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

Lea lea123 at wp.pl
Tue Jan 10 22:44:15 MST 2006 

It appears that for an existing SIP dialog, the Request-URI 
after the REFER keyword, needs to be taken from the first value 
in the Route: header, not from the Contact: header.

See the RFC-3261, Section: 12.2.1.1 : Generating the Request

This might be the reason why the Request-URI after the REFER 
keyword is wrong in the Asterisk SIP debug output below.

This od course does not solve the 2nd problem of the wrong value 
in the Request-To: header.

Apparently Asterisk is cannot recognize when the argument to the 
app_transfer() is:
app_transfer(Tech/user at FQDN)
  or
app_transfer(Tech/user at AsteriskPeer)

...Finally, there is the issue of the missing ;method=INVITE in 
the Refer-To: header.


Regards,
Daniel Leeds

On 11-01-2006,1:56 Daniel Leeds wrote:
> 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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> > 
> 
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