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Thread: Call-Info header removal

  1. #1

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    according to section 20 of the RFC 3162, a proxy may not remove a Call-Info header from any request. However, that's just what the Quadro does when I try to initiate a directed page from a Linksys phone (they have a phone specific prefix to page a specific extension).

    Here's what happens if I try to page extension 52 from extension 56.

    Code:
    15:28:54 Receive SIP message # (14/10/2007 13:28:54:588 GMT) # UDP #
    1034 bytes # from: 192.168.1.100:5065 # to: 192.168.1.10:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    INVITE sip:52@192.168.1.10 SIP/2.0
    
    Via: SIP/2.0/UDP 192.168.1.100:5065;branch=z9hG4bK-3d379873
    
    From: "Stephan" <sip:56@192.168.1.10>;tag=2931f4831b06df3o5
    
    To: <sip:52@192.168.1.10>
    
    Call-Info: <sip:192.168.1.10>;answer-after=0
    
    Call-ID: eafbaf74-236c07b3@192.168.1.100
    
    CSeq: 101 INVITE
    
    Max-Forwards: 70
    
    Contact: "Stephan"  <sip:56@192.168.1.100:5065>;+sip.instance="<0000000 0-0000-0000-0000-000E08DD6BFA>"
     
    Expires: 240
    
    User-Agent: Linksys/SPA962-5.1.15(aSC)
    
    Content-Length: 397
    
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE
    
    Allow-Events: dialog
    
    Supported: replaces
    
    Content-Type: application/sdp
    
    
    
    v=0
    
    o=- 384539 384539 IN IP4 192.168.1.100
    
    s=-
    
    c=IN IP4 192.168.1.100
    
    t=0 0
    
    m=audio 16420 RTP/AVP 8 0 2 4 18 96 97 98 101
    
    a=rtpmap:8 PCMA/8000
    
    a=rtpmap:0 PCMU/8000
    
    a=rtpmap:2 G726-32/8000
    
    a=rtpmap:4 G723/8000
    
    a=rtpmap:18 G729a/8000
    
    a=rtpmap:96 G726-40/8000
    
    a=rtpmap:97 G726-24/8000
    
    a=rtpmap:98 G726-16/8000
    
    a=rtpmap:101 telephone-event/8000
    
    a=fmtp:101 0-15
    
    a=ptime:30
    
    a=sendrecv
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:28:54 Try to send SIP message # (14/10/2007 13:28:54:593 GMT) #
    UDP # 250 bytes # from: 192.168.1.10:5060 # to: 192.168.1.100:5065
    
    
    
    ***************************** SIP message buffer start *****************************
    
    SIP/2.0 100 Trying
    
    Via: SIP/2.0/UDP 192.168.1.100:5065;branch=z9hG4bK-3d379873
    
    To: <sip:52@192.168.1.10>
    
    From: "Stephan" <sip:56@192.168.1.10>;tag=2931f4831b06df3o5
    
    CSeq: 101 INVITE
    
    Call-ID: eafbaf74-236c07b3@192.168.1.100
    
    Content-Length: 0
    
    
    
    *****************************  SIP message buffer end ******************************
    
    
    
    
    ------------------- Application Log Started At 2007/10/14 15:28:54 -------------------
    
    15:28:54 TLayer::MsgToTU # Msg type: 51 # TID: 7128 # DID: 0
    
    15:28:54 UACore::TLReqMsgProc # Got INVITE SIP request
    
    15:28:54 Try to send SIP message # (14/10/2007 13:28:54:609 GMT) #
    UDP # 432 bytes # from: 192.168.1.10:5060 # to: 192.168.1.100:5065
    
    
    
    ***************************** SIP message buffer start *****************************
    
    SIP/2.0 401 Unauthorized
    
    Via: SIP/2.0/UDP 192.168.1.100:5065;branch=z9hG4bK-3d379873
    
    To: <sip:52@192.168.1.10>
    
    From: "Stephan" <sip:56@192.168.1.10>;tag=2931f4831b06df3o5
    
    CSeq: 101 INVITE
    
    Call-ID: eafbaf74-236c07b3@192.168.1.100
    
    Server: Epygi Quadro SIP User Agent/v4.1.33 (QUADRO-2X)
    
    WWW-Authenticate: Digest  realm="quadro.epygi-config.com",nonce="752dc605b4398a9ebe02b 0e8e8fa65ce",opaque="1192368534"
     
    Content-Length: 0
    
    
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:28:54 Receive SIP message # (14/10/2007 13:28:54:624 GMT) # UDP
    # 443 bytes # from: 192.168.1.100:5065 # to: 192.168.1.10:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    ACK sip:52@192.168.1.10 SIP/2.0
    
    Via: SIP/2.0/UDP 192.168.1.100:5065;branch=z9hG4bK-3d379873
    
    From: "Stephan" <sip:56@192.168.1.10>;tag=2931f4831b06df3o5
    
    To: <sip:52@192.168.1.10>
    
    Call-ID: eafbaf74-236c07b3@192.168.1.100
    
    CSeq: 101 ACK
    
    Max-Forwards: 70
    
    Contact: "Stephan"  <sip:56@192.168.1.100:5065>;+sip.instance="<0000000 0-0000-0000-0000-000E08DD6BFA>"
     
    User-Agent: Linksys/SPA962-5.1.15(aSC)
    
    Content-Length: 0
    
    Allow-Events: dialog
    
    
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:28:54 Receive SIP message # (14/10/2007 13:28:54:635 GMT) # UDP
    # 1248 bytes # from: 192.168.1.100:5065 # to: 192.168.1.10:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    INVITE sip:52@192.168.1.10 SIP/2.0
    
    Via: SIP/2.0/UDP 192.168.1.100:5065;branch=z9hG4bK-d6ad5c83
    
    From: "Stephan" <sip:56@192.168.1.10>;tag=2931f4831b06df3o5
    
    To: <sip:52@192.168.1.10>
    
    Call-Info: <sip:192.168.1.10>;answer-after=0
    
    Call-ID: eafbaf74-236c07b3@192.168.1.100
    
    CSeq: 102 INVITE
    
    Max-Forwards: 70
    
    Authorization:  Digest
    username="56",realm="quadro.epygi-config.com",nonce= "752dc605b4398a9ebe02b0e8e8fa65ce",uri="sip:52@192.168.1.10" ,algorithm=MD5,response="29f03d26c97df4441714f0178442d5df",o paque="1192368534"
    
    Contact: "Stephan"  <sip:56@192.168.1.100:5065>;+sip.instance="<0000000 0-0000-0000-0000-000E08DD6BFA>"
     
    Expires: 240
    
    User-Agent: Linksys/SPA962-5.1.15(aSC)
    
    Content-Length: 397
    
    Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER, SUBSCRIBE
    
    Allow-Events: dialog
    
    Supported: replaces
    
    Content-Type: application/sdp
    
    
    
    v=0
    
    o=- 384539 384539 IN IP4 192.168.1.100
    
    s=-
    
    c=IN IP4 192.168.1.100
    
    t=0 0
    
    m=audio 16420 RTP/AVP 8 0 2 4 18 96 97 98 101
    
    a=rtpmap:8 PCMA/8000
    
    a=rtpmap:0 PCMU/8000
    
    a=rtpmap:2 G726-32/8000
    
    a=rtpmap:4 G723/8000
    
    a=rtpmap:18 G729a/8000
    
    a=rtpmap:96 G726-40/8000
    
    a=rtpmap:97 G726-24/8000
    
    a=rtpmap:98 G726-16/8000
    
    a=rtpmap:101 telephone-event/8000
    
    a=fmtp:101 0-15
    
    a=ptime:30
    
    a=sendrecv
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:28:54 Try to send SIP message # (14/10/2007 13:28:54:654 GMT) #
    UDP # 250 bytes # from: 192.168.1.10:5060 # to: 192.168.1.100:5065
    
    
    
    ***************************** SIP message buffer start *****************************
    
    SIP/2.0 100 Trying
    
    Via: SIP/2.0/UDP 192.168.1.100:5065;branch=z9hG4bK-d6ad5c83
    
    To: <sip:52@192.168.1.10>
    
    From: "Stephan" <sip:56@192.168.1.10>;tag=2931f4831b06df3o5
    
    CSeq: 102 INVITE
    
    Call-ID: eafbaf74-236c07b3@192.168.1.100
    
    Content-Length: 0
    
    
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:28:54 TLayer::MsgToTU # Msg type: 51 # TID: 7129 # DID: 0
    
    15:28:54 UACore::TLReqMsgProc # Got INVITE SIP request
    
    15:28:54 SipSessDlg: # Call ID Info # SID: 5121183858310124256 # 15:28:54 SipID: eafbaf74-236c07b3@192.168.1.100
    
    15:28:54 SipSessDlg::ChangeSessTimerFlag # user 56 registered on UA # SessionTimer = 0
    
    15:28:54 SipSessDlg::ChangeSessTimerFlag # user 56 # SessionTimer = 0
    
    15:28:54 SipSessDlg::IncInviteProc # Got INVITE message # OID: 7132 # SID: 5121183858310124256
    
    15:28:54 TargetQualifier::QualifyTargets # Try to qualify targets for dest 192.168.1.100 # OID: 7132
    
    15:28:54 TargetQualifier::NatHandling # Don't need nat settings - direct reach # OID: 7132
    
    15:28:54 UA --> CM # MakeCall # from: 56@192.168.1.10:, to: 52,
    child: (empty), media exist, GUID: (empty) # ContactInfo: 192.168.1.100
    # SID: 5121183858310124256
    
    15:28:55 CallsAgent::OnMakeCall # MakeCall from CM # SID: 5094685911557058
    
    15:28:55 CallsAgent::OnMakeCall # Empty GUID from CM # SID: 5094685911557058
    
    15:28:55 SipSessDlg: # Call ID Info # SID: 5094685911557058 #
    15:28:55  SipID:
    a5b8dcbf-bc75-4fc2-8f58-4f5fe5962758@quadro.epygi-co nfig.com
    
    15:28:55 SipSessDlg::ChangeSessTimerFlag # user 52 registered on UA # SessionTimer = 0
    
    15:28:55 SipSessDlg::ChangeSessTimerFlag # user 52 # SessionTimer = 0
    
    15:28:55 CM --> UA # MakeCall # from: 56, to:
    52@192.168.1.104:1025, media exist # replaceSID: 0 # privacy: 0 #
    AlertInfo: not exist # AddInfo: [SP#192.168.1.10:5060][BND#52] #OID:
    7138 # SID: 5094685911557058
    
    15:28:55 SipDlg::DetermineInitTarget # Call to primary server # OID: 7138
    
    15:28:55 TargetQualifier::QualifyTargets # Try to qualify targets for dest 192.168.1.104 # OID: 7138
    
    15:28:55 TargetQualifier::NatHandling # Don't need nat settings - direct reach # OID: 7138
    
    15:28:55 SipSessDlg::SelfMediaHandling # State: local offer, Peer
    Public Key: empty, SRTP flag: false, Sesssion Key: empty # OID: 7138 #
    SID: 5094685911557058
    
    15:28:55 Try to send SIP message # (14/10/2007 13:28:55:068 GMT) #
    UDP # 896 bytes # from: 192.168.1.10:5060 # to: 192.168.1.104:1025
    
    
    
    ***************************** SIP message buffer start *****************************
    
    INVITE sip:52@192.168.1.104:1025 SIP/2.0
    
    Via: SIP/2.0/UDP  192.168.1.10:5060;rport;branch=z9hG4bKEPSVBUSddce5f6e-a602-4 f07-bd63-eb0fde95fc4b
    
    To: <sip:52@192.168.1.10:5060>
    
    From: "Stephan SPA962"  <sip:56@192.168.1.10>;tag=1192355697b378de45-fb89-4bf6 -9dc7-c7f84e1fea7c
    
    CSeq: 213 INVITE
    
    Call-ID:  a5b8dcbf-bc75-4fc2-8f58-4f5fe5962758...ygi-config.com 
    
    Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, UPDATE
    
    Contact: "Stephan SPA962" <sip:56@192.168.1.10:5060>
    
    Content-Type: application/sdp
    
    Supported: replaces, norefersub
    
    User-Agent: Epygi Quadro SIP User Agent/v4.1.33 (QUADRO-2X)
    
    Max-Forwards: 70
    
    Content-Length: 225
    
    
    
    v=0
    
    o=56 354 89 IN IP4 192.168.1.10
    
    s=-
    
    c=IN IP4 192.168.1.10
    
    t=0 0
    
    m=audio 6046 RTP/AVP 0 8 18 101
    
    a=rtpmap:0 PCMU/8000
    
    a=rtpmap:8 PCMA/8000
    
    a=rtpmap:18 G729/8000
    
    a=rtpmap:101 telephone-event/8000
    
    a=fmtp:101 0-15
    
    *****************************  SIP message buffer end ******************************
    As you can see, the originating phone tries to page by use of

    Call-Info: <sip:192.168.1.10>;answer-after=0

    After authentication of the request, the quadro forwards the INVITE to the destination after having removed that header field. Thus, the page returns in a regular call.

    The same happens when I initiate a directed page from a grandstream phone:

    Code:
    15:35:32 Receive SIP message # (14/10/2007 13:35:32:060 GMT) # UDP #
    1023 bytes # from: 192.168.1.122:5060 # to: 192.168.1.10:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    INVITE sip:52@192.168.1.10;user=phone SIP/2.0
    
    Via: SIP/2.0/UDP 192.168.1.122:5060;branch=z9hG4bK33817644ce07d017
    
    From: "Stephan GXP2020"  <sip:54@192.168.1.10;user=phone>;tag=5e7391b6b1a6cfa6
     
    
    To: <sip:52@192.168.1.10;user=phone>
    
    Contact: <sip:54@192.168.1.122:5060;transport=udp;user=phone>
    
    Supported: replaces, timer, path
    
    Call-ID: 0785a080d29041d5@192.168.1.122
    
    Call-Info: answer-after=0
    
    CSeq: 22063 INVITE
    
    User-Agent: Grandstream GXP2020 1.1.4.22
    
    Max-Forwards: 70
    
    Allow:  INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UP DATE,PRACK,MESSAGE
    
    Content-Type: application/sdp
    
    Content-Length: 405
    
    
    
    v=0
    
    o=54 8000 8000 IN IP4 192.168.1.122
    
    s=SIP Call
    
    c=IN IP4 192.168.1.122
    
    t=0 0
    
    m=audio 5004 RTP/AVP 8 0 4 18 2 97 9 3 101
    
    a=sendrecv
    
    a=rtpmap:8 PCMA/8000
    
    a=rtpmap:0 PCMU/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 G722/16000
    
    a=rtpmap:3 GSM/8000
    
    a=ptime:20
    
    a=rtpmap:101 telephone-event/8000
    
    a=fmtp:101 0-11
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:35:32 Try to send SIP message # (14/10/2007 13:35:32:064 GMT) #
    UDP # 287 bytes # from: 192.168.1.10:5060 # to: 192.168.1.122:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    SIP/2.0 100 Trying
    
    Via: SIP/2.0/UDP 192.168.1.122:5060;branch=z9hG4bK33817644ce07d017
    
    To: <sip:52@192.168.1.10;user=phone>
    
    From: "Stephan GXP2020"  <sip:54@192.168.1.10;user=phone>;tag=5e7391b6b1a6cfa6
     
    
    CSeq: 22063 INVITE
    
    Call-ID: 0785a080d29041d5@192.168.1.122
    
    Content-Length: 0
    
    
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:35:32 TLayer::MsgToTU # Msg type: 51 # TID: 7346 # DID: 0
    
    15:35:32 UACore::TLReqMsgProc # Got INVITE SIP request
    
    15:35:32 Try to send SIP message # (14/10/2007 13:35:32:089 GMT) #
    UDP # 469 bytes # from: 192.168.1.10:5060 # to: 192.168.1.122:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    SIP/2.0 401 Unauthorized
    
    Via: SIP/2.0/UDP 192.168.1.122:5060;branch=z9hG4bK33817644ce07d017
    
    To: <sip:52@192.168.1.10;user=phone>
    
    From: "Stephan GXP2020"  <sip:54@192.168.1.10;user=phone>;tag=5e7391b6b1a6cfa6
     
    
    CSeq: 22063 INVITE
    
    Call-ID: 0785a080d29041d5@192.168.1.122
    
    Server: Epygi Quadro SIP User Agent/v4.1.33 (QUADRO-2X)
    
    WWW-Authenticate: Digest  realm="quadro.epygi-config.com",nonce="5de995c77e703bdefa036 64b949d5568",opaque="1192368932"
     
    Content-Length: 0
    
    
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:35:32 Receive SIP message # (14/10/2007 13:35:32:143 GMT) # UDP
    # 535 bytes # from: 192.168.1.122:5060 # to: 192.168.1.10:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    ACK sip:52@192.168.1.10;user=phone SIP/2.0
    
    Via: SIP/2.0/UDP 192.168.1.122:5060;branch=z9hG4bK33817644ce07d017
    
    From: "Stephan GXP2020"  <sip:54@192.168.1.10;user=phone>;tag=5e7391b6b1a6cfa6
     
    
    To: <sip:52@192.168.1.10;user=phone>
    
    Contact: <sip:54@192.168.1.122:5060;transport=udp;user=phone>
    
    Supported: path
    
    Call-ID: 0785a080d29041d5@192.168.1.122
    
    CSeq: 22063 ACK
    
    User-Agent: Grandstream GXP2020 1.1.4.22
    
    Max-Forwards: 70
    
    Allow:  INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UP DATE,PRACK,MESSAGE
    
    Content-Length: 0
    
    
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:35:32 Receive SIP message # (14/10/2007 13:35:32:176 GMT) # UDP
    # 1254 bytes # from: 192.168.1.122:5060 # to: 192.168.1.10:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    INVITE sip:52@192.168.1.10;user=phone SIP/2.0
    
    Via: SIP/2.0/UDP 192.168.1.122:5060;branch=z9hG4bK8d80017cdfd5179f
    
    From: "Stephan GXP2020"  <sip:54@192.168.1.10;user=phone>;tag=5e7391b6b1a6cfa6
     
    
    To: <sip:52@192.168.1.10;user=phone>
    
    Contact: <sip:54@192.168.1.122:5060;transport=udp;user=phone>
    
    Supported: replaces, timer, path
    
    Authorization: Digest username="54",
    realm="quadro.epygi-config.com", algorithm=MD5,
    uri="sip:52@192.168.1.10;user=phone",  opaque="1192368932",
    nonce="5de995c77e703bdefa03664b949d556 8",
    response="73c4ce938c7231667bb74c0dac7576cc"
     
    Call-ID: 0785a080d29041d5@192.168.1.122
    
    Call-Info: answer-after=0
    
    CSeq: 22064 INVITE
    
    User-Agent: Grandstream GXP2020 1.1.4.22
    
    Max-Forwards: 70
    
    Allow:  INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UP DATE,PRACK,MESSAGE
    
    Content-Type: application/sdp
    
    Content-Length: 405
    
    
    
    v=0
    
    o=54 8000 8001 IN IP4 192.168.1.122
    
    s=SIP Call
    
    c=IN IP4 192.168.1.122
    
    t=0 0
    
    m=audio 5004 RTP/AVP 8 0 4 18 2 97 9 3 101
    
    a=sendrecv
    
    a=rtpmap:8 PCMA/8000
    
    a=rtpmap:0 PCMU/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 G722/16000
    
    a=rtpmap:3 GSM/8000
    
    a=ptime:20
    
    a=rtpmap:101 telephone-event/8000
    
    a=fmtp:101 0-11
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:35:32 Try to send SIP message # (14/10/2007 13:35:32:181 GMT) #
    UDP # 287 bytes # from: 192.168.1.10:5060 # to: 192.168.1.122:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    SIP/2.0 100 Trying
    
    Via: SIP/2.0/UDP 192.168.1.122:5060;branch=z9hG4bK8d80017cdfd5179f
    
    To: <sip:52@192.168.1.10;user=phone>
    
    From: "Stephan GXP2020"  <sip:54@192.168.1.10;user=phone>;tag=5e7391b6b1a6cfa6
     
    
    CSeq: 22064 INVITE
    
    Call-ID: 0785a080d29041d5@192.168.1.122
    
    Content-Length: 0
    
    
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:35:32 TLayer::MsgToTU # Msg type: 51 # TID: 7347 # DID: 0
    
    15:35:32 UACore::TLReqMsgProc # Got INVITE SIP request
    
    15:35:32 SipSessDlg: # Call ID Info # SID: 5121185567706633984 # 15:35:32 SipID: 0785a080d29041d5@192.168.1.122
    
    15:35:32 SipSessDlg::ChangeSessTimerFlag # user 54 registered on UA # SessionTimer = 0
    
    15:35:32 SipSessDlg::ChangeSessTimerFlag # user 54 # SessionTimer = 0
    
    15:35:32 SipSessDlg::IncInviteProc # Got INVITE message # OID: 7348 # SID: 5121185567706633984
    
    15:35:32 TargetQualifier::QualifyTargets # Try to qualify targets for dest 192.168.1.122 # OID: 7348
    
    15:35:32 TargetQualifier::NatHandling # Don't need nat settings - direct reach # OID: 7348
    
    15:35:32 UA --> CM # MakeCall # from: 54@192.168.1.10:, to: 52,
    child: (empty), media exist, GUID: (empty) # ContactInfo: 192.168.1.122
    # SID: 5121185567706633984
    
    15:35:32 CallsAgent::OnMakeCall # MakeCall from CM # SID: 5096391013967042
    
    15:35:32 CallsAgent::OnMakeCall # Empty GUID from CM # SID: 5096391013967042
    
    15:35:32 SipSessDlg: # Call ID Info # SID: 5096391013967042 #
    15:35:32  SipID:
    8eb9cf49-57a7-4f3d-919d-9c9f9f664a07@quadro.epygi-co nfig.com
    
    15:35:32 SipSessDlg::ChangeSessTimerFlag # user 52 registered on UA # SessionTimer = 0
    
    15:35:32 SipSessDlg::ChangeSessTimerFlag # user 52 # SessionTimer = 0
    
    15:35:32 CM --> UA # MakeCall # from: 54, to:
    52@192.168.1.104:1025, media exist # replaceSID: 0 # privacy: 0 #
    AlertInfo: not exist # AddInfo: [SP#192.168.1.10:5060][BND#52] #OID:
    7351 # SID: 5096391013967042
    
    15:35:32 SipDlg::DetermineInitTarget # Call to primary server # OID: 7351
    
    15:35:32 TargetQualifier::QualifyTargets # Try to qualify targets for dest 192.168.1.104 # OID: 7351
    
    15:35:32 TargetQualifier::NatHandling # Don't need nat settings - direct reach # OID: 7351
    
    15:35:32 SipSessDlg::SelfMediaHandling # State: local offer, Peer
    Public Key: empty, SRTP flag: false, Sesssion Key: empty # OID: 7351 #
    SID: 5096391013967042
    
    15:35:32 Try to send SIP message # (14/10/2007 13:35:32:505 GMT) #
    UDP # 906 bytes # from: 192.168.1.10:5060 # to: 192.168.1.104:1025
    
    
    
    ***************************** SIP message buffer start *****************************
    
    INVITE sip:52@192.168.1.104:1025 SIP/2.0
    
    Via: SIP/2.0/UDP  192.168.1.10:5060;rport;branch=z9hG4bKEPSVBUS616baa10-6b48-4 a85-92c1-7f4b0cac6c34
    
    To: <sip:52@192.168.1.10:5060>
    
    From: "Stephan Grandstream"  <sip:54@192.168.1.10>;tag=1192355697e6c8e9cb-7d94-45eb -ad2b-04013f14e4b3
    
    CSeq: 335 INVITE
    
    Call-ID:  8eb9cf49-57a7-4f3d-919d-9c9f9f664a07...ygi-config.com 
    
    Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, UPDATE
    
    Contact: "Stephan Grandstream" <sip:54@192.168.1.10:5060>
    
    Content-Type: application/sdp
    
    Supported: replaces, norefersub
    
    User-Agent: Epygi Quadro SIP User Agent/v4.1.33 (QUADRO-2X)
    
    Max-Forwards: 70
    
    Content-Length: 225
    
    
    
    v=0
    
    o=54 34 321 IN IP4 192.168.1.10
    
    s=-
    
    c=IN IP4 192.168.1.10
    
    t=0 0
    
    m=audio 6050 RTP/AVP 0 8 18 101
    
    a=rtpmap:0 PCMU/8000
    
    a=rtpmap:8 PCMA/8000
    
    a=rtpmap:18 G729/8000
    
    a=rtpmap:101 telephone-event/8000
    
    a=fmtp:101 0-15
    
    *****************************  SIP message buffer end ******************************
    It also uses the Call-Info header, which is once again removed by the Quadro.

    I suspect this is also why I can't get a directed page betwen two Snom phones to work.




  2. #2

    Default

    Just checked how the Snom does it.. and they use the same header

    Code:
    15:43:49 Receive SIP message # (14/10/2007 13:43:49:417 GMT) # UDP #
    1469 bytes # from: 192.168.1.104:1024 # to: 192.168.1.10:5060
    
    
    
    ***************************** SIP message buffer start *****************************
    
    INVITE sip:53@192.168.1.10;user=phone;intercom=true SIP/2.0
    
    Via: SIP/2.0/UDP 192.168.1.104:1025;branch=z9hG4bK-ra1pfdgk0n8z;rport
    
    From: "Snom Epygi" <sip:52@192.168.1.10:5060>;tag=04a1jw3k7b
    
    To: <sip:53@192.168.1.10;user=phone;intercom=true>
    
    Call-ID: 3c26a45c29ca-3sh4xxp3rs5i
    
    CSeq: 2 INVITE
    
    Max-Forwards: 70
    
    Contact: <sip:52@192.168.1.104:1025>;flow-id=1
    
    P-Key-Flags: resolution="31x13", keys="4"
    
    User-Agent: snom370/7.1.24
    
    Accept: application/sdp
    
    Alert-Info:  <http://www.notused.com>;info=alert-autoanswer;delay=0 
    
    Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
    
    Allow-Events: talk, hold, refer, call-info
    
    Supported: timer, 100rel, replaces, callerid
    
    Session-Expires: 3600;refresher=uas
    
    Min-SE: 90
    
    Authorization:  Digest
    username="52",realm="quadro.epygi-config.com",nonce= "63db787bcbde579555ca1b6033f37caa",uri="sip:53@192.168.1.10; user=phone;intercom=true",response="592eedb659c63a22a46a6725 a1871f87",opaque="1192369429",algorithm=MD5
     
    Content-Type: application/sdp
    
    Content-Length: 419
    
    
    
    v=0
    
    o=root 861886447 861886447 IN IP4 192.168.1.104
    
    s=call
    
    c=IN IP4 192.168.1.104
    
    t=0 0
    
    m=audio 52384 RTP/AVP 0 8 9 2 3 18 4
    
    a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:5mdI7//WuV3UovSV6SdqlH8r6C1+qu7rn181UdZz
    
    a=rtpmap:0 pcmu/8000
    
    a=rtpmap:8 pcma/8000
    
    a=rtpmap:9 g722/8000
    
    a=rtpmap:2 g726-32/8000
    
    a=rtpmap:3 gsm/8000
    
    a=rtpmap:18 g729/8000
    
    a=rtpmap:4 g723/8000
    
    a=ptime:20
    
    a=encryption:optional
    
    a=sendrecv
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:43:49 Try to send SIP message # (14/10/2007 13:43:49:422 GMT) #
    UDP # 283 bytes # from: 192.168.1.10:5060 # to: 192.168.1.104:1024
    
    
    
    ***************************** SIP message buffer start *****************************
    
    SIP/2.0 100 Trying
    
    Via: SIP/2.0/UDP 192.168.1.104:1025;rport=1024;branch=z9hG4bK-ra1pfdgk0n8z
    
    To: <sip:53@192.168.1.10;user=phone;intercom=true>
    
    From: "Snom Epygi" <sip:52@192.168.1.10:5060>;tag=04a1jw3k7b
    
    CSeq: 2 INVITE
    
    Call-ID: 3c26a45c29ca-3sh4xxp3rs5i
    
    Content-Length: 0
    
    
    
    *****************************  SIP message buffer end ******************************
    
    
    
    15:43:49 TLayer::MsgToTU # Msg type: 51 # TID: 7648 # DID: 0
    
    15:43:49 UACore::TLReqMsgProc # Got INVITE SIP request
    
    15:43:49 SipSessDlg: # Call ID Info # SID: 5121187702305621171 # 15:43:49 SipID: 3c26a45c29ca-3sh4xxp3rs5i
    
    15:43:49 SipSessDlg::ChangeSessTimerFlag # user 52 registered on UA # SessionTimer = 0
    
    15:43:49 SipSessDlg::ChangeSessTimerFlag # user 52 # SessionTimer = 0
    
    15:43:49 SipSessDlg::IncInviteProc # Got INVITE message # OID: 7649 # SID: 5121187702305621171
    
    15:43:49 TargetQualifier::QualifyTargets # Try to qualify targets for dest 192.168.1.104 # OID: 7649
    
    15:43:49 TargetQualifier::NatHandling # Don't need nat settings - direct reach # OID: 7649
    
    15:43:49 UA --> CM # MakeCall # from: 52@192.168.1.10:5060, to:
    53, child: (empty), media exist, GUID: (empty) # ContactInfo:
    192.168.1.104 # SID: 5121187702305621171
    
    15:43:49 CallsAgent::OnMakeCall # MakeCall from CM # SID: 5098525612925036
    
    15:43:49 CallsAgent::OnMakeCall # Empty GUID from CM # SID: 5098525612925036
    
    15:43:49 SipSessDlg: # Call ID Info # SID: 5098525612925036 #
    15:43:49  SipID:
    dc669a33-70a8-4424-969f-79ba5492bdeb@quadro.epygi-co nfig.com
    
    15:43:49 SipSessDlg::ChangeSessTimerFlag # user 53 registered on UA # SessionTimer = 1
    
    15:43:49 SipSessDlg::ChangeSessTimerFlag # user 53 # SessionTimer = 1
    
    15:43:49 CM --> UA # MakeCall # from: 52, to:
    53@192.168.1.146:2048, media exist # replaceSID: 0 # privacy: 20000 #
    AlertInfo: not exist # AddInfo: [SP#192.168.1.10:5060][BND#53] #OID:
    7650 # SID: 5098525612925036
    
    15:43:49 SipDlg::DetermineInitTarget # Call to primary server # OID: 7650
    
    15:43:49 TargetQualifier::QualifyTargets # Try to qualify targets for dest 192.168.1.146 # OID: 7650
    
    15:43:49 TargetQualifier::NatHandling # Don't need nat settings - direct reach # OID: 7650
    
    15:43:49 SipSessDlg::SelfMediaHandling # State: local offer, Peer
    Public Key: empty, SRTP flag: false, Sesssion Key: empty # OID: 7650 #
    SID: 5098525612925036
    
    15:43:49 Try to send SIP message # (14/10/2007 13:43:49:668 GMT) #
    UDP # 878 bytes # from: 192.168.1.10:5060 # to: 192.168.1.146:2048
    
    
    
    ***************************** SIP message buffer start *****************************
    
    INVITE sip:53@192.168.1.146:2048 SIP/2.0
    
    Via: SIP/2.0/UDP  192.168.1.10:5060;rport;branch=z9hG4bKEPSVBUS76ee2952-25ce-4 0b9-a53b-f13785a36eec
    
    To: <sip:53@192.168.1.10:5060>
    
    From: "Stephan Snom"  <sip:52@192.168.1.10>;tag=11923556970aa3b73e-b6aa-44be -8dda-654892b94ff4
    
    CSeq: 847 INVITE
    
    Call-ID:  dc669a33-70a8-4424-969f-79ba5492bdeb...ygi-config.com 
    
    Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, UPDATE
    
    Contact: "Stephan Snom" <sip:52@192.168.1.10:5060>
    
    Content-Type: application/sdp
    
    Supported: replaces, norefersub, timer
    
    User-Agent: Epygi Quadro SIP User Agent/v4.1.33 (QUADRO-2X)
    
    Min-SE: 20
    
    Session-Expires: 180
    
    Max-Forwards: 70
    
    Content-Length: 170
    
    
    
    v=0
    
    o=52 130 583 IN IP4 192.168.1.10
    
    s=-
    
    c=IN IP4 192.168.1.10
    
    t=0 0
    
    m=audio 6054 RTP/AVP 0 8 18
    
    a=rtpmap:0 PCMU/8000
    
    a=rtpmap:8 PCMA/8000
    
    a=rtpmap:18 G729/8000
    
    *****************************  SIP message buffer end ******************************
    Since it is removed, it's no surprise that directed intercom doesn't work on the snom either.



  3. #3

    Default

    This will be passed to the Epygi development, to see if there is a real contradiction between RFCs and Quadro behaviour. As to direct paging between Snom phones, we do not guarantee the correctfunctionalityof the phones' features, activated not by Quadro, but from the phones itself. That can bring to the phones' malfumctionand it is documented on all our documents.

  4. #4

    Default

    Of course what I said about Snom was incorrect.. they use Alert-Info. However, the situation is the same.. the RFC demands that this header be left alone (it's on page 161 of the RFC).

    we do not guarantee the correctfunctionalityof the phones' features, activated not by Quadro
    Then again, you cannot guarantee it either by doing it from the proxy, unless you inspect requests, and match the phone type to a certain behavior.. your current paging doesn't work for Grandstream and Aastra phones because the Grandstreams don't like the uri in the Call-Info header, and because Aastra uses the Alert-Info header in a special way

  5. #5

    Default



    We support paging for the Epygi Supported phones only and that are the following phones (just look at Paging Group Online Help):


    SNOM 320 (SIP phone)


    SNOM 360 (SIP phone)


    Aastra 480i (SIP phone)


    Aastra 9133i (SIP phone)


    Aastra 9112i (SIP phone)


    Aastra 480e (analog phone)


    Also the phones of 5x seriesare included for the next release. No need to do some experiments, to find out the information that are already provided in our Help. As you can see there is no Grandsream phones in this list, as that phones aren't Supported phones, but just Tested phones. Some functionality may not work, or work in other way on that phones. And we have no problem with paging of Aastra phones. Maybe you have some beta-version on your Aastra, we don't know.I will suggest you again to read all the documentation regarding IP-Phones configuration and usage from our "Downloads" section before using IP-Phones with Quadro..

  6. #6

    Default

    Also the phones of 5x seriesare included for the next release.
    I take it that's the release beyond 4.1.33 which explains why it doesn't work on my Aastra i57 (running the officially latest firmware 2.1.0.2145)

    The latest document on Quadro features on supported phones (QuadroFeaturesonEpygiSupportedIPPhonesList-Rev1.1-MedRes.pd f) does not mention anything about paging and Aastra's 5 series (I did check before posting) - I did not check the online help though. I also know to use the PBX features with preference, but you cannot blame me for looking at phone specific features when the PBX provided feature does not work / doesn't really get me where I want to go. Plus in this case, it turns out it's not so much a feature limitation than the proxy touching header fields it's not supposed to touch so experimenting did help locate a problem in the software - and I already did the failure analysis for you including checking the RFC. I don't think that's a negative.


  7. #7

    Default



    Sorry, if you understand my comment in a wrong way We do not blame anyone. More, you are free to do anything with the phones, we just warn you, that the manual configuration is not desired and it will bring some negative effects. The problems connected to wrong configuration or usage of non-recommended featuresare out of our scope, because we support not the phone's features, but the Quadro features on the phones. Thank you for such interest to our products - we value all the feedbacks from our customers, that helps as to improve our product.


    BTW, here in our testlab paging with Aastra5xi series works OK.

  8. #8

    Default

    @aramk: will the header removal be fixed in a future release? If not, I'd like to hear the argument why this isn't an RFC violation.

  9. #9

    Default

    I'll pass this to our SIP developers. They'll answer to all your questions.

  10. #10

    Default

    Stephan, sorry for asking Did you mean SIP rfc-3162, as it is written in your first comment, or you mean rfc-3261 ? Here is all the information that I've found about Call-Info header in SIP RFC.

    rfc-3261

    "20.9 Call-Info
    The Call-Info header field provides additional information about the
    caller or callee, depending on whether it is found in a request or
    response. The purpose of the URI is described by the "purpose"
    parameter. The "icon" parameter designates an image suitable as an
    iconic representation of the caller or callee. The "info" parameter
    describes the caller or callee in general, for example, through a web
    page. The "card" parameter provides a business card, for example, in
    vCard [36] or LDIF [37] formats. Additional tokens can be registered
    using IANA and the procedures in Section 27.
    Use of the Call-Info header field can pose a security risk. If a
    callee fetches the URIs provided by a malicious caller, the callee
    may be at risk for displaying inappropriate or offensive content,
    dangerous or illegal content, and so on. Therefore, it is
    RECOMMENDED that a UA only render the information in the Call-Info
    header field if it can verify the authenticity of the element that
    originated the header field and trusts that element. This need not
    be the peer UA; a proxy can insert this header field into requests.
    Example:
    Call-Info: <http://wwww.example.com/alice/photo.jpg> ;purpose=icon,
    <http://www.example.com/alice/> ;purpose=info"

    Here are 2 arguments to not use Call-Info header.
    1. You can read in text above, that "Use of the Call-Info header field can pose a security risk", and
    2. My understanding of the sentence from your first comment "a proxy may not remove a Call-Info header" is that we may or may not remove Call-Info header, as it is not a mandatory field.
    so I don't think that Call-Info header removal is a RFC violation.

    Anyway, we support Call-Info header in Quadro 5.0.x versions for SLA functionality only.

    I already passed your question to our development, maybe they'll have some additions.
    Last edited by aramk; 02-11-2008 at 07:03 AM.

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