Shoretel SIP Transfer

Discussion in 'General' started by tknman0700, Nov 2, 2009.

  1. tknman0700

    Joined:
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    I have been working with Asterisk in the form of the Digium AA50 appliance for about a year now.. I am new to FreePBX so I am trying to learn what is different in this way of administration. Here is a scenario I have.

    I have a sip trunk between Elastix and the Shoretel. I can call from the Elastix to the Shoretel no problem but when I reach a menu in the shoretel and the shoretel tries to do a blind transfer I think they do it differently than one might wish... when the calls comes to my shoretel phone the Name of the called reads "exit" on my Shoretel phone. This is sort of frustrating and I am trying to determine why this is.

    Here is a sip debug of what happens...

    Code:
        -- Executing [8001@from-internal:1] Macro("SIP/1000-091104b0", "user-callerid|SKIPTTL|") in new stack
        -- Executing [s@macro-user-callerid:1] Set("SIP/1000-091104b0", "AMPUSER=1000") in new stack
        -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1000-091104b0", "0?report") in new stack
        -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1000-091104b0", "1|Set|REALCALLERIDNUM=1000") in new stack
        -- Executing [s@macro-user-callerid:4] Set("SIP/1000-091104b0", "AMPUSER=1000") in new stack
        -- Executing [s@macro-user-callerid:5] Set("SIP/1000-091104b0", "AMPUSERCIDNAME=1000") in new stack
        -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1000-091104b0", "0?report") in new stack
        -- Executing [s@macro-user-callerid:7] Set("SIP/1000-091104b0", "AMPUSERCID=1000") in new stack
        -- Executing [s@macro-user-callerid:8] Set("SIP/1000-091104b0", "CALLERID(all)="1000" <1000>") in new stack
        -- Executing [s@macro-user-callerid:9] Set("SIP/1000-091104b0", "REALCALLERIDNUM=1000") in new stack
        -- Executing [s@macro-user-callerid:10] ExecIf("SIP/1000-091104b0", "0|Set|CHANNEL(language)=") in new stack
        -- Executing [s@macro-user-callerid:11] GotoIf("SIP/1000-091104b0", "1?continue") in new stack
        -- Goto (macro-user-callerid,s,20)
        -- Executing [s@macro-user-callerid:20] NoOp("SIP/1000-091104b0", "Using CallerID "1000" <1000>") in new stack
        -- Executing [8001@from-internal:2] Set("SIP/1000-091104b0", "_NODEST=") in new stack
        -- Executing [8001@from-internal:3] Macro("SIP/1000-091104b0", "record-enable|1000|OUT|") in new stack
        -- Executing [s@macro-record-enable:1] GotoIf("SIP/1000-091104b0", "1?check") in new stack
        -- Goto (macro-record-enable,s,4)
        -- Executing [s@macro-record-enable:4] AGI("SIP/1000-091104b0", "recordingcheck|20091103-001322|1257225202.28") in new stack
        -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
      recordingcheck|20091103-001322|1257225202.28: Outbound recording not enabled
        -- AGI Script recordingcheck completed, returning 0
        -- Executing [s@macro-record-enable:5] MacroExit("SIP/1000-091104b0", "") in new stack
        -- Executing [8001@from-internal:4] Macro("SIP/1000-091104b0", "dialout-trunk|2|8001||") in new stack
        -- Executing [s@macro-dialout-trunk:1] Set("SIP/1000-091104b0", "DIAL_TRUNK=2") in new stack
        -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1000-091104b0", "0?sub-pincheck|s|1") in new stack
        -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1000-091104b0", "0?disabletrunk|1") in new stack
        -- Executing [s@macro-dialout-trunk:4] Set("SIP/1000-091104b0", "DIAL_NUMBER=8001") in new stack
        -- Executing [s@macro-dialout-trunk:5] Set("SIP/1000-091104b0", "DIAL_TRUNK_OPTIONS=tr") in new stack
        -- Executing [s@macro-dialout-trunk:6] Set("SIP/1000-091104b0", "OUTBOUND_GROUP=OUT_2") in new stack
        -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1000-091104b0", "0?nomax") in new stack
        -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/1000-091104b0", "0?chanfull") in new stack
        -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1000-091104b0", "0?skipoutcid") in new stack
        -- Executing [s@macro-dialout-trunk:10] Set("SIP/1000-091104b0", "DIAL_TRUNK_OPTIONS=") in new stack
        -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1000-091104b0", "outbound-callerid|2") in new stack
        -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1000-091104b0", "0|SetCallerPres|") in new stack
        -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1000-091104b0", "0|Set|REALCALLERIDNUM=1000") in new stack
        -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1000-091104b0", "1?normcid") in new stack
        -- Goto (macro-outbound-callerid,s,6)
        -- Executing [s@macro-outbound-callerid:6] Set("SIP/1000-091104b0", "USEROUTCID=") in new stack
        -- Executing [s@macro-outbound-callerid:7] Set("SIP/1000-091104b0", "EMERGENCYCID=") in new stack
        -- Executing [s@macro-outbound-callerid:8] Set("SIP/1000-091104b0", "TRUNKOUTCID=exit") in new stack
        -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1000-091104b0", "1?trunkcid") in new stack
        -- Goto (macro-outbound-callerid,s,12)
        -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1000-091104b0", "1|Set|CALLERID(all)=exit") in new stack
        -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/1000-091104b0", "1?exit") in new stack
        -- Goto (macro-outbound-callerid,s,11)
        -- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/1000-091104b0", "") in new stack
        -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1000-091104b0", "0|AGI|fixlocalprefix") in new stack
        -- Executing [s@macro-dialout-trunk:13] Set("SIP/1000-091104b0", "OUTNUM=8001") in new stack
        -- Executing [s@macro-dialout-trunk:14] Set("SIP/1000-091104b0", "custom=SIP/test") in new stack
        -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1000-091104b0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
        -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1000-091104b0", "dialout-trunk-predial-hook|") in new stack
        -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1000-091104b0", "") in new stack
        -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1000-091104b0", "0?bypass|1") in new stack
        -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1000-091104b0", "0?customtrunk") in new stack
        -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1000-091104b0", "SIP/test/8001|300|") in new stack
    Audio is at 10.1.2.144 port 19818
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x800 (g726) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (no NAT) to 10.1.4.3:5060:
    INVITE sip:8001@10.1.4.3 SIP/2.0
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK48074559;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>
    Contact: <sip:Unknown@10.1.2.144>
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Tue, 03 Nov 2009 05:13:22 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 312
    
    v=0
    o=root 3111 3111 IN IP4 10.1.2.144
    s=session
    c=IN IP4 10.1.2.144
    t=0 0
    m=audio 19818 RTP/AVP 0 8 3 111 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    
    ---
        -- Called test/8001
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK48074559;rport
    WWW-Authenticate: Digest realm="ShoreTel",domain="sip:shoretel.com",nonce="ShoreTel:1781834944",stale=false,algorithm=md5,opaque="0"
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>;tag=hssUA_1781834944-111523928
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 102 INVITE
    Supported: timer
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
    Content-Length: 0
    
    
    <------------->
    --- (11 headers 0 lines) ---
    Transmitting (no NAT) to 10.1.4.3:5060:
    ACK sip:8001@10.1.4.3 SIP/2.0
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK48074559;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>;tag=hssUA_1781834944-111523928
    Contact: <sip:Unknown@10.1.2.144>
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
    Audio is at 10.1.2.144 port 19818
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x800 (g726) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (no NAT) to 10.1.4.3:5060:
    INVITE sip:8001@10.1.4.3 SIP/2.0
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2030f228;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>
    Contact: <sip:Unknown@10.1.2.144>
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Authorization: Digest username="test", realm="ShoreTel", algorithm=MD5, uri="sip:shoretel.com", nonce="ShoreTel:1781834944", response="36115d8a62ccbe80e269b48e0c65428a", opaque="0"
    Date: Tue, 03 Nov 2009 05:13:22 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 312
    
    v=0
    o=root 3111 3112 IN IP4 10.1.2.144
    s=session
    c=IN IP4 10.1.2.144
    t=0 0
    m=audio 19818 RTP/AVP 0 8 3 111 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    
    ---
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2030f228;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 103 INVITE
    Supported: timer
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
    Content-Length: 0
    
    
    <------------->
    --- (10 headers 0 lines) ---
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2030f228;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 103 INVITE
    Supported: timer
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
    Content-Length: 0
    
    
    <------------->
    --- (10 headers 0 lines) ---
        -- SIP/test-0908be28 is ringing
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2030f228;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 103 INVITE
    Supported: timer
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
    Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
    Content-Type: application/sdp
    Content-Length: 174
    
    v=0
    o=ShoreTel 0 11 IN IP4 10.1.4.3
    s=-
    c=IN IP4 10.1.4.3
    t=0 0
    m=audio 10080 RTP/AVP 0 102
    a=rtpmap:0 PCMU/8000/1
    a=rtpmap:102 telephone-event/8000
    a=fmtp:102 0-15
    
    <------------->
    --- (12 headers 9 lines) ---
    Found RTP audio format 0
    Found RTP audio format 102
    Peer audio RTP is at port 10.1.4.3:10080
    Found audio description format PCMU for ID 0
    Found audio description format telephone-event for ID 102
    Capabilities: us - 0x180e (gsm|ulaw|alaw|g726|g722), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Peer audio RTP is at port 10.1.4.3:10080
    list_route: hop: <sip:1204@10.1.4.3:5060>
    set_destination: Parsing <sip:1204@10.1.4.3:5060> for address/port to send to
    set_destination: set destination to 10.1.4.3, port 5060
    Transmitting (no NAT) to 10.1.4.3:5060:
    ACK sip:1204@10.1.4.3:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK724e0f73;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    Contact: <sip:Unknown@10.1.2.144>
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 103 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
        -- SIP/test-0908be28 answered SIP/1000-091104b0
        -- Packet2Packet bridging SIP/1000-091104b0 and SIP/test-0908be28
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    INVITE sip:Unknown@10.1.2.144 SIP/2.0
    Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787734944-113813917
    Max-Forwards: 70
    Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
    Supported: timer,replaces
    From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 2 INVITE
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
    Content-Type: application/sdp
    Content-Length: 186
    
    v=0
    o=ShoreTel 0 12 IN IP4 10.1.4.3
    s=-
    c=IN IP4 10.1.4.3
    t=0 0
    m=audio 10080 RTP/AVP 0 102
    a=rtpmap:0 PCMU/8000/1
    a=rtpmap:102 telephone-event/8000
    a=fmtp:102 0-15
    a=inactive
    
    <------------->
    --- (13 headers 10 lines) ---
    Sending to 10.1.4.3 : 5060 (no NAT)
    Found RTP audio format 0
    Found RTP audio format 102
    Peer audio RTP is at port 10.1.4.3:10080
    Found audio description format PCMU for ID 0
    Found audio description format telephone-event for ID 102
    Capabilities: us - 0x180e (gsm|ulaw|alaw|g726|g722), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Peer audio RTP is at port 10.1.4.3:10080
    localhost*CLI>
    <--- Transmitting (no NAT) to 10.1.4.3:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787734944-113813917;received=10.1.4.3
    From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 2 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: <sip:Unknown@10.1.2.144>
    Content-Length: 0
    
    
    <------------>
    Audio is at 10.1.2.144 port 19818
    Adding codec 0x4 (ulaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    localhost*CLI>
    <--- Reliably Transmitting (no NAT) to 10.1.4.3:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787734944-113813917;received=10.1.4.3
    From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 2 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: <sip:Unknown@10.1.2.144>
    Content-Type: application/sdp
    Content-Length: 234
    
    v=0
    o=root 3111 3113 IN IP4 10.1.2.144
    s=session
    c=IN IP4 10.1.2.144
    t=0 0
    m=audio 19818 RTP/AVP 0 102
    a=rtpmap:0 PCMU/8000
    a=rtpmap:102 telephone-event/8000
    a=fmtp:102 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=inactive
    
    <------------>
        -- Started music on hold, class 'default', on SIP/1000-091104b0
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    ACK sip:Unknown@10.1.2.144 SIP/2.0
    Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787754944-111523936
    Max-Forwards: 70
    From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 2 ACK
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
    Content-Length: 0
    
    
    <------------->
    --- (10 headers 0 lines) ---
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    REFER sip:Unknown@10.1.2.144 SIP/2.0
    Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787756944-111523938
    Max-Forwards: 70
    Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
    Supported: replaces
    From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 3 REFER
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
    Refer-To: "1172" <sip:1172@10.1.4.3:5060>
    Referred-By: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
    Content-Length: 0
    
    
    <------------->
    --- (14 headers 0 lines) ---
    Call 77015734440f26db754a299a0a518727@10.1.2.144 got a SIP call transfer from caller: (REFER)!
    SIP transfer to extension 1172@from-internal-xfer by 1204@10.1.4.3:5060
    localhost*CLI>
    <--- Transmitting (no NAT) to 10.1.4.3:5060 --->
    SIP/2.0 202 Accepted
    Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787756944-111523938;received=10.1.4.3
    From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 3 REFER
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Contact: <sip:Unknown@10.1.2.144>
    Content-Length: 0
    
    
    <------------>
    set_destination: Parsing <sip:1204@10.1.4.3:5060> for address/port to send to
    set_destination: set destination to 10.1.4.3, port 5060
    Reliably Transmitting (no NAT) to 10.1.4.3:5060:
    NOTIFY sip:1204@10.1.4.3:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2c8d9608;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    Contact: <sip:Unknown@10.1.2.144>
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 104 NOTIFY
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Event: refer;id=3
    Subscription-state: active
    Content-Type: message/sipfrag;version=2.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Length: 21
    
    SIP/2.0 183 Ringing
    
    ---
    set_destination: Parsing <sip:1204@10.1.4.3:5060> for address/port to send to
    set_destination: set destination to 10.1.4.3, port 5060
    Reliably Transmitting (no NAT) to 10.1.4.3:5060:
    NOTIFY sip:1204@10.1.4.3:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK39284fae;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    Contact: <sip:Unknown@10.1.2.144>
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 105 NOTIFY
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Event: refer;id=3
    Subscription-state: terminated;reason=noresource
    Content-Type: message/sipfrag;version=2.0
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Length: 16
    
    SIP/2.0 200 Ok
    
    ---
        -- Stopped music on hold on SIP/1000-091104b0
        -- Executing [h@from-internal-xfer:1] Macro("SIP/1000-091104b0", "hangupcall") in new stack
        -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/1000-091104b0", "w") in new stack
        -- Executing [s@macro-hangupcall:2] NoCDR("SIP/1000-091104b0", "") in new stack
        -- Executing [s@macro-hangupcall:3] GotoIf("SIP/1000-091104b0", "1?skiprg") in new stack
        -- Goto (macro-hangupcall,s,6)
        -- Executing [s@macro-hangupcall:6] GotoIf("SIP/1000-091104b0", "1?skipblkvm") in new stack
        -- Goto (macro-hangupcall,s,9)
        -- Executing [s@macro-hangupcall:9] GotoIf("SIP/1000-091104b0", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,11)
        -- Executing [s@macro-hangupcall:11] Hangup("SIP/1000-091104b0", "") in new stack
      == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1000-091104b0' in macro 'hangupcall'
      == Spawn h extension (from-internal-xfer, s, 1) exited non-zero on 'SIP/1000-091104b0'
    Scheduling destruction of SIP dialog '77015734440f26db754a299a0a518727@10.1.2.144' in 32000 ms (Method: REFER)
      == Spawn extension (from-internal-xfer, 1172, 0) exited non-zero on 'SIP/1000-091104b0' in macro 'dialout-trunk'
      == Spawn extension (from-internal-xfer, 1172, 0) exited non-zero on 'SIP/1000-091104b0'
        -- Executing [1172@from-internal-xfer:1] Macro("SIP/1000-091104b0", "user-callerid|SKIPTTL|") in new stack
        -- Executing [s@macro-user-callerid:1] Set("SIP/1000-091104b0", "AMPUSER=1000") in new stack
        -- Executing [s@macro-user-callerid:2] GotoIf("SIP/1000-091104b0", "0?report") in new stack
        -- Executing [s@macro-user-callerid:3] ExecIf("SIP/1000-091104b0", "0|Set|REALCALLERIDNUM=") in new stack
        -- Executing [s@macro-user-callerid:4] Set("SIP/1000-091104b0", "AMPUSER=1000") in new stack
        -- Executing [s@macro-user-callerid:5] Set("SIP/1000-091104b0", "AMPUSERCIDNAME=1000") in new stack
        -- Executing [s@macro-user-callerid:6] GotoIf("SIP/1000-091104b0", "0?report") in new stack
        -- Executing [s@macro-user-callerid:7] Set("SIP/1000-091104b0", "AMPUSERCID=1000") in new stack
        -- Executing [s@macro-user-callerid:8] Set("SIP/1000-091104b0", "CALLERID(all)="1000" <1000>") in new stack
        -- Executing [s@macro-user-callerid:9] Set("SIP/1000-091104b0", "REALCALLERIDNUM=1000") in new stack
        -- Executing [s@macro-user-callerid:10] ExecIf("SIP/1000-091104b0", "0|Set|CHANNEL(language)=") in new stack
        -- Executing [s@macro-user-callerid:11] GotoIf("SIP/1000-091104b0", "1?continue") in new stack
        -- Goto (macro-user-callerid,s,20)
        -- Executing [s@macro-user-callerid:20] NoOp("SIP/1000-091104b0", "Using CallerID "1000" <1000>") in new stack
        -- Executing [1172@from-internal-xfer:2] Set("SIP/1000-091104b0", "_NODEST=") in new stack
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK2c8d9608;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 104 NOTIFY
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
    Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
    Content-Length: 0
    
    
    <------------->
    --- (10 headers 0 lines) ---
        -- Executing [1172@from-internal-xfer:3] Macro("SIP/1000-091104b0", "record-enable|1000|OUT|") in new stack
        -- Executing [s@macro-record-enable:1] GotoIf("SIP/1000-091104b0", "0?check") in new stack
        -- Executing [s@macro-record-enable:2] ResetCDR("SIP/1000-091104b0", "w") in new stack
        -- Executing [s@macro-record-enable:3] StopMonitor("SIP/1000-091104b0", "") in new stack
        -- Executing [s@macro-record-enable:4] AGI("SIP/1000-091104b0", "recordingcheck|20091103-001328|1257225202.28") in new stack
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK39284fae;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    To: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 105 NOTIFY
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: "VM-AutoAttendant" <sip:1204@10.1.4.3:5060>
    Allow: INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,MESSAGE,REFER,REGISTER,UPDATE
    Content-Length: 0
    
    
    <------------->
        -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
    --- (10 headers 0 lines) ---
    SIP Response message for INCOMING dialog NOTIFY arrived
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    BYE sip:Unknown@10.1.2.144 SIP/2.0
    Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787794944-113813923
    Max-Forwards: 70
    From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 4 BYE
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Content-Length: 0
    
    
    <------------->
    --- (9 headers 0 lines) ---
    Sending to 10.1.4.3 : 5060 (no NAT)
    Scheduling destruction of SIP dialog '77015734440f26db754a299a0a518727@10.1.2.144' in 32000 ms (Method: BYE)
    localhost*CLI>
    <--- Transmitting (no NAT) to 10.1.4.3:5060 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.4.3:5060;branch=z9hG4bK1787794944-113813923;received=10.1.4.3
    From: <sip:8001@10.1.4.3>;tag=hssUA_1781890944-113813915
    To: "exit" <sip:Unknown@10.1.2.144>;tag=as46ee42fb
    Call-ID: 77015734440f26db754a299a0a518727@10.1.2.144
    CSeq: 4 BYE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Length: 0
    
    
    <------------>
      recordingcheck|20091103-001328|1257225202.28: Outbound recording not enabled
        -- AGI Script recordingcheck completed, returning 0
        -- Executing [s@macro-record-enable:5] MacroExit("SIP/1000-091104b0", "") in new stack
        -- Executing [1172@from-internal-xfer:4] Macro("SIP/1000-091104b0", "dialout-trunk|2|1172||") in new stack
        -- Executing [s@macro-dialout-trunk:1] Set("SIP/1000-091104b0", "DIAL_TRUNK=2") in new stack
        -- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/1000-091104b0", "0?sub-pincheck|s|1") in new stack
        -- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/1000-091104b0", "0?disabletrunk|1") in new stack
        -- Executing [s@macro-dialout-trunk:4] Set("SIP/1000-091104b0", "DIAL_NUMBER=1172") in new stack
        -- Executing [s@macro-dialout-trunk:5] Set("SIP/1000-091104b0", "DIAL_TRUNK_OPTIONS=tr") in new stack
        -- Executing [s@macro-dialout-trunk:6] Set("SIP/1000-091104b0", "OUTBOUND_GROUP=OUT_2") in new stack
        -- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/1000-091104b0", "0?nomax") in new stack
        -- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/1000-091104b0", "0?chanfull") in new stack
        -- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/1000-091104b0", "0?skipoutcid") in new stack
        -- Executing [s@macro-dialout-trunk:10] Set("SIP/1000-091104b0", "DIAL_TRUNK_OPTIONS=") in new stack
        -- Executing [s@macro-dialout-trunk:11] Macro("SIP/1000-091104b0", "outbound-callerid|2") in new stack
        -- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/1000-091104b0", "0|SetCallerPres|") in new stack
        -- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/1000-091104b0", "0|Set|REALCALLERIDNUM=1000") in new stack
        -- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/1000-091104b0", "1?normcid") in new stack
        -- Goto (macro-outbound-callerid,s,6)
        -- Executing [s@macro-outbound-callerid:6] Set("SIP/1000-091104b0", "USEROUTCID=") in new stack
        -- Executing [s@macro-outbound-callerid:7] Set("SIP/1000-091104b0", "EMERGENCYCID=") in new stack
        -- Executing [s@macro-outbound-callerid:8] Set("SIP/1000-091104b0", "TRUNKOUTCID=exit") in new stack
        -- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/1000-091104b0", "1?trunkcid") in new stack
        -- Goto (macro-outbound-callerid,s,12)
        -- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/1000-091104b0", "1|Set|CALLERID(all)=exit") in new stack
        -- Executing [s@macro-outbound-callerid:13] GotoIf("SIP/1000-091104b0", "1?exit") in new stack
        -- Goto (macro-outbound-callerid,s,11)
        -- Executing [s@macro-outbound-callerid:11] MacroExit("SIP/1000-091104b0", "") in new stack
        -- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/1000-091104b0", "0|AGI|fixlocalprefix") in new stack
        -- Executing [s@macro-dialout-trunk:13] Set("SIP/1000-091104b0", "OUTNUM=1172") in new stack
        -- Executing [s@macro-dialout-trunk:14] Set("SIP/1000-091104b0", "custom=SIP/test") in new stack
        -- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/1000-091104b0", "0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)") in new stack
        -- Executing [s@macro-dialout-trunk:16] Macro("SIP/1000-091104b0", "dialout-trunk-predial-hook|") in new stack
        -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/1000-091104b0", "") in new stack
        -- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/1000-091104b0", "0?bypass|1") in new stack
        -- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/1000-091104b0", "0?customtrunk") in new stack
        -- Executing [s@macro-dialout-trunk:19] Dial("SIP/1000-091104b0", "SIP/test/1172|300|") in new stack
    Audio is at 10.1.2.144 port 19304
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x800 (g726) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (no NAT) to 10.1.4.3:5060:
    INVITE sip:1172@10.1.4.3 SIP/2.0
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK64a2ecfc;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
    To: <sip:1172@10.1.4.3>
    Contact: <sip:Unknown@10.1.2.144>
    Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Tue, 03 Nov 2009 05:13:29 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 312
    
    v=0
    o=root 3111 3111 IN IP4 10.1.2.144
    s=session
    c=IN IP4 10.1.2.144
    t=0 0
    m=audio 19304 RTP/AVP 0 8 3 111 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    
    ---
        -- Called test/1172
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK64a2ecfc;rport
    WWW-Authenticate: Digest realm="ShoreTel",domain="sip:shoretel.com",nonce="ShoreTel:1787926944",stale=false,algorithm=md5,opaque="0"
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
    To: <sip:1172@10.1.4.3>;tag=hssUA_1787926944-111523940
    Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
    CSeq: 102 INVITE
    Supported: timer
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
    Content-Length: 0
    
    
    <------------->
    --- (11 headers 0 lines) ---
    Transmitting (no NAT) to 10.1.4.3:5060:
    ACK sip:1172@10.1.4.3 SIP/2.0
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK64a2ecfc;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
    To: <sip:1172@10.1.4.3>;tag=hssUA_1787926944-111523940
    Contact: <sip:Unknown@10.1.2.144>
    Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
    CSeq: 102 ACK
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Content-Length: 0
    
    
    ---
    Audio is at 10.1.2.144 port 19304
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding codec 0x2 (gsm) to SDP
    Adding codec 0x800 (g726) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    Reliably Transmitting (no NAT) to 10.1.4.3:5060:
    INVITE sip:1172@10.1.4.3 SIP/2.0
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK116e6ec8;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
    To: <sip:1172@10.1.4.3>
    Contact: <sip:Unknown@10.1.2.144>
    Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
    CSeq: 103 INVITE
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Authorization: Digest username="test", realm="ShoreTel", algorithm=MD5, uri="sip:shoretel.com", nonce="ShoreTel:1787926944", response="98e4f7115557d1ef106e255365c66d52", opaque="0"
    Date: Tue, 03 Nov 2009 05:13:29 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
    Supported: replaces
    Content-Type: application/sdp
    Content-Length: 312
    
    v=0
    o=root 3111 3112 IN IP4 10.1.2.144
    s=session
    c=IN IP4 10.1.2.144
    t=0 0
    m=audio 19304 RTP/AVP 0 8 3 111 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:111 G726-32/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:off - - - -
    a=ptime:20
    a=sendrecv
    
    ---
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK116e6ec8;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
    To: <sip:1172@10.1.4.3>
    Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
    CSeq: 103 INVITE
    Supported: timer
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
    Content-Length: 0
    
    
    <------------->
    --- (10 headers 0 lines) ---
    localhost*CLI>
    <--- SIP read from 10.1.4.3:5060 --->
    SIP/2.0 180 Ringing
    Via: SIP/2.0/UDP 10.1.2.144:5060;branch=z9hG4bK116e6ec8;rport
    From: "exit" <sip:Unknown@10.1.2.144>;tag=as0e5ae63a
    To: <sip:1172@10.1.4.3>;tag=hssUA_1788012944-113813927
    Call-ID: 69441aee56c2f80310352f0f599a1d57@10.1.2.144
    CSeq: 103 INVITE
    Supported: timer
    User-Agent: ShoreTel_FSS_SIP_UATK ShoreTel 7
    Contact: DefaultProfile <sip:DefaultProfile@10.1.4.3:5060>
    Content-Length: 0
    
    
    <------------->
    --- (10 headers 0 lines) ---
        -- SIP/test-b7d0f8d0 is ringing
    localhost*CLI> exit
    [root@localhost ~]#
    
    I dialed from the Elastix to the Shoretel via 8001. The call was bridged to the Shoretel Auto Attendant. When I dialed 1172 into that AA menu the call is bridged to my Shoretel phone but the callerid name is "exit". My understanding of this is because the Shoretel sends a referral in a way that confuses the Asterisk system into thinking it is an extension local to the machine. I setup a outbound route over the sip trunk that looks like this _1XXX to dial across the sip trunk and the call is bridged but I am really trying to understand why this "exit" is present as the caller id name.

    Thanks
     

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