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dlynton

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Posts posted by dlynton

  1. Hold music not playing on MSS

    in Microsoft Speech Server

    Posted

    You mean the PBX plays noise after receiving the 302 from the speech server? That sounds perfectly okay to me...? I can't imagine that a redirect would cause music on hold even in older versions.

     

    It used to always play music on hold. This is a supervised transfer, where the caller hears hold music while the server dials a new call, waits for confirmation from the called party, then connects the two parties. so it's not strictly a redirect. i was a little surprised that we were getting pbxnsip music on hold from a MSS supervised transfer, but that is how it used to work.

  2. Hold music not playing on MSS

    in Microsoft Speech Server

    Posted

    After upgrading to the latest version of PBXNSIP, we are no longer getting hold music when we do a supervised transfer. As you can see in the call log below, pbxnsip is not attempting to play moh.wav, but it does say "call hold from trunk". We haven't changed anything in our application, just upgraded to the latest version of pbxnsip. Any ideas?

     

     

    REGISTER sip:callcentric.com SIP/2.0

    Via: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-2fbc56286f9c2e41e5b5a6024332fdf6;rport

    From: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990

    To: "CallCentricSanta" <sip:17772557741@callcentric.com>

    Call-ID: 9enn0ykw@pbx

    CSeq: 27448 REGISTER

    Max-Forwards: 70

    Contact: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>"

    User-Agent: pbxnsip-PBX/2.1.6.2448

    Expires: 3600

    Content-Length: 0

     

     

    [9] 2008/03/05 11:31:17: SIP Rx udp:204.11.192.23:5080:

    SIP/2.0 200 Ok

    v: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-2fbc56286f9c2e41e5b5a6024332fdf6;rport=5060

    f: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990

    t: "CallCentricSanta" <sip:17772557741@callcentric.com>

    i: 9enn0ykw@pbx

    CSeq: 27448 REGISTER

    m: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>";expires=97

    l: 0

     

     

    [8] 2008/03/05 11:31:33: Found interface on 192.168.54.103 with netmask 255.255.255.0

    [8] 2008/03/05 11:31:33: Found interface on 127.0.0.1 with netmask 255.0.0.0

    [9] 2008/03/05 11:32:05: Resolve 163: url sip:callcentric.com

    [9] 2008/03/05 11:32:05: Resolve 163: naptr callcentric.com

    [9] 2008/03/05 11:32:05: Resolve 163: srv tls _sips._tcp.callcentric.com

    [9] 2008/03/05 11:32:05: Resolve 163: srv tcp _sip._tcp.callcentric.com

    [9] 2008/03/05 11:32:05: Resolve 163: srv udp _sip._udp.callcentric.com

    [9] 2008/03/05 11:32:05: Resolve 163: a udp alpha1.callcentric.com 5080

    [9] 2008/03/05 11:32:05: Resolve 163: udp 204.11.192.22 5080

    [9] 2008/03/05 11:32:05: Resolve 163: a udp alpha2.callcentric.com 5080

    [9] 2008/03/05 11:32:05: Resolve 163: udp 204.11.192.23 5080

    [9] 2008/03/05 11:32:05: Resolve 163: a udp callcentric.com 5060

    [9] 2008/03/05 11:32:05: Resolve 163: udp 204.11.192.23 5060

    [8] 2008/03/05 11:32:05: Trunk 6 (callcentric) has outbound proxy udp:204.11.192.22:5080 udp:204.11.192.23:5060 udp:204.11.192.23:5080

    [9] 2008/03/05 11:32:05: Resolve 164: udp 204.11.192.23 5080

    [9] 2008/03/05 11:32:05: SIP Tx udp:204.11.192.23:5080:

    REGISTER sip:callcentric.com SIP/2.0

    Via: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-5c7846da9d261398d53f498c45f07a0e;rport

    From: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990

    To: "CallCentricSanta" <sip:17772557741@callcentric.com>

    Call-ID: 9enn0ykw@pbx

    CSeq: 27450 REGISTER

    Max-Forwards: 70

    Contact: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>"

    User-Agent: pbxnsip-PBX/2.1.6.2448

    Expires: 3600

    Content-Length: 0

     

     

    [9] 2008/03/05 11:32:05: SIP Rx udp:204.11.192.23:5080:

    SIP/2.0 200 Ok

    v: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-5c7846da9d261398d53f498c45f07a0e;rport=5060

    f: "CallCentricSanta" <sip:17772557741@callcentric.com>;tag=41990

    t: "CallCentricSanta" <sip:17772557741@callcentric.com>

    i: 9enn0ykw@pbx

    CSeq: 27450 REGISTER

    m: <sip:17772557741@192.168.54.103:5060;transport=udp;line=1679091c>;+sip.instance="<urn:uuid:63e9e564-771f-4789-9477-0fa6b0f68aa7>";expires=98

    l: 0

     

     

    [9] 2008/03/05 11:32:12: SIP Rx udp:192.168.54.100:5060:

    INVITE sip:7135599056@192.168.54.103 SIP/2.0

    Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport

    From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

    To: <sip:7135599056@192.168.54.103>

    Contact: <sip:2814610506@192.168.54.100>

    Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

    CSeq: 102 INVITE

    User-Agent: Asterisk PBX

    Max-Forwards: 70

    Date: Wed, 05 Mar 2008 17:49:53 GMT

    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

    Content-Type: application/sdp

    Content-Length: 265

     

    v=0

    o=root 5779 5779 IN IP4 192.168.54.100

    s=session

    c=IN IP4 192.168.54.100

    t=0 0

    m=audio 10858 RTP/AVP 3 0 8 101

    a=rtpmap:3 GSM/8000

    a=rtpmap:0 PCMU/8000

    a=rtpmap:8 PCMA/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=silenceSupp:off - - - -

     

    [7] 2008/03/05 11:32:12: UDP: Opening socket on port 56516

    [7] 2008/03/05 11:32:12: UDP: Opening socket on port 56517

    [5] 2008/03/05 11:32:12: Identify trunk (IP address/port and domain match) 5

    [9] 2008/03/05 11:32:12: Resolve 165: aaaa udp 192.168.54.100 5060

    [9] 2008/03/05 11:32:12: Resolve 165: a udp 192.168.54.100 5060

    [9] 2008/03/05 11:32:12: Resolve 165: udp 192.168.54.100 5060

    [9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.100:5060:

    SIP/2.0 100 Trying

    Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport=5060

    From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

    To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c

    Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

    CSeq: 102 INVITE

    Content-Length: 0

     

     

    [6] 2008/03/05 11:32:12: Sending RTP for 1a92846a480027bf329359bc5a096045@192.168.54.100#00b1880f9c to 192.168.54.100:10858

    [5] 2008/03/05 11:32:12: Trunk asterisk outbound sends call to 7135599056

    [7] 2008/03/05 11:32:12: Calling extension 7135599056

    [6] 2008/03/05 11:32:12: Redirecting to external voicemail account 7135599056 destination sip:011997135599056@192.168.54.103

    [9] 2008/03/05 11:32:12: Dialplan: Evaluating !^(7132234676)@.*!sip:\1@\r;user=phone!i against 011997135599056@192.168.54.103

    [9] 2008/03/05 11:32:12: Dialplan: Evaluating !^01199([0-9]*)@.*!sip:\1@\r;user=phone!i against 011997135599056@192.168.54.103

    [5] 2008/03/05 11:32:12: Dialplan default: Match 011997135599056@192.168.54.103 to <sip:7135599056@192.168.54.200;user=phone> on trunk MS Speech Server

    [5] 2008/03/05 11:32:12: Using "DELESANDRI T " <sip:2814610506@192.168.54.103> as redirect from

    [5] 2008/03/05 11:32:12: Charge user 7135599056 for redirecting calls

    [8] 2008/03/05 11:32:12: Play audio_moh/noise.wav

    [7] 2008/03/05 11:32:12: UDP: Opening socket on port 58934

    [7] 2008/03/05 11:32:12: UDP: Opening socket on port 58935

    [9] 2008/03/05 11:32:12: Resolve 166: url sip:192.168.54.200:5060;transport=tcp

    [9] 2008/03/05 11:32:12: Resolve 166: a tcp 192.168.54.200 5060

    [9] 2008/03/05 11:32:12: Resolve 166: tcp 192.168.54.200 5060

    [9] 2008/03/05 11:32:12: SIP Tx tcp:192.168.54.200:5060:

    INVITE sip:7135599056@192.168.54.200;user=phone SIP/2.0

    Via: SIP/2.0/TCP 192.168.54.103:2335;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport

    From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140

    To: <sip:7135599056@192.168.54.200;user=phone>

    Call-ID: f89e1cc4@pbx

    CSeq: 883 INVITE

    Max-Forwards: 70

    Contact: <sip:2814610506@192.168.54.103:2335;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2448

    Diversion: <tel:7135599056>;reason=unavailable;screen=no;privacy=off

    P-Preferred-Identity: <sip:7135599056@192.168.54.103>

    Content-Type: application/sdp

    Content-Length: 220

     

    v=0

    o=- 35388 35388 IN IP4 192.168.54.103

    s=-

    c=IN IP4 192.168.54.103

    t=0 0

    m=audio 58934 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=sendrecv

     

    [9] 2008/03/05 11:32:12: Resolve 167: aaaa udp 192.168.54.100 5060

    [9] 2008/03/05 11:32:12: Resolve 167: a udp 192.168.54.100 5060

    [9] 2008/03/05 11:32:12: Resolve 167: udp 192.168.54.100 5060

    [9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.100:5060:

    SIP/2.0 183 Ringing

    Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport=5060

    From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

    To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c

    Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

    CSeq: 102 INVITE

    Contact: <sip:7135599056@192.168.54.103:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2448

    Content-Type: application/sdp

    Content-Length: 220

     

    v=0

    o=- 63166 63166 IN IP4 192.168.54.103

    s=-

    c=IN IP4 192.168.54.103

    t=0 0

    m=audio 56516 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=sendrecv

     

    [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:5060:

    SIP/2.0 100 Trying

    FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

    TO: <sip:7135599056@192.168.54.200;user=phone>

    CSEQ: 883 INVITE

    CALL-ID: f89e1cc4@pbx

    VIA: SIP/2.0/TCP 192.168.54.103:2335;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport

    CONTENT-LENGTH: 0

     

     

    [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:5060:

    SIP/2.0 302 Moved Temporarily

    FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

    TO: <sip:7135599056@192.168.54.200;user=phone>;tag=f4d3cd3c6c

    CSEQ: 883 INVITE

    CALL-ID: f89e1cc4@pbx

    VIA: SIP/2.0/TCP 192.168.54.103:2335;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport

    CONTACT: <sip:7135599056@192.168.54.200:2257;user=phone;transport=Tcp;maddr=192.168.54.200;x-mss-call-id=f89e1cc4%40pbx>

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [7] 2008/03/05 11:32:12: Call f89e1cc4@pbx#5140: Clear last INVITE

    [9] 2008/03/05 11:32:12: Resolve 168: url sip:7135599056@192.168.54.200;user=phone

    [9] 2008/03/05 11:32:12: Resolve 168: udp 192.168.54.200 5060

    [9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.200:5060:

    ACK sip:7135599056@192.168.54.200;user=phone SIP/2.0

    Via: SIP/2.0/UDP 192.168.54.103:5060;branch=z9hG4bK-1bdb15e9e72d3681cef849f25fc0b0c5;rport

    From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140

    To: <sip:7135599056@192.168.54.200;user=phone>;tag=f4d3cd3c6c

    Call-ID: f89e1cc4@pbx

    CSeq: 883 ACK

    Max-Forwards: 70

    Contact: <sip:2814610506@192.168.54.103:5060;transport=udp>

    P-Preferred-Identity: <sip:7135599056@192.168.54.103>

    Content-Length: 0

     

     

    [5] 2008/03/05 11:32:12: Redirecting call

    [9] 2008/03/05 11:32:12: Resolve 169: aaaa tcp 192.168.54.200 2257

    [9] 2008/03/05 11:32:12: Resolve 169: a tcp 192.168.54.200 2257

    [9] 2008/03/05 11:32:12: Resolve 169: tcp 192.168.54.200 2257

    [9] 2008/03/05 11:32:12: SIP Tx tcp:192.168.54.200:2257:

    INVITE sip:7135599056@192.168.54.200:2257;user=phone;transport=Tcp;maddr=192.168.54.200;x-mss-call-id=f89e1cc4%40pbx SIP/2.0

    Via: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport

    From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140

    To: <sip:7135599056@192.168.54.200;user=phone>

    Call-ID: f89e1cc4@pbx

    CSeq: 884 INVITE

    Max-Forwards: 70

    Contact: <sip:2814610506@192.168.54.103:2336;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2448

    Diversion: <tel:7135599056>;reason=unavailable;screen=no;privacy=off

    P-Preferred-Identity: <sip:7135599056@192.168.54.103>

    Content-Type: application/sdp

    Content-Length: 220

     

    v=0

    o=- 35388 35388 IN IP4 192.168.54.103

    s=-

    c=IN IP4 192.168.54.103

    t=0 0

    m=audio 58934 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=sendrecv

     

    [8] 2008/03/05 11:32:12: UDP: recvfrom receives ICMP message

    [5] 2008/03/05 11:32:12: Connection refused on udp:192.168.54.200:5060

    [6] 2008/03/05 11:32:12: Could not determine destination address on 168

    [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:2257:

    SIP/2.0 100 Trying

    FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

    TO: <sip:7135599056@192.168.54.200;user=phone>

    CSEQ: 884 INVITE

    CALL-ID: f89e1cc4@pbx

    VIA: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport

    CONTENT-LENGTH: 0

     

     

    [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:2257:

    SIP/2.0 180 Ringing

    FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

    TO: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

    CSEQ: 884 INVITE

    CALL-ID: f89e1cc4@pbx

    VIA: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport

    CONTENT-LENGTH: 0

    SERVER: RTCC/3.0.0.0

     

     

    [8] 2008/03/05 11:32:12: Play audio_en/ringback.wav

    [9] 2008/03/05 11:32:12: SIP Rx tcp:192.168.54.200:2257:

    SIP/2.0 200 OK

    FROM: "DELESANDRI T "<sip:2814610506@192.168.54.103>;tag=5140

    TO: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

    CSEQ: 884 INVITE

    CALL-ID: f89e1cc4@pbx

    VIA: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-758cb24853e9a80d8f297be22b1b6852;rport

    CONTACT: <sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200>;automata

    CONTENT-LENGTH: 198

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    SERVER: RTCC/3.0.0.0

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.54.200

    s=Microsoft Speech Server session

    c=IN IP4 192.168.54.200

    t=0 0

    m=audio 13440 RTP/AVP 0 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/03/05 11:32:12: Call f89e1cc4@pbx#5140: Clear last INVITE

    [7] 2008/03/05 11:32:12: Set packet length to 20

    [6] 2008/03/05 11:32:12: Sending RTP for f89e1cc4@pbx#5140 to 192.168.54.200:13440

    [9] 2008/03/05 11:32:12: Resolve 170: aaaa tcp 192.168.54.200 2257

    [9] 2008/03/05 11:32:12: Resolve 170: a tcp 192.168.54.200 2257

    [9] 2008/03/05 11:32:12: Resolve 170: tcp 192.168.54.200 2257

    [9] 2008/03/05 11:32:12: SIP Tx tcp:192.168.54.200:2257:

    ACK sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200 SIP/2.0

    Via: SIP/2.0/TCP 192.168.54.103:2336;branch=z9hG4bK-a0f9d3e52482a7033ffd2563d08a6970;rport

    From: "DELESANDRI T " <sip:2814610506@192.168.54.103>;tag=5140

    To: <sip:7135599056@192.168.54.200;user=phone>;tag=73d97a1bbf

    Call-ID: f89e1cc4@pbx

    CSeq: 884 ACK

    Max-Forwards: 70

    Contact: <sip:2814610506@192.168.54.103:2336;transport=tcp>

    P-Preferred-Identity: <sip:7135599056@192.168.54.103>

    Content-Length: 0

     

     

    [7] 2008/03/05 11:32:12: Determine pass-through mode after receiving response

    [9] 2008/03/05 11:32:12: Resolve 171: aaaa udp 192.168.54.100 5060

    [9] 2008/03/05 11:32:12: Resolve 171: a udp 192.168.54.100 5060

    [9] 2008/03/05 11:32:12: Resolve 171: udp 192.168.54.100 5060

    [9] 2008/03/05 11:32:12: SIP Tx udp:192.168.54.100:5060:

    SIP/2.0 200 Ok

    Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK708f672c;rport=5060

    From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

    To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c

    Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

    CSeq: 102 INVITE

    Contact: <sip:7135599056@192.168.54.103:5060;transport=udp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2448

    Content-Type: application/sdp

    Content-Length: 220

     

    v=0

    o=- 63166 63166 IN IP4 192.168.54.103

    s=-

    c=IN IP4 192.168.54.103

    t=0 0

    m=audio 56516 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=sendrecv

     

    [7] 2008/03/05 11:32:12: f89e1cc4@pbx#5140: RTP pass-through mode

    [7] 2008/03/05 11:32:12: 1a92846a480027bf329359bc5a096045@192.168.54.100#00b1880f9c: RTP pass-through mode

    [9] 2008/03/05 11:32:12: SIP Rx udp:192.168.54.100:5060:

    ACK sip:7135599056@192.168.54.103:5060;transport=udp SIP/2.0

    Via: SIP/2.0/UDP 192.168.54.100:5060;branch=z9hG4bK5784477d;rport

    From: "DELESANDRI T " <sip:2814610506@192.168.54.100>;tag=as70b76224

    To: <sip:7135599056@192.168.54.103>;tag=00b1880f9c

    Contact: <sip:2814610506@192.168.54.100>

    Call-ID: 1a92846a480027bf329359bc5a096045@192.168.54.100

    CSeq: 102 ACK

    User-Agent: Asterisk PBX

    Max-Forwards: 70

    Content-Length: 0

     

     

    [9] 2008/03/05 11:32:39: SIP Rx tcp:192.168.54.200:2257:

    INVITE sip:2814610506@192.168.54.103:2336;transport=tcp SIP/2.0

    FROM: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

    TO: <sip:2814610506@192.168.54.103>;tag=5140

    CSEQ: 1 INVITE

    CALL-ID: f89e1cc4@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.54.200:2257;branch=z9hG4bK2ae9f2

    CONTACT: <sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200;ms-opaque=6b1c435266dc81e2>;automata

    CONTENT-LENGTH: 210

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

     

    v=0

    o=- 0 0 IN IP4 192.168.54.200

    s=Microsoft Speech Server session

    c=IN IP4 192.168.54.200

    t=0 0

    m=audio 13440 RTP/AVP 0 8 101

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=sendonly

    a=ptime:20

     

    [7] 2008/03/05 11:32:39: Set packet length to 20

    [9] 2008/03/05 11:32:39: Resolve 172: tcp 192.168.54.200 2257

    [9] 2008/03/05 11:32:39: SIP Tx tcp:192.168.54.200:2257:

    SIP/2.0 200 Ok

    Via: SIP/2.0/TCP 192.168.54.200:2257;branch=z9hG4bK2ae9f2

    From: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

    To: <sip:2814610506@192.168.54.103>;tag=5140

    Call-ID: f89e1cc4@pbx

    CSeq: 1 INVITE

    Contact: <sip:2814610506@192.168.54.103:2336;transport=tcp>

    Supported: 100rel, replaces, norefersub

    Allow-Events: refer

    Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

    Accept: application/sdp

    User-Agent: pbxnsip-PBX/2.1.6.2448

    Content-Type: application/sdp

    Content-Length: 232

     

    v=0

    o=- 35388 35388 IN IP4 192.168.54.103

    s=-

    c=IN IP4 192.168.54.103

    t=0 0

    m=audio 58934 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=ptime:20

    a=recvonly

     

    [7] 2008/03/05 11:32:39: f89e1cc4@pbx#5140: Media-aware pass-through mode

    [6] 2008/03/05 11:32:39: Call hold from trunk

    [9] 2008/03/05 11:32:39: SIP Rx tcp:192.168.54.200:2257:

    ACK sip:2814610506@192.168.54.103:2336;transport=tcp SIP/2.0

    FROM: <sip:7135599056@192.168.54.200;user=phone>;epid=8D9F696238;tag=73d97a1bbf

    TO: <sip:2814610506@192.168.54.103>;tag=5140

    CSEQ: 1 ACK

    CALL-ID: f89e1cc4@pbx

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.54.200:2257;branch=z9hG4bKa050b1ce

    CONTENT-LENGTH: 0

    USER-AGENT: RTCC/3.0.0.0

     

     

    [5] 2008/03/05 11:32:39: SIP port accept from 192.168.54.200:2728

    [9] 2008/03/05 11:32:39: SIP Rx tcp:192.168.54.200:2728:

    INVITE sip:7132234676@192.168.54.103:5060;user=phone SIP/2.0

    FROM: <sip:2814610506@vtspeech07.dc.voicetarget.com:2257;user=phone>;epid=8D9F696238;tag=891bce9f3f

    TO: <sip:7132234676@192.168.54.103:5060;user=phone>

    CSEQ: 5 INVITE

    CALL-ID: a8066fed-b09c-4d54-9b9d-bca36ec8415c

    MAX-FORWARDS: 70

    VIA: SIP/2.0/TCP 192.168.54.200:2728;branch=z9hG4bKa3f789a

    CONTACT: <sip:vtspeech07.dc.voicetarget.com:2257;transport=Tcp;maddr=192.168.54.200;ms-opaque=6b1c435266dc81e2>;automata

    CONTENT-LENGTH: 340

    USER-AGENT: RTCC/3.0.0.0

    CONTENT-TYPE: application/sdp

    ALLOW: UPDATE

    ALLOW: Ack, Cancel, Bye,Invite,Message,Info,Service,Options,BeNotify

     

    v=0

    o=- 0 0 IN IP4 192.168.54.200

    s=Microsoft Speech Server session

    c=IN IP4 192.168.54.200

    t=0 0

    m=audio 63872 RTP/AVP 114 115 4 0 8 97 101

    a=rtpmap:114 x-msrta/16000

    a=fmtp:114 bitrate=29000

    a=rtpmap:115 x-msrta/8000

    a=fmtp:115 bitrate=11800

    a=rtpmap:97 RED/8000

    a=rtpmap:101 telephone-event/8000

    a=fmtp:101 0-16

    a=ptime:20

     

    [7] 2008/03/05 11:32:39: UDP: Opening socket on port 55368

    [7] 2008/03/05 11:32:39: UDP: Opening socket on port 55369

    [5] 2008/03/05 11:32:39: Identify trunk (IP address and domain match) 4

    [9] 2008/03/05 11:32:39: Resolve 173: tcp 192.168.54.200 2728

    [9] 2008/03/05 11:32:39: SIP Tx tcp:192.168.54.200:2728:

    SIP/2.0 100 Trying

    Via: SIP/2.0/TCP 192.168.54.200:2728;branch=z9hG4bKa3f789a

    From: <sip:2814610506@vtspeech07.dc.voicetarget.com:2257;user=phone>;epid=8D9F696238;tag=891bce9f3f

    To: <sip:7132234676@192.168.54.103:5060;user=phone>;tag=f73b3fc676

    Call-ID: a8066fed-b09c-4d54-9b9d-bca36ec8415c

    CSeq: 5 INVITE

    Content-Length: 0

  3. Supervised transfer DTMF problem

    in Microsoft Interoperability

    Posted

    Is there a codec preference or anything else I can try changing in the trunk settings? Anything else you can recommend to troublshoot this problem? I assume that a supervised transfer / conference call would typically work fine in your pbx? i.e. DTMF tones are usually heard by all parties?

     

    If you recall, I'm the one you helped set up for connectivity with Speech Server & Asterisk. Do you know anything about Asterisk that we should be checking to make sure the DTMF gets passed through? I noticed in the Asterisk call log "getdtmf on channel 41: Operation now in progress" only appeard once during the two test calls I made.

     

    It's quite frustrating to be so close, yet so far! We've come a long way to get everything working between MSS, Asterisk and PBXNSIP. It will be such a relief to finally get this behind us, so any other help you can offer is greatly appreciated!

  4. Supervised transfer DTMF problem

    in Microsoft Interoperability

    Posted

    I searched for replace in my log and found the sip header as well as pbxnsip reporting "Supported: 100rel, replaces, norefersub". I did *not* see any instance of "Replaces: Call-ID...not found". I see PBXNSIP assign a call id of ced43e0d-b8a6-46b1-bfb0-3da0b8e9d40a and then that is used later in the REPLACES= of the sip header. Do you know what the to-tag and from-tag are for? I'm attaching the log from my call (I have the log level set to 10--what can I set it to while being confident I'll get all the relevant messages?). The SIP REFER looks like it's doing everything right, and the supervised transfer works great except for the DTMF tones. Any other ideas?

    log.txt

  5. Supervised transfer DTMF problem

    in Microsoft Interoperability

    Posted

    When performing a wireshark trace there isn't any traffic when I send DTMF tones. The trace at the time of the transfer is almost identical. After this point the call should be only between the caller and the called party. I see a SIP REFER, and then a BYE response on both the supervised and blind transfer calls.

     

    Here is the only difference I've found between the blind transfer (DTMF can be heard) and the supervised transfer (DTMF tones sound like clicks).

     

    Blind transfer: REFER-TO: <sip:7132234676@192.168.54.103:5060;transport=tcp;user=phone>

     

    Supervised: REFER-TO: <sip:7132234676@192.168.54.103:5060;transport=tcp;user=phone?REPLACES=2d886047-461f-483c-b59e-179845662721%3Bto-tag%3D7b97da7e0c%3Bfrom-tag%3D264b596bbf>

     

    Do you have any idea what the REPLACES querystring is doing?

     

    Thanks for your help! We are so close to being able to deploy our new IVR application on MS Speech Server + PBXNSIP! This is our last issue after a barrage of problems.

  6. Supervised transfer DTMF problem

    in Microsoft Interoperability

    Posted

    I'm using Microsoft Speech Server to initiate a supervised transfer. After connecting the two callers, neither party can hear DTMF tones that they punch in. The tones come through sounding like a click. This doesn't happen when I do a blind transfer. Can you speculate if this is being caused by PBXNSIP and if there is a way to troubleshoot it?

     

    Thanks,

    Daniel

  7. Exchange UM Integration

    in Microsoft Exchange

    Posted

    Has anyone else gotten pbxnsip to work with Microsoft Speech Server 2007? I am trying to set it up as an extension but the pbx is not even attempting to forward the call. It just rings, rings, and eventually hangs up. Can anyone help me with this? I will be an extremely loyal customer to pbxnsip if we get this working! I haven't found anything else that will work for me. Thanks!

     

    E-mail Daniel@lyntonweb.com

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