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Transferring calls from voice response application through Asterisk RRS feed

  • Question

  • I've managed to have the following setup:
    VoIP provider <=> Asterisk 1.6.1 Beta 1 <=> Speech Server 2007

    When I try to do a blind transfer:
    Code Snippet
    blindTransfer.CalledParty = new SipUriTelephonyAddress(
      new SipUri(SipUriScheme.Sip, "202", "192.168.10.60", 0, "user=phone"));

     

    the call simply hangs up. Is there a way to configure Asterisk dialplan to transfer the call.

    This is the configuration for the incoming calls:
    Code Snippet
    [incoming-calls]
    exten => PHONENUMBER,1,SIPAddHeader(Diversion: \;reason=no-answer\;screen=no\;privacy=off)
    exten => PHONENUMBER,n,Dial(SIP/222@192.168.10.118)
    exten => PHONENUMBER,n,Hangup()


    This is what Wireshark shows:

    Code Snippet
    No.  Time            Source                Destination           Protocol    Info
    56   13:17:45.620505 192.168.10.60         192.168.10.118        TCP         60943 > ca-1 [SYN] Seq=0 Win=5840 Len=0 MSS=1460 TSV=74989039 TSER=0 WS=6
    57   13:17:45.620550 192.168.10.118        192.168.10.60         TCP         ca-1 > 60943 [SYN, ACK] Seq=0 Ack=1 Win=16384 Len=0 MSS=1460 WS=0 TSV=0 TSER=0
    58   13:17:45.620778 192.168.10.60         192.168.10.118        TCP         60943 > ca-1 [ACK] Seq=1 Ack=1 Win=5888 Len=0 TSV=74989041 TSER=0
    59   13:17:45.621248 192.168.10.60         192.168.10.118        SIP/SDP     Request: INVITE sip:222@192.168.10.118:5064, with session description
    60   13:17:45.623507 192.168.10.118        192.168.10.60         SIP         Status: 100 Trying
    61   13:17:45.624099 192.168.10.60         192.168.10.118        TCP         60943 > ca-1 [ACK] Seq=970 Ack=285 Win=6912 Len=0 TSV=74989050 TSER=770271
    63   13:17:45.626696 192.168.10.118        192.168.10.60         SIP         Status: 180 Ringing
    64   13:17:45.626959 192.168.10.60         192.168.10.118        TCP         60943 > ca-1 [ACK] Seq=970 Ack=623 Win=8000 Len=0 TSV=74989062 TSER=770271
    68   13:17:45.728953 192.168.10.118        192.168.10.60         SIP/SDP     Status: 200 OK, with session description
    69   13:17:45.729987 192.168.10.60         192.168.10.118        TCP         60943 > ca-1 [ACK] Seq=970 Ack=1357 Win=9472 Len=0 TSV=74989147 TSER=770272
    71   13:17:45.745730 192.168.10.60         192.168.10.118        SIP         Request: ACK sip:asidsk016.asi.local:5064;transport=Tcp;maddr=192.168.10.118
    84   13:17:45.890125 192.168.10.118        192.168.10.60         TCP         ca-1 > 60943 [ACK] Seq=1357 Ack=1441 Win=64095 Len=0 TSV=770274 TSER=74989188
    2080 13:17:55.975857 192.168.10.118        192.168.10.60         SIP         Request: REFER sip:INTERNALPHONE@192.168.10.60:5060;transport=TCP
    2081 13:17:55.976625 192.168.10.60         192.168.10.118        TCP         60943 > ca-1 [ACK] Seq=1441 Ack=1948 Win=10944 Len=0 TSV=74999419 TSER=770374
    2082 13:17:55.978454 192.168.10.60         192.168.10.118        SIP         Status: 202 Accepted
    2091 13:17:56.149282 192.168.10.118        192.168.10.60         TCP         ca-1 > 60943 [ACK] Seq=1948 Ack=1951 Win=65535 Len=0 TSV=770376 TSER=74999423
    2092 13:17:56.149435 192.168.10.60         192.168.10.118        SIP/sipfrag Request: NOTIFY sip:asidsk016.asi.local:5064;transport=Tcp;maddr=192.168.10.118, with Sipfrag(SIP/2.0 200 Ok)
    2093 13:17:56.150662 192.168.10.118        192.168.10.60         SIP         Status: 200 OK
    2094 13:17:56.191130 192.168.10.60         192.168.10.118        TCP         60943 > ca-1 [ACK] Seq=3338 Ack=2281 Win=12416 Len=0 TSV=74999635 TSER=770376
    2095 13:17:56.191169 192.168.10.118        192.168.10.60         SIP         Request: BYE sip:INTERNALPHONE@192.168.10.60:5060;transport=TCP
    2096 13:17:56.191304 192.168.10.60         192.168.10.118        TCP         60943 > ca-1 [ACK] Seq=3338 Ack=2991 Win=13888 Len=0 TSV=74999636 TSER=770376

     

    Thursday, October 16, 2008 8:56 PM

All replies

  • Is anyone familiar with this setup? I'm still trying to figure out how to handle this?
    Tuesday, November 25, 2008 3:31 AM
  • I trying to implement a configuration like this.
    But I don't have any result in the communication between Asterisk and Microsoft Speech Server 2007. Can you help with this???


    Tuesday, February 9, 2010 5:44 PM
  • Please could you tell us how is the sip configuration for a connection with MSS 2007. I really need it because i have the same structure provider>>asterisk>>MSS2007. thanks
    Tuesday, March 2, 2010 8:29 PM