[Asterisk-Dev] transfer() is not compliant with SIP RFC-3515 and RFC-3261

[lea123 at wp.pl]

Dear Developers, 

Can you please confirm that:

"The transfer() application is not compliant with RFC-3515 and 
RFC-3261 Section: 12.2.1.1"

If I issue the following in the dialplan:
   exten => 5,1,Transfer
(SIP/18005551212 at sphone.vopr.vonage.net:5061)
Asterisk SIP debug output gives:
   REFER sip:transit.vonage.net:5060 SIP/2.0

Note that the Request-URI after the keyword REFER is wrong !!!
"transit.vonage.net" is NOT the correct destination for this 
request.

Why is this happening ?


Regards,
Daniel Leeds


Asterisk SIP debug output:
-----------------------------------------------------
<-- SIP read from sphone.vopr.vonage.net:5061:
INVITE sip:pstin_7748712 at asterisk.aa.eu:5060;suppress-features=- 
SIP/2.0
Via: SIP/2.0/UDP sphone.vopr.vonage.net:5061
Via: SIP/2.0/UDP inbound9.vonage.net:5060
Via: SIP/2.0/UDP transit.vonage.net:5060;branch=z9hG4bK589
Record-Route: <sip:pstin_7748712 at sphone.vopr.vonage.net:5061>
Record-Route: <sip:pstin_7748712 at inbound9.vonage.net:5060>
From: "PSTN_7744854" 
<sip:pstn_7744854 at transit.vonage.net>;tag=1557830321
To: <sip:pstin_7748712 at inbound2.vonage.net>
Call-ID: EED802D5-834B11DA-A1CCB29D-2266172B at transit.vonage.net
CSeq: 101 INVITE
Contact: <sip:pstn_7744854 at transit.vonage.net:5060>;rtpupdated=-
Max-Forwards: 13
Content-Type: application/sdp
Content-Length:   355

v=0
o=CiscoSystemsSIP-GW-UserAgent 2082 5504 IN IP4 
transit.vonage.net
s=SIP Call
c=IN IP4 transit.vonage.net
t=0 0
m=audio 17350 RTP/AVP 0 18 2 100 101
c=IN IP4 transit.vonage.net
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


Transmitting (no NAT) to sphone.vopr.vonage.net:5061:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
sphone.vopr.vonage.net:5061;received=sphone.vopr.vonage.net
Via: SIP/2.0/UDP inbound9.vonage.net:5060
Via: SIP/2.0/UDP transit.vonage.net:5060;branch=z9hG4bK589
From: "PSTN_7744854" 
<sip:pstn_7744854 at transit.vonage.net>;tag=1557830321
To: <sip:pstin_7748712 at inbound2.vonage.net>
Call-ID: EED802D5-834B11DA-A1CCB29D-2266172B at transit.vonage.net
CSeq: 101 INVITE
User-Agent: X-PRO Vonage release 1105x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY
Max-Forwards: 70
Contact: <sip:pstin_7748712 at asterisk.aa.eu>
Content-Length: 0


Reliably Transmitting (no NAT) to sphone.vopr.vonage.net:5061:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 
sphone.vopr.vonage.net:5061;received=sphone.vopr.vonage.net
Via: SIP/2.0/UDP inbound9.vonage.net:5060
Via: SIP/2.0/UDP transit.vonage.net:5060;branch=z9hG4bK589
Record-Route: <sip:pstin_7748712 at sphone.vopr.vonage.net:5061>
Record-Route: <sip:pstin_7748712 at inbound9.vonage.net:5060>
From: "PSTN_7744854" 
<sip:pstn_7744854 at transit.vonage.net>;tag=1557830321
To: <sip:pstin_7748712 at inbound2.vonage.net>;tag=as141368b9
Call-ID: EED802D5-834B11DA-A1CCB29D-2266172B at transit.vonage.net
Seq: 101 INVITE
User-Agent: X-PRO Vonage release 1105x
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY
Max-Forwards: 70
Contact: <sip:pstin_7748712 at asterisk.aa.eu>
Content-Type: application/sdp
Content-Length: 486

v=0
o=root 3084 3084 IN IP4 asterisk.aa.eu
s=session
c=IN IP4 asterisk.aa.eu
t=0 0
m=audio 15112 RTP/AVP 4 18 3 0 8 2 5 10 7 110 97 101
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -



<-- SIP read from sphone.vopr.vonage.net:5061:
ACK sip:pstin_7748712 at asterisk.aa.eu:5060 SIP/2.0
Via: SIP/2.0/UDP sphone.vopr.vonage.net:5061
Via: SIP/2.0/UDP inbound9.vonage.net:5060
Via: SIP/2.0/UDP transit.vonage.net:5060;branch=z9hG4bK1F67
From: "PSTN_7744854" 
<sip:pstn_7744854 at transit.vonage.net>;tag=1557830321
To: <sip:pstin_7748712 at inbound2.vonage.net>;tag=as141368b9
Call-ID: EED802D5-834B11DA-A1CCB29D-2266172B at transit.vonage.net
CSeq: 101 ACK
Max-Forwards: 13
Content-Length: 0
--- (10 headers 0 lines)---
  == CDR updated on SIP/pstin_7748712-a6ab
    -- Executing Playback("SIP/pstin_7748712-a6ab", "pbx-
transfer") in new stack
    -- Playing 'pbx-transfer' (language 'en')
    -- Executing Transfer("SIP/pstin_7748712-
a6ab", "SIP/18005551212 at sphone.vopr.vonage.net:5061") in new 
stack
set_destination: Parsing 
<sip:pstin_7748712 at sphone.vopr.vonage.net:5061> for address/port 
to send to
set_destination: set destination to sphone.vopr.vonage.net, port 
5061

########################################################
#  Here Asterisk Answers and initiates the transfer () #
########################################################


Reliably Transmitting (no NAT) to sphone.vopr.vonage.net:5061:
REFER sip:pstn_7744854 at transit.vonage.net:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk.aa.eu:5060;branch=z9hG4bK37019918;rport
Route: 
<sip:pstin_7748712 at inbound9.vonage.net:5060>,<sip:pstn_7744854 at tr
ansit.vonage.net:5060>
From: <sip:pstin_7748712 at inbound2.vonage.net>;tag=as141368b9
To: "PSTN_7744854" 
<sip:pstn_7744854 at transit.vonage.net>;tag=1557830321
Contact: <sip:pstin_7748712 at asterisk.aa.eu>
Call-ID: EED802D5-834B11DA-A1CCB29D-2266172B at transit.vonage.net
CSeq: 102 REFER
User-Agent: X-PRO Vonage release 1105x
Max-Forwards: 70
Refer-To: <sip:18005551212 at sphone.vopr.vonage.net:5061>
Referred-By: <sip:pstin_7748712 at asterisk.aa.eu>
---



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