[asterisk-users] BLINDTRANSFER and SIP hardphones

[Olivier]

2009/4/24 Kevin P. Fleming <kpfleming at digium.com>

> 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:7533 at 192.168.100.254:5060;user=phone>;tag=c0a80101-135de8
To: <sip:7530 at 192.168.100.254:5060;user=phone>;tag=as37f823b2
Call-ID: 364221-c0a80101-0-4 at 192.168.100.122
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:7530 at 192.168.100.254 <sip%3A7530 at 192.168.100.254>>
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:7533 at 192.168.100.254 <sip%3A7533 at 192.168.100.254>
>;tag=as2d189259
To: <sip:7530 at 192.168.100.123:5060;user=phone>;tag=c0a80101-135999
Call-ID: 1ba5b4c707b15fec0909665f6e9ea2d7 at 192.168.100.254
CSeq: 102 INVITE
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:7531 at 192.168.100.254:5060;user=phone>
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 192.168.100.123:5060:
ACK sip:7530 at 192.168.100.123:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK47ed73d6;rport
Max-Forwards: 70
From: "Alain" <sip:7533 at 192.168.100.254 <sip%3A7533 at 192.168.100.254>
>;tag=as2d189259
To: <sip:7530 at 192.168.100.123:5060;user=phone>;tag=c0a80101-135999
Contact: <sip:7533 at 192.168.100.254 <sip%3A7533 at 192.168.100.254>>
Call-ID: 1ba5b4c707b15fec0909665f6e9ea2d7 at 192.168.100.254
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.0-rc4
Content-Length: 0


---
Really destroying SIP dialog '
1ba5b4c707b15fec0909665f6e9ea2d7 at 192.168.100.254' 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:7531 at 192.168.100.88:29462;rinstance=160ae873c74c4480 SIP/2.0
Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK1f07ec53;rport
Max-Forwards: 70
From: "Alain" <sip:7533 at 192.168.100.254 <sip%3A7533 at 192.168.100.254>
>;tag=as0104afde
To: <sip:7531 at 192.168.100.88:29462;rinstance=160ae873c74c4480>
Contact: <sip:7533 at 192.168.100.254 <sip%3A7533 at 192.168.100.254>>
Call-ID: 54b4a9fe0fbf10a51da9c2f3010614a6 at 192.168.100.254
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:7531 at 192.168.100.88:29462;rinstance=160ae873c74c4480>
From: "Alain" <sip:7533 at 192.168.100.254 <sip%3A7533 at 192.168.100.254>
>;tag=as0104afde
Call-ID: 54b4a9fe0fbf10a51da9c2f3010614a6 at 192.168.100.254
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: kpfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
>
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