2009/4/24 Kevin P. Fleming <[email protected]> > Olivier wrote: > > > When a SIP hardphone is transfering a call while ringing (caller and > > callee don't speak to each other) using phone's Transfer key, it seems > > BLINDTRANSFER remains empty. > > Though I can see a 302 MOVED TEMPORARILY message coming in. > > If the person performing the transfer has dialed the transferee's number > and hears the call ringing, that is not a blind transfer, it is an > attended to transfer to a call that hasn't been answered yet. There > won't be any variables set for blind transfer, as it isn't one.
Here is an extract from SIP debug (7530 is transferring incoming call from 7533 to 7531) : osiris2*CLI> <--- Transmitting (no NAT) to 192.168.100.122:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.100.122:5060 ;branch=z9hG4bK4697915359658203609-1269236;received=192.168.100.122 From: "Alain"<sip:[email protected]:5060;user=phone>;tag=c0a80101-135de8 To: <sip:[email protected]:5060;user=phone>;tag=as37f823b2 Call-ID: [email protected] CSeq: 2 INVITE Server: Asterisk PBX 1.6.1.0-rc4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: <sip:[email protected] <sip%[email protected]>> Content-Length: 0 <-------------> osiris2*CLI> <--- SIP read from UDP://192.168.100.123:5060 ---> SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK47ed73d6;rport From: "Alain"<sip:[email protected] <sip%[email protected]> >;tag=as2d189259 To: <sip:[email protected]:5060;user=phone>;tag=c0a80101-135999 Call-ID: [email protected] CSeq: 102 INVITE Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO Contact: <sip:[email protected]:5060;user=phone> Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 192.168.100.123:5060: ACK sip:[email protected]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK47ed73d6;rport Max-Forwards: 70 From: "Alain" <sip:[email protected] <sip%[email protected]> >;tag=as2d189259 To: <sip:[email protected]:5060;user=phone>;tag=c0a80101-135999 Contact: <sip:[email protected] <sip%[email protected]>> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.0-rc4 Content-Length: 0 --- Really destroying SIP dialog ' [email protected]' Method: INVITE Audio is at 192.168.100.254 port 13840 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.100.88:29462: INVITE sip:[email protected]:29462;rinstance=160ae873c74c4480 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK1f07ec53;rport Max-Forwards: 70 From: "Alain" <sip:[email protected] <sip%[email protected]> >;tag=as0104afde To: <sip:[email protected]:29462;rinstance=160ae873c74c4480> Contact: <sip:[email protected] <sip%[email protected]>> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.0-rc4 Date: Fri, 24 Apr 2009 05:43:54 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 271 v=0 o=root 525634823 525634823 IN IP4 192.168.100.254 s=Asterisk PBX 1.6.1.0-rc4 c=IN IP4 192.168.100.254 t=0 0 m=audio 13840 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- osiris2*CLI> <--- SIP read from UDP://192.168.100.88:29462 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK1f07ec53;rport=5060 To: <sip:[email protected]:29462;rinstance=160ae873c74c4480> From: "Alain" <sip:[email protected] <sip%[email protected]> >;tag=as0104afde Call-ID: [email protected] CSeq: 102 INVITE Content-Length: 0 So when receiving 302 Moved Temporarily, Asterisk (version 1.6.1-rc4) is issuing a new INVITE and doesn't set any BLINDTRANSFER variable. Thinking back about that, I would say it should have done so. Your opinion ? Would you classify that as an attended transfer ? > > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: [email protected] > Check us out at www.digium.com & www.asterisk.org > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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