On Tue, Jan 7, 2014 at 5:03 PM, Matthew Jordan <[email protected]> wrote:
> On Tue, Jan 7, 2014 at 9:27 AM, Joshua Colp <[email protected]> wrote: > > Olle E. Johansson wrote: > >> > >> On 07 Jan 2014, at 15:55, SVN commits to the Digium > >> repositories<[email protected]> wrote: > >> > >>> if (endpoint->nat.rewrite_contact&& (contact = > >>> pjsip_msg_find_hdr(rdata->msg_info.msg, PJSIP_H_CONTACT, NULL))&& - > >>> > >>> (PJSIP_URI_SCHEME_IS_SIP(contact->uri) || > >>> PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) { + > !contact->star&& > >>> (PJSIP_URI_SCHEME_IS_SIP(contact->uri) || > >>> PJSIP_URI_SCHEME_IS_SIPS(contact->uri))) { pjsip_sip_uri *uri = > >>> pjsip_uri_get_uri(contact->uri); > >>> > >>> pj_cstr(&uri->host, rdata->pkt_info.src_name); > >>> > >> > >> Seems like this code assumes that SIP only have SIP: and SIPS: as > >> URI's. We should actually be quite transparent in the schemas > >> supported - especially proxys but also b2bua's like Asterisk. Tel: > >> uri's are not unknown. In the security area we are discussing > >> improved end-2-end security which may end up using a new SIP uri. > > > > > > Hrm, I've been trying to think of the best place to express this > information > > but I'm coming up empty. I'm not sure it's something for the config > > documentation, maybe more the wiki. > > > > > >> Instead of testing with two functions for two classes of schemas a > >> table could be used? That would be more extensible. And please > >> implement tel: support :-) > > > > > > The code above stems from PJSIP itself and is used to determine what > > structure PJSIP has used to store the parsed URI for manipulation. I'm > not > > sure a table would buy us anything in this area. As a developer you would > > still need to know what structure was used and what it looks like. > > > > We probably should have an issue made to (finally) implement TEL > support. I know PJSIP supports it, but there'd obviously be some > subtle things that would have to change in how we make use of PJSIP. > > I'll make an issue for it later today. > > -- > Matthew Jordan > Digium, Inc. | Engineering Manager > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: http://digium.com & http://asterisk.org > > -- > _____________________________________________________________________ > -- 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 > I love this fix, as the issue it fixes is causing my Asterisk 12 install to die... https://gist.github.com/danjenkins/e604c978b2a803140dce Any idea when the next point release will be? Is it so far off I should just recompile against the 12 branch later on? Dan
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