<br><br><div class="gmail_quote">2009/4/24 Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span><br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="im">Olivier wrote:<br>
<br>
> When a SIP hardphone is transfering a call while ringing (caller and<br>
> callee don't speak to each other) using phone's Transfer key, it seems<br>
> BLINDTRANSFER remains empty.<br>
> Though I can see a 302 MOVED TEMPORARILY message coming in.<br>
<br>
</div>If the person performing the transfer has dialed the transferee's number<br>
and hears the call ringing, that is not a blind transfer, it is an<br>
attended to transfer to a call that hasn't been answered yet. There<br>
won't be any variables set for blind transfer, as it isn't one.</blockquote><div><br><br>Here is an extract from SIP debug (7530 is transferring incoming call from 7533 to 7531) :<br><br>osiris2*CLI><br><--- Transmitting (no NAT) to <a href="http://192.168.100.122:5060">192.168.100.122:5060</a> ---><br>
SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 192.168.100.122:5060;branch=z9hG4bK4697915359658203609-1269236;received=192.168.100.122<br>From: "Alain"<sip:7533@192.168.100.254:5060;user=phone>;tag=c0a80101-135de8<br>
To: <sip:7530@192.168.100.254:5060;user=phone>;tag=as37f823b2<br>Call-ID: <a href="mailto:364221-c0a80101-0-4@192.168.100.122">364221-c0a80101-0-4@192.168.100.122</a><br>CSeq: 2 INVITE<br>Server: Asterisk PBX 1.6.1.0-rc4<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>Contact: <<a href="mailto:sip%3A7530@192.168.100.254">sip:7530@192.168.100.254</a>><br>Content-Length: 0<br><br><br>
<br><br><-------------><br>osiris2*CLI><br><--- SIP read from UDP://<a href="http://192.168.100.123:5060">192.168.100.123:5060</a> ---><br>SIP/2.0 302 Moved Temporarily<br>Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK47ed73d6;rport<br>
From: "Alain"<<a href="mailto:sip%3A7533@192.168.100.254">sip:7533@192.168.100.254</a>>;tag=as2d189259<br>To: <sip:7530@192.168.100.123:5060;user=phone>;tag=c0a80101-135999<br>Call-ID: <a href="mailto:1ba5b4c707b15fec0909665f6e9ea2d7@192.168.100.254">1ba5b4c707b15fec0909665f6e9ea2d7@192.168.100.254</a><br>
CSeq: 102 INVITE<br>Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO<br>Contact: <sip:7531@192.168.100.254:5060;user=phone><br>Content-Length: 0<br><br><br><-------------><br>
--- (9 headers 0 lines) ---<br>Transmitting (no NAT) to <a href="http://192.168.100.123:5060">192.168.100.123:5060</a>:<br>ACK sip:7530@192.168.100.123:5060;user=phone SIP/2.0<br>Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK47ed73d6;rport<br>
Max-Forwards: 70<br>From: "Alain" <<a href="mailto:sip%3A7533@192.168.100.254">sip:7533@192.168.100.254</a>>;tag=as2d189259<br>To: <sip:7530@192.168.100.123:5060;user=phone>;tag=c0a80101-135999<br>Contact: <<a href="mailto:sip%3A7533@192.168.100.254">sip:7533@192.168.100.254</a>><br>
Call-ID: <a href="mailto:1ba5b4c707b15fec0909665f6e9ea2d7@192.168.100.254">1ba5b4c707b15fec0909665f6e9ea2d7@192.168.100.254</a><br>CSeq: 102 ACK<br>User-Agent: Asterisk PBX 1.6.1.0-rc4<br>Content-Length: 0<br><br><br>---<br>
Really destroying SIP dialog '<a href="mailto:1ba5b4c707b15fec0909665f6e9ea2d7@192.168.100.254">1ba5b4c707b15fec0909665f6e9ea2d7@192.168.100.254</a>' Method: INVITE<br>Audio is at 192.168.100.254 port 13840<br>Adding codec 0x8 (alaw) to SDP<br>
Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (no NAT) to <a href="http://192.168.100.88:29462">192.168.100.88:29462</a>:<br>INVITE sip:7531@192.168.100.88:29462;rinstance=160ae873c74c4480 SIP/2.0<br>
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK1f07ec53;rport<br>Max-Forwards: 70<br>From: "Alain" <<a href="mailto:sip%3A7533@192.168.100.254">sip:7533@192.168.100.254</a>>;tag=as0104afde<br>To: <sip:7531@192.168.100.88:29462;rinstance=160ae873c74c4480><br>
Contact: <<a href="mailto:sip%3A7533@192.168.100.254">sip:7533@192.168.100.254</a>><br>Call-ID: <a href="mailto:54b4a9fe0fbf10a51da9c2f3010614a6@192.168.100.254">54b4a9fe0fbf10a51da9c2f3010614a6@192.168.100.254</a><br>
CSeq: 102 INVITE<br>User-Agent: Asterisk PBX 1.6.1.0-rc4<br>Date: Fri, 24 Apr 2009 05:43:54 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>Content-Type: application/sdp<br>
Content-Length: 271<br><br>v=0<br>o=root 525634823 525634823 IN IP4 192.168.100.254<br>s=Asterisk PBX 1.6.1.0-rc4<br>c=IN IP4 192.168.100.254<br>t=0 0<br>m=audio 13840 RTP/AVP 8 101<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>---<br>osiris2*CLI><br><--- SIP read from UDP://<a href="http://192.168.100.88:29462">192.168.100.88:29462</a> ---><br>SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK1f07ec53;rport=5060<br>To: <sip:7531@192.168.100.88:29462;rinstance=160ae873c74c4480><br>From: "Alain" <<a href="mailto:sip%3A7533@192.168.100.254">sip:7533@192.168.100.254</a>>;tag=as0104afde<br>
Call-ID: <a href="mailto:54b4a9fe0fbf10a51da9c2f3010614a6@192.168.100.254">54b4a9fe0fbf10a51da9c2f3010614a6@192.168.100.254</a><br>CSeq: 102 INVITE<br>Content-Length: 0<br><br><br><br><br>So when receiving 302 Moved Temporarily, Asterisk (version 1.6.1-rc4) is issuing a new INVITE and doesn't set any BLINDTRANSFER variable.<br>
Thinking back about that, I would say it should have done so.<br><br>Your opinion ?<br>Would you classify that as an attended transfer ?<br><br><br> </div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br>
<br>
--<br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
skype: kpfleming | jabber: <a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a><br>
Check us out at <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
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</blockquote></div><br>