[Asterisk-Dev] REFER / NOTIFY methods according to RFC-3515

[Lea]

Olle,

Thank you very much for replying !

The info that the transfer() application works differently from 
the Dial( ,t) application is not anywhere in the documentation.
Please consider adding it for the poor unaware users.

Anyway, I tried to transfer an incoming SIP call (
after it has been answered) using the following dialplan:
exten => 5,1,Transfer(SIP/18005551212 at sphone.vopr.vonage.net)
using the same channel as for the incoming call.

...and the Asterisk SIP debug output showed a REFER request 
being generated (see its listing below).  Great !

Unfortunately this REFER request was not accepted by the Vonage 
server and later Asterisk retransmitted it several times after 
which Asterisk gave up. Thus the TRANSFER FAILED :(

IMPORTANT:  When I tried to do an identical transfer of incoming 
call using an X-Pro Softphone, I immediately received a "202 
Accepted" response, thus the TRANSFER SUCCEEDED :)

The REFER request generated by the X-Pro Sofphone is very 
similar to Asterisk's (see the listings below).
The differences are:
- the host address after the REFER keyword is different
- the Refer-To header has an additional ;method=INVITE
- the Proxy Authorization header has a different uri=

Can you shed some light on the logic used by Asterisk when 
transfering calls using the REFER method ?
Do you understand why X-Pro Sofphone works, but Asterisk does 
not work ?

Regards,
Daniel Leeds


Debug output of Asterisk:
-------------------------------------------------
Reliably Transmitting (no NAT) to sphone.vopr.vonage.net:5061:
REFER sip:transit.vonage.net:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk.aa.eu:5060;branch=z9hG4bK35d81ab7;rport
Route: <sip:pstn_7748712@ 
inbound4.vonage.net:5060>,<sip:transit.vonage.net:5060>
From: <sip:pstn_7748712 at inbound4.vonage.net>;tag=as2dd427cd
To: “us_pstn_caller” <sip:transit.vonage.net>;tag=1180845188
Contact: <sip:pstn_7748712 at asterisk.aa.eu>
Call-ID: 33D60779-816711DA-BA1CDF96-A870513F at transit.vonage.net
CSeq: 103 REFER
User-Agent: Asterisk
Max-Forwards: 70
Proxy-Authorization: Digest username="pstn_7748712", 
realm="sphone.vopr.vonage.net", algorithm=MD5, 
uri="sip:sphone.vopr.vonage.net", nonce="1384655670", 
response="b29c87eeb7ec31847c4348b2e0b3bcbf", opaque=""
Date: Tue, 10 Jan 2006 23:25:30 GMT
Refer-To: <sip:18005551212 at sphone.vopr.vonage.net>
Referred-By: <sip:pstn_7748712 at asterisk.aa.eu>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, 
NOTIFY
Content-Length: 0

Debug output of X-Pro Softphone:
-------------------------------------------------
Transmitting to sphone.vopr.vonage.net:5061
REFER sip:pstn_7748712 at sphone.vopr.vonage.net:5061 SIP/2.0
Via: SIP/2.0/UDP 
asterisk.aa.eu:5060;rport;branch=z9hG4bK1CFB3639BC0147BE98519AA14
B1FE8A1
Route: 
<sip:pstn_7748712 at inbound4.vonage.net:5060>,<sip:transit.vonage.c
om:5060>
From: <sip:pstn_7748712 at inbound4.vonage.net>;tag=3655015093
To: “us_pstn_caller” <sip:transit.vonage.com>;tag=2112301987
Contact: <sip:pstn_7748712 at asterisk.aa.eu:5060>
Call-ID: CF6FC3F9-800111DA-999DDF96-A870513F at transit.vonage.com
CSeq: 7237 REFER
User-Agent: X-PRO release 1105x
Max-Forwards: 70
Proxy-Authorization: Digest 
username="pstn_7748712",realm="sphone.vopr.vonage.net",nonce="529
605995",response="ff4eadf3e625217233fdbff90fd67ec5",uri="sip:pstn
_7748712 at sphone.vopr.vonage.net:5061",algorithm=MD5
Refer-To: <sip:18005551212 at sphone.vopr.vonage.net;method=INVITE>
Referred-By: Vonage User 
<sip:pstn_7748712 at sphone.vopr.vonage.net>
Content-Length: 0


On 10-01-2006,. 16:36 Olle E Johansson wrote:
> We do support REFER/NOTIFY.
> I am working (and have been for a long time) on improving the 
support, but the basic support for transfers with REFER is there.

In another topic Olle E Johansson wrote:
>We only issue an outbound REFER by you using the transfer() 
application in the dial plan.

> Daniel Leeds wrote:
> > I noticed that in the file "chan_sip.c" there is a constant 
> > SIP_REFER. 
> > 
> > Does this mean that Asterisk can initiate SIP-SIP transfers 
with 
> > the REFER / NOTIFY methods according to RFC-3515 ? 
> > 
> > If not, why not ? ...after all RFC-3515 is almost 3 years 
old 
> > and I thought Asterisk is on the cutting edge of VoIP... 
> 
> > 
> > Anyway, if somebody has some code to make Asterisk compliant 
> > with this RFC-3515 or some hidden #defines or maybe compiler 
> > options to enable this crucial transfer functionality, 
please 
> > scribble something back. 
> 
> /O
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