DTMF Tone "#" is not submitted correctly (including log)

Discussion in 'Windows' started by support@stc.ch, Feb 11, 2010.

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  1. support@stc.ch

    Joined:
    Feb 11, 2010
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    Hi Developers,

    we have the same problem with DTMF. I can't tell you much about SIP headers or something like this, but...

    ...we have an Asterisk 1.4 PBX. Your "iPhone" connects to it via SIP. Everything works fine except if you try to log in to the Asterisk Voice Mail System. In my example, you have to dial *97 (what works). Asterisk now asks for the Voice Mail password. Typing the correct password "518518", followed by a "#" isn't submitted correctly, see debug below. Instead your phone submits it as "51581581851". With other Softphones it works.

    Hope that helps. It would be nice if you could launch an immediate fix for this bug.


    Best Regards
    An Asterisk User :)


    <------------->
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    INVITE sip:*97@srv-pbx:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-9f661c4c0f65800b-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 1 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
    Content-Type: application/sdp
    Supported: replaces
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 282

    v=0
    o=3cxVCE 35343735 29479860 IN IP4 10.1.1.118
    s=3cxVCE Audio Call
    c=IN IP4 10.1.1.118
    t=0 0
    m=audio 40014 RTP/AVP 8 0 3 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv

    <------------->
    --- (13 headers 13 lines) ---
    Sending to 10.1.1.118 : 3653 (no NAT)
    Using INVITE request as basis request - ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    srv-pbx*CLI>
    <--- Reliably Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 407 Proxy Authentication Required
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-9f661c4c0f65800b-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as5c6af395
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 1 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="48a38d6a"
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.' in 32000 ms (Method: INVITE)
    Found user '518'
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    ACK sip:*97@srv-pbx:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-9f661c4c0f65800b-1---d8754z-;rport
    Max-Forwards: 70
    To: <sip:*97@srv-pbx:5060>;tag=as5c6af395
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 1 ACK
    Content-Length: 0


    <------------->
    --- (8 headers 0 lines) ---
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    INVITE sip:*97@srv-pbx:5060 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-c17b7f4f57297601-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 2 INVITE
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO
    Content-Type: application/sdp
    Proxy-Authorization: Digest username="518",realm="asterisk",nonce="48a38d6a",uri="sip:*97@srv-pbx:5060",response="8d9e5ad3750cf9812a61f8b496f9a422",algorithm=MD5
    Supported: replaces
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 282

    v=0
    o=3cxVCE 35343735 29479860 IN IP4 10.1.1.118
    s=3cxVCE Audio Call
    c=IN IP4 10.1.1.118
    t=0 0
    m=audio 40014 RTP/AVP 8 0 3 101
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:3 GSM/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-15
    a=ptime:20
    a=sendrecv

    <------------->
    --- (14 headers 13 lines) ---
    Sending to 10.1.1.118 : 3653 (NAT)
    Using INVITE request as basis request - ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    Found user '518'
    Found RTP audio format 8
    Found RTP audio format 0
    Found RTP audio format 3
    Found RTP audio format 101
    Found audio description format PCMA for ID 8
    Found audio description format PCMU for ID 0
    Found audio description format GSM for ID 3
    Found audio description format telephone-event for ID 101
    Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
    Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    Peer audio RTP is at port 10.1.1.118:40014
    Looking for *97 in from-internal (domain srv-pbx)
    list_route: hop: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    srv-pbx*CLI>
    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 100 Trying
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-c17b7f4f57297601-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 2 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Contact: <sip:*97@10.1.1.19>
    Content-Length: 0


    <------------>
    -- Executing [*97@from-internal:1] Answer("SIP/518-00000050", "") in new stack
    Audio is at 10.1.1.19 port 17582
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x8 (alaw) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP
    srv-pbx*CLI>
    <--- Reliably Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-c17b7f4f57297601-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 2 INVITE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Contact: <sip:*97@10.1.1.19>
    Content-Type: application/sdp
    Content-Length: 266

    v=0
    o=root 15297 15297 IN IP4 10.1.1.19
    s=session
    c=IN IP4 10.1.1.19
    t=0 0
    m=audio 17582 RTP/AVP 0 8 101
    a=rtpmap:0 PCMU/8000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:101 telephone-event/8000
    a=fmtp:101 0-16
    a=silenceSupp:eek:ff - - - -
    a=ptime:20
    a=sendrecv

    <------------>
    -- Executing [*97@from-internal:2] Wait("SIP/518-00000050", "1") in new stack
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    ACK sip:*97@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-fb796b3de4571346-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 2 ACK
    Proxy-Authorization: Digest username="518",realm="asterisk",nonce="48a38d6a",uri="sip:*97@srv-pbx:5060",response="8d9e5ad3750cf9812a61f8b496f9a422",algorithm=MD5
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 0


    <------------->
    --- (11 headers 0 lines) ---
    -- Executing [*97@from-internal:3] Macro("SIP/518-00000050", "user-callerid|") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/518-00000050", "AMPUSER=518") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/518-00000050", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/518-00000050", "1|Set|REALCALLERIDNUM=518") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/518-00000050", "AMPUSER=518") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/518-00000050", "AMPUSERCIDNAME=myuser") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/518-00000050", "0?report") in new stack
    -- Executing [s@macro-user-callerid:7] Set("SIP/518-00000050", "AMPUSERCID=518") in new stack
    -- Executing [s@macro-user-callerid:8] Set("SIP/518-00000050", "CALLERID(all)="myuser" <518>") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/518-00000050", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/518-00000050", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/518-00000050", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] NoOp("SIP/518-00000050", "Using CallerID "myuser" <518>") in new stack
    -- Executing [*97@from-internal:4] Macro("SIP/518-00000050", "get-vmcontext|518") in new stack
    -- Executing [s@macro-get-vmcontext:1] Set("SIP/518-00000050", "VMCONTEXT=default") in new stack
    -- Executing [s@macro-get-vmcontext:2] GotoIf("SIP/518-00000050", "0?200:300") in new stack
    -- Goto (macro-get-vmcontext,s,300)
    -- Executing [s@macro-get-vmcontext:300] NoOp("SIP/518-00000050", "") in new stack
    -- Executing [*97@from-internal:5] MailboxExists("SIP/518-00000050", "518@default") in new stack
    -- Executing [*97@from-internal:6] GotoIf("SIP/518-00000050", "1?mbexist") in new stack
    -- Goto (from-internal,*97,106)
    -- Executing [*97@from-internal:106] VoiceMailMain("SIP/518-00000050", "518@default") in new stack
    -- <SIP/518-00000050> Playing 'vm-password' (language 'de')
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    SUBSCRIBE sip:Unknown@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-873f0021b720085f-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: "myuser"<sip:518@srv-pbx:5060>;tag=as59262a93
    From: "myuser"<sip:518@srv-pbx:5060>;tag=421e4238
    Call-ID: NTZkNDM2NDZmYjJkNDE4ODU3ZjVhNWQxNzhlMTMyNDE.
    CSeq: 26 SUBSCRIBE
    Expires: 120
    User-Agent: 3CXPhone 4.0.10858.0
    Authorization: Digest username="518",realm="asterisk",nonce="2e3652e0",uri="sip:Unknown@10.1.1.19",response="8840dd87c2c8dd419426f14fcc387de0",algorithm=MD5
    Event: message-summary
    Content-Length: 0


    <------------->
    --- (13 headers 0 lines) ---
    Found peer '518'
    srv-pbx*CLI>
    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 401 Unauthorized
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-873f0021b720085f-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=421e4238
    To: "myuser"<sip:518@srv-pbx:5060>;tag=as59262a93
    Call-ID: NTZkNDM2NDZmYjJkNDE4ODU3ZjVhNWQxNzhlMTMyNDE.
    CSeq: 26 SUBSCRIBE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26d2f58d", stale=true
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog 'NTZkNDM2NDZmYjJkNDE4ODU3ZjVhNWQxNzhlMTMyNDE.' in 6464 ms (Method: SUBSCRIBE)
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    SUBSCRIBE sip:Unknown@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-1570cc06487bd14a-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: "myuser"<sip:518@srv-pbx:5060>;tag=as59262a93
    From: "myuser"<sip:518@srv-pbx:5060>;tag=421e4238
    Call-ID: NTZkNDM2NDZmYjJkNDE4ODU3ZjVhNWQxNzhlMTMyNDE.
    CSeq: 27 SUBSCRIBE
    Expires: 120
    User-Agent: 3CXPhone 4.0.10858.0
    Authorization: Digest username="518",realm="asterisk",nonce="26d2f58d",uri="sip:Unknown@10.1.1.19",response="bd17512416677363b31070468f1ae0ed",algorithm=MD5
    Event: message-summary
    Content-Length: 0


    <------------->
    --- (13 headers 0 lines) ---
    Found peer '518'
    Scheduling destruction of SIP dialog 'NTZkNDM2NDZmYjJkNDE4ODU3ZjVhNWQxNzhlMTMyNDE.' in 130000 ms (Method: SUBSCRIBE)

    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-1570cc06487bd14a-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=421e4238
    To: "myuser"<sip:518@srv-pbx:5060>;tag=as59262a93
    Call-ID: NTZkNDM2NDZmYjJkNDE4ODU3ZjVhNWQxNzhlMTMyNDE.
    CSeq: 27 SUBSCRIBE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Expires: 120
    Contact: <sip:Unknown@10.1.1.19>;expires=120
    Content-Length: 0


    <------------>
    Reliably Transmitting (NAT) to 10.1.1.118:3653:
    NOTIFY sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.19:5060;branch=z9hG4bK06426370;rport
    Route: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    From: "Unknown" <sip:Unknown@10.1.1.19>;tag=as59262a93
    To: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>;tag=421e4238
    Contact: <sip:Unknown@10.1.1.19>
    Call-ID: NTZkNDM2NDZmYjJkNDE4ODU3ZjVhNWQxNzhlMTMyNDE.
    CSeq: 114 NOTIFY
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Event: message-summary
    Content-Type: application/simple-message-summary
    Subscription-State: active
    Content-Length: 89

    Messages-Waiting: yes
    Message-Account: sip:*97@10.1.1.19
    Voice-Message: 2/1 (0/0)

    ---
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.19:5060;branch=z9hG4bK06426370;rport=5060
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>;tag=421e4238
    From: "Unknown"<sip:Unknown@10.1.1.19>;tag=as59262a93
    Call-ID: NTZkNDM2NDZmYjJkNDE4ODU3ZjVhNWQxNzhlMTMyNDE.
    CSeq: 114 NOTIFY
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 0


    <------------>
    Scheduling destruction of SIP dialog '1265876019-3532-NB-ZRH101@10.1.1.101' in 32000 ms (Method: REGISTER)
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    INFO sip:*97@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-bb72eb06d422554c-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 3 INFO
    Content-Type: application/dtmf-relay
    Proxy-Authorization: Digest username="518",realm="asterisk",nonce="48a38d6a",uri="sip:*97@10.1.1.19",response="6de16d49b92c4fd2484990a88ce22b22",algorithm=MD5
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 24

    Signal=5
    Duration=250

    <------------->
    --- (12 headers 2 lines) ---
    Receiving INFO!
    * DTMF-relay event received: 5

    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-bb72eb06d422554c-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 3 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    INFO sip:*97@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-837a6a72516baa78-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 4 INFO
    Content-Type: application/dtmf-relay
    Proxy-Authorization: Digest username="518",realm="asterisk",nonce="48a38d6a",uri="sip:*97@10.1.1.19",response="6de16d49b92c4fd2484990a88ce22b22",algorithm=MD5
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 24

    Signal=1
    Duration=250

    <------------->
    --- (12 headers 2 lines) ---
    Receiving INFO!
    * DTMF-relay event received: 1

    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-837a6a72516baa78-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 4 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    INFO sip:*97@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-df62a40fcb50f631-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 5 INFO
    Content-Type: application/dtmf-relay
    Proxy-Authorization: Digest username="518",realm="asterisk",nonce="48a38d6a",uri="sip:*97@10.1.1.19",response="6de16d49b92c4fd2484990a88ce22b22",algorithm=MD5
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 24

    Signal=8
    Duration=250

    <------------->
    --- (12 headers 2 lines) ---
    Receiving INFO!
    * DTMF-relay event received: 8
    srv-pbx*CLI>
    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-df62a40fcb50f631-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 5 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    INFO sip:*97@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-56619761021a251c-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 6 INFO
    Content-Type: application/dtmf-relay
    Proxy-Authorization: Digest username="518",realm="asterisk",nonce="48a38d6a",uri="sip:*97@10.1.1.19",response="6de16d49b92c4fd2484990a88ce22b22",algorithm=MD5
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 24

    Signal=5
    Duration=250

    <------------->
    --- (12 headers 2 lines) ---
    Receiving INFO!
    * DTMF-relay event received: 5

    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-56619761021a251c-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 6 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    INFO sip:*97@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-03466312ea1c655b-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 7 INFO
    Content-Type: application/dtmf-relay
    Proxy-Authorization: Digest username="518",realm="asterisk",nonce="48a38d6a",uri="sip:*97@10.1.1.19",response="6de16d49b92c4fd2484990a88ce22b22",algorithm=MD5
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 24

    Signal=1
    Duration=250

    <------------->
    --- (12 headers 2 lines) ---
    Receiving INFO!
    * DTMF-relay event received: 1
    srv-pbx*CLI>
    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-03466312ea1c655b-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 7 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    INFO sip:*97@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-160ccf6d2c53a862-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 8 INFO
    Content-Type: application/dtmf-relay
    Proxy-Authorization: Digest username="518",realm="asterisk",nonce="48a38d6a",uri="sip:*97@10.1.1.19",response="6de16d49b92c4fd2484990a88ce22b22",algorithm=MD5
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 24

    Signal=8
    Duration=250

    <------------->
    --- (12 headers 2 lines) ---
    Receiving INFO!
    * DTMF-relay event received: 8

    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-160ccf6d2c53a862-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 8 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------->
    --- (12 headers 2 lines) ---
    Receiving INFO!
    * DTMF-relay event received: #
    srv-pbx*CLI>
    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-523ac46ba33e2652-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 9 INFO
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    -- Incorrect password '51581581851' for user '518' (context = default)
    -- <SIP/518-00000050> Playing 'vm-incorrect' (language 'de')
    -- <SIP/518-00000050> Playing 'vm-password' (language 'de')
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.118:3653 --->
    BYE sip:*97@10.1.1.19 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-8d13d04ae01dd865-1---d8754z-;rport
    Max-Forwards: 70
    Contact: <sip:518@10.1.1.118:3653;rinstance=b823cd74871da9f9>
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 10 BYE
    Proxy-Authorization: Digest username="518",realm="asterisk",nonce="48a38d6a",uri="sip:*97@10.1.1.19",response="b977526ae2a9dd5d224d239716fcee25",algorithm=MD5
    User-Agent: 3CXPhone 4.0.10858.0
    Content-Length: 0


    <------------->
    --- (11 headers 0 lines) ---
    Sending to 10.1.1.118 : 3653 (NAT)
    srv-pbx*CLI>
    <--- Transmitting (NAT) to 10.1.1.118:3653 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.118:3653;branch=z9hG4bK-d8754z-8d13d04ae01dd865-1---d8754z-;received=10.1.1.118;rport=3653
    From: "myuser"<sip:518@srv-pbx:5060>;tag=bc49613b
    To: <sip:*97@srv-pbx:5060>;tag=as20977d54
    Call-ID: ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.
    CSeq: 10 BYE
    User-Agent: Asterisk PBX
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    <------------>
    -- Executing [h@from-internal:1] Macro("SIP/518-00000050", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/518-00000050", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/518-00000050", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/518-00000050", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s@macro-hangupcall:9] Hangup("SIP/518-00000050", "") in new stack
    == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/518-00000050' in macro 'hangupcall'
    == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/518-00000050'
    Really destroying SIP dialog 'ZTkwMzhjZTc2ZjVhNzkwYTBkY2ZmNTJiODQ3ZWQ3NGQ.' Method: BYE



    <------------->
    --- (10 headers 0 lines) ---
    Really destroying SIP dialog '5b02dcd479b279a232dab965375d3097@10.1.1.19' Method: OPTIONS
    Reliably Transmitting (NAT) to 10.1.1.125:30186:
    OPTIONS sip:519@10.1.1.125:30186;rinstance=b2b7960e29bc6df6 SIP/2.0
    Via: SIP/2.0/UDP 10.1.1.19:5060;branch=z9hG4bK23a424a9;rport
    From: "Unknown" <sip:Unknown@10.1.1.19>;tag=as1937582a
    To: <sip:519@10.1.1.125:30186;rinstance=b2b7960e29bc6df6>
    Contact: <sip:Unknown@10.1.1.19>
    Call-ID: 62e3eba0792d905f53e3edc40db31f2a@10.1.1.19
    CSeq: 102 OPTIONS
    User-Agent: Asterisk PBX
    Max-Forwards: 70
    Date: Thu, 11 Feb 2010 13:04:11 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces
    Content-Length: 0


    ---
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.125:30186 --->
    SIP/2.0 200 OK
    Via: SIP/2.0/UDP 10.1.1.19:5060;branch=z9hG4bK23a424a9;rport=5060
    Contact: <sip:10.1.1.125:30186>
    To: <sip:519@10.1.1.125:30186;rinstance=b2b7960e29bc6df6>;tag=6625c35c
    From: "Unknown"<sip:Unknown@10.1.1.19>;tag=as1937582a
    Call-ID: 62e3eba0792d905f53e3edc40db31f2a@10.1.1.19
    CSeq: 102 OPTIONS
    Accept: application/sdp
    Accept-Language: en
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
    User-Agent: X-Lite release 1104o stamp 56125
    Content-Length: 0


    <------------->
    --- (12 headers 0 lines) ---
    Really destroying SIP dialog '62e3eba0792d905f53e3edc40db31f2a@10.1.1.19' Method: OPTIONS
    srv-pbx*CLI>
    <--- SIP read from 10.1.1.125:30186 --->
     
  2. SY

    SY Well-Known Member
    3CX Support

    Joined:
    Jan 26, 2007
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    Hi Asterisk User

    I don't know what is the reason of this post, but...

    You can reread your log ( provided by Asterisk? ) and see following:
    1. Log says about 7 SIP INFO packets confirmed by asterisk.
    2. The sequence of SIP INFO packets delivers exact 518518# as you expected

    Do you have any idea where from asterisk got "51581581851"?

    Thanks :)
    P.S. By the way, I don't see any correlation between subject and information provided in log. There is explicit confirmation that asterisk has received #
     
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  3. EricStyles

    Joined:
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    I'm seeing something exactly like the original author is seeing but I'm pretty sure it's not with " # " DTMF tone, but it's DTMF related, from what I can see from one of my users, that uses 3cx, and my testing of the client on multiple computers.

    What happens is, when the user is prompted for input (PIN number) for a conference bridge (meetme) user enters 4 digit pin, system rejects PIN. From my Asterisk logs I see the PIN number being captured but it's adding/repeating more digits. i.e.

    user enters PIN 0037

    Asterisk captures the input as = 00003377

    Server = PBX is Asterisk 1.6

    Client = 3cx softphone (latest stable)

    Note this happens with any DTMF tone sent, VM, IVR prompts, etc ... What I think is happing is 3cx is sending the DTMF tones way too quickly. I have not run wireshark yet, nor have done any other testing to verify my diagnoses, as the fix is easy (use a different client). All other softphone clients I have tested with including Linux clients can't reproduce the above behavior, and is only indicative to 3cx.

    Just an observation (2 cents) but going through the rest of the 3cx forums it's seems obvious 3cx and DTMF tones don't play along very nicely?

    I hope this helps,

    Cheers,
    Eric
     
  4. SY

    SY Well-Known Member
    3CX Support

    Joined:
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    Eric,

    The problem has been described and commented in previous posts. There is only one log and it is provided by Asterisk.

    Could you please run wireshark to verify your diagnoses and then update your information?
    How much time you need to make it?

    Thanks
     
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  5. EricStyles

    Joined:
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    I'm a pretty busy guy at the moment, (one man IT shop) and my Asterisk logs reflect the same as the original author. Also I notices a debug on 3cx but it's not outputting anything, which is why I was going to go down analyzing packets path with wireshark. So at the moment, my user is working with a different client, if my user and myself have the time I'll try to get a packet capture and post here. Time frame wise, I'm just not sure at the moment.

    Cheers,
    Eric
     
  6. Vali_3CX

    Vali_3CX Well-Known Member
    Staff Member 3CX Support

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    Hi Eric
    Eric, which DTMF mode is your 3CXPhone configured to use?
    Regards
    vali
     
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  7. SY

    SY Well-Known Member
    3CX Support

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    3CX Phone supports three different styles of DTMF delivery. (see Connection/Advanced configuration dialog)
    It is recommended to choose only one which will provide best interoperability with your PBX.
    It is not recommended to combine styles, because some PBXes and/or software/hardware may be confused in case if each DTMF signal will be delivered simultaneously through "in-band", "RTP payload" and "SIP INFO" channels
     
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  8. EricStyles

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    Asterisk will use rfc2833 for DTMF relay by default but will switch to inband DTMF tones if the remote side does not indicate support of rfc2833 in SDP. So either or, Asterisk tries to play nice with both. I wish I had the time to help you guys out, I'm involved in a couple of other open source VoIP projects, maybe if you guys setup just a standard vanilla Asterisk Server to test with. PBX In A Flash has ISO's that will you get you up and running in no time at all.
     
  9. SY

    SY Well-Known Member
    3CX Support

    Joined:
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    Technical descriptions of how does Asterisk handle DTMFs is really interesting but not in this case.
    Asterisk collects DTMFs from SIP INFO as well (see log provided in initial post).
    Wireshark capture is not required :) .
    Asterisk logs can provide you information how DTMF signals were collected and where from Asterisk has collected "dublicates" of each digit.
    For information about how 3CX Phone delivers DTMF signals and how to configure this process, see my previous comment.

    Regards
     
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  10. leejor

    leejor Well-Known Member

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    I've had problems in the past with devices detecting in-band as well as the SIP messages resulting in double digits being detected. This usually requires some trial and error to discover which single method works best for you in all cases.
     
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