On 06 Oct 2014, at 16:21, Matthew Jordan <mjor...@digium.com> wrote:
> On Mon, Oct 6, 2014 at 1:25 AM, Olle E. Johansson <o...@edvina.net> wrote: >> >> On 06 Oct 2014, at 02:59, SVN commits to the Digium repositories >> <svn-comm...@lists.digium.com> wrote: >> >>> >>> An OPTIONS request that is sent to Asterisk but not to a specific endpoint >>> is >>> currently sent a 404 in response. This is because, not surprisingly, an >>> empty >>> extension is never going to be found in the dialplan. >>> >>> This patch makes it so that we only attempt to look up the endpoint in the >>> dialplan if it is specified in the OPTIONS request URI. >> >> An empty extension in Asteirsk is the "s" extension, right? >> In chan_sip we answer with 200 OK if an "s" extension can be found, >> otherwise an error is generated. >> >> If that's a good or bad solution is a differernt thing - but we do handle >> empty extensions, >> especially in the dahdi channel. >> > > Yup - chan_sip, prior to calling ast_exists_extension, will often > insert an "s" if the request URI does not contain a user portion as > well. We aren't currently doing that in the part of chan_pjsip that > handles OPTIONS requests. My commit message probably could have been a > bit clearer - the ast_exists_extension does not handle 'blank' > extensions by itself, but other places prevent a blank extension from > being passed to that function by providing the "s" extension if > nothing else was specified. When a channel is created it uses s@default by default... > > For OPTIONS requests in particular, I'm not sure that really is super > useful. Clearly, an inbound call is a different matter. But for > OPTIONS requests, having to provide the "s" extension to prevent a 404 > when the request URI did not specify a username feels odd. > > We could do that if people liked, however :-) I don't like that behaviour. A bigger issue is if we should at any point try to implement "real" options answers. After authentication we could add an SDP... /O -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
