I've got same trouble. And fix it like you. On Tue, 08 Jun 2010 09:34:13 +0600, Matthew Lehner <mleh...@gmail.com> wrote:
> I am setting up opensips to act as a proxy between a SIP trunk > provider and more than one asterisk server. I am using alias_db to > determine which asterisk server a particular DID/user should be > relayed to. I am also using record_route() to ensure my proxy stays in > the entire dialog of the call. > > The initial requests go through just fine, but subsequent requests in > the same dialog from the SIP provider are not getting routed properly > because of loose_route(). > > When the request from the SIP provider arrives, it hits loose_route() > and the RURI gets changed to sip:222.222.222.227;lr=on which does not > contain a username and so alias_db can no longer match the call > details and route the request to the proper asterisk server. > > The way I understood loose_route() was supposed to work is.. it checks > the top-most Route header to see if it is the local proxy.. if it is > it removes that Route and if there is another Route below it.. it will > change the RURI to that. > > If I just don't do loose_route() on requests from the SIP provider, > everything works as expected.. but this does not seem like the right > solution to the problem. > > I have included debug output from opensips, along with some of my own logging. > > 333.333.333.x is the SIP provider > 222.222.222.x is my opensips proxy > > Regards, > > Matt > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature > database 5180 (20100607) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > -- Pavel Eremin _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
