[Asterisk-Dev] no 1xx messages following reinvite with authentication

Alex Zeffertt ajz at cambridgebroadband.com
Wed Sep 29 03:49:59 MST 2004


Hi All,

Please ignore this last message.  It was the fault of the UA which was
buggering up the Via field in it's INVITE+auth messages.

Thanks,

Alex

On Tue, 28 Sep 2004 14:57:02 +0100
Alex Zeffertt <ajz at cambridgebroadband.com> wrote:

> On a related note....
> 
> I'm having a problem with SIP authentication.  When my UA INVITEs
> Asterisk I get the following exchange:
> 
> 	UA					Asterisk
> 		-INVITE----------->
> 		<-407 Auth reqd--
> 		-ACK--------------->
> 		-INVITE+auth---->
> 		-INVITE+auth---->
> 		-INVITE+auth---->
> 		-INVITE+auth---->
> 	            ....
> 		<-200 OK---------- (other end goes off-hook)
> 		-ACK--------------->
> 
> The problem is that the UA never gets an informative 1xx response to
> its INVITE+auth message.
> 
> This means that the UA can only assume that Asterisk has not received
> the INVITE+auth and it keeps retransmitting it.  It also means that
> the UA cannot generate an appropriate progress tone, e.g. a ring tone.
> 
> Why doesn't Asterisk generate a "100 Trying"  or "180 Ringing" message
> when using authentication?
> 
> Alex
> 
> 
> 
>  On Tue, 28 Sep 2004 09:31:52 -0400
> "Jan Hulala" <hulala at xodatel.com> wrote:
> 
> > Thank you guys, I already solved this issue.
> > 
> > 
> > Jan
> > 
> > 
> > ----- Original Message -----
> > From: "Alex Zeffertt" <ajz at cambridgebroadband.com>
> > To: "Jan Hulala" <hulala at xodatel.com>; "Asterisk Developers Mailing
> > List"<asterisk-dev at lists.digium.com>
> > Sent: Tuesday, September 28, 2004 6:05 AM
> > Subject: Re: [Asterisk-Dev] Don't kick my ass, please help...
> > 
> > 
> > > I can't help without knowing your secret :-)
> > > Send it and I'll show you what linphone puts in its md5sum.
> > >
> > > Alex
> > >
> > >
> > > On Mon, 27 Sep 2004 02:18:40 -0400
> > > "Jan Hulala" <hulala at xodatel.com> wrote:
> > >
> > > > Please help mi with my MD5-ing... :(((
> > > >
> > > > What wrong is in my response calculation (see below)...?
> > > >
> > > > Thanks,
> > > > Jan
> > > >
> > > >
> > > > ---------------------------------------------------------------
> > > > -----------------
> > > >
> > > >
> > > > SIP message from server:
> > > >
> > > >       SIP/2.0 407 Proxy Authentication Required
> > > >       .
> > > >       .
> > > >       .
> > > >       Proxy-Authenticate: Digest realm="asterisk",
> > > >       nonce="3594843a" Content-Length: 0
> > > >
> > > >
> > > > Perl script for calculation REGISTER reponse:
> > > >
> > > >       $md5 = Digest::MD5->new;
> > > >       $md5->add($username, ':', $realm, ':', $password);
> > > >       $HXA = $md5->hexdigest;
> > > >
> > > >       $md5 = Digest::MD5->new;
> > > >       $md5->add('md5', ':');
> > > >       $HXB = $md5->hexdigest;
> > > >
> > > >       $md5 = Digest::MD5->new;
> > > >       $md5->add($HXA, ':', $nonce, ':', $HXB);
> > > >       $response = $md5->hexdigest;
> > > >
> > > >  ---> Proxy-Authorization: Digest realm="asterisk",
> > > >  nonce="3594843a",
> > > >         username="XXXXXXXXX", algorithm="md5",
> > > >         response="cd8b4824f70bfdd6b686f9abd851cf3b"
> > > >
> > > >
> > > > SIP response message:
> > > >
> > > >       SIP/2.0 401 Unauthorized
> > > >       .
> > > >       .
> > > >       .
> > > >       Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> > > >       Content-Length: 0
> > > >
> > > >
> > >
> > >
> > > --
> 



More information about the asterisk-dev mailing list