[asterisk-dev] SIP issue (chan_sip.c:7724 set_address_from_contact) with Nortel CS1K

Raj Jain rj2807 at gmail.com
Sun Apr 1 12:38:35 MST 2007


Right. The content of "phone-context" is irrelevant except that * needs to
be able to correctly parse it. Once you overcome the parsing issue in * (the
presence of semicolon in userinfo), the ACK will be delivered to maddr
47.160.109.70 anyway.

Finding the media address (c= line in 200 OK's SDP) is an entirely different
thing than finding the SIP remote target address (Contact URI in this case)
because that's where the ACK will be sent. Right now the problem is that
since * is not able to parse Contact header in 200 OK correctly, it is
unable to send the ACK.


On 4/1/07, Christopher Thompson <christopher.thompson at zen.co.uk> wrote:
>
> There was an @ in the section I *BLANKED* originally, the phone-context
> field.
>
> I am not entirely certain any of this will solve the issue anyway.
> Nortel's "phone-context" is basically "for show" in that that most
> enterprise domains etc are not typically accessable from the net anyway.
> Asterisk has already found the media address and port so I am not sure what
> the purpose of any of this is?
>
>
> Raj Jain wrote:
>
>  <sip:65602093;phone-context=*BLANKED*:5060;maddr=47.160.109.70
> ;transport=udp;user=phone>
>
> The above URI seems syntactically incorrect to me. This is a tel URI
> converted to a SIP URI. Agreed that anything before the @ sign (including
> the semicolon) needs to treated opaquely as userinfo. The host, which is a
> mandatory part of SIP URI, seems to be missing in the above URI.
>
> Someone mentioned that there is an @host in the above URI. I can't see it.
> What am I missing?
>
> Raj
>
>
> On 4/1/07, Olle E Johansson <olle at voop.com> wrote:
> >
> >
> > 1 apr 2007 kl. 13.02 skrev Mikael Magnusson:
> >
> > > Olle E Johansson wrote:
> > >> 30 mar 2007 kl. 23.52 skrev Christopher Thompson:
> > >>> Contact: <sip:65602093;phone-context=*BLANKED*:
> > >>> 5060;maddr=47.160.109.70;transport=udp;user=phone>
> > >> This is the culprit. There's no @ sign in this contact. nortel has
> > >> a way of implementing their own stuff, not really
> > >> bothering about interoperability.
> > >
> > > Are you sure there isn't a @ in *BLANKED*? That would make it a
> > > valid sip uri. Similar to the following from RFC 4504:
> > >
> > >     sip:5551234;phone-context=+1212 at example.net;user=phone
> >
> > If so, then we're back to us not correctly supporting ;parameters in
> > the username part, which was supposed to be fixed.
> >
> > However, I'm sure it is fixed in Asterisk SPE 1.0B.
> >
> > /O
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