a",uri="sip:8100000@192.168.1.100",algorithm=MD5,response="fc9351980db59df69276c8325bb1ebff"
    Contact: Linea 1 <sip:3000@192.168.1.50:5060>
    User-Agent: Linksys/PAP2T-3.1.15(LS)
    Content-Length: 0


    <------------->
    --- (11 headers 0 lines) ---
    Really destroying SIP dialog 'c4612e0e-bb4b4c81@192.168.1.50' Method: ACK
     

Share This Page