Adrian, So in that case I would need to have a PBX, such as Asterisk, or FreeSWITCH to handle this case, which would mean the PBX has to be in the call path the entire time?
Thanks. On 9/15/10 12:17 PM, Adrian Georgescu wrote: > Call transfer using SIP require cooperation between the 3 end-points involved. > > On the public internet, which is probably why you want to use OpenSIPS, this > does not work primarily because of accounting reasons. You can configure the > SIP proxy to route the Refer and Notify alright but as an end-point you will > not succeed at instructing for instance a remote PSTN gateway to transfer > the call to some other end-point because you have no trust relationship with > it and nobody can be billed for this new call leg. > > Handling the routing of REFER in OpenSIPS is trivial but it will not solve > your problem. > > The only way you can make transfer work reliably is behind the same PBX. > > Adrian > > > On Sep 15, 2010, at 6:02 PM, David J. wrote: > >> Seems like when I try to transfer a call from one user to the next, it >> does not do anything, so I am guessing we have to handle the REFER message? >> >> What is the best practice for handling REFER messages? >> >> Thanks. >> >> >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
