Thanks for the fast reaction. 1. siproxd removed 2. Sipgate needs an "outboundproxy" equal to the "host". Here was one problem with GS3.1 because it automatically removed this in case both were equal. The rest stays as it was: - incoming NAT+forward rules unchanged - no outbound NAT rules added (like Static-port) - firewall rule exist, so that asterisk (10.150.0.14) is allowed to pass firewall (outgoing) - /etc/asterisk/sip.conf: no "externhost" set
Way It doesn't run with siproxd is not cleared, but for me is it good enough that it runs (anyway) now :-) It runs for incoming and outgoing calls, both are routed. Thanks for the report how you have done. Claudio BTW: What do you mean with "client" and not "peer"? Allowed sip-types are peer, user or friend (http://www.voip-info.org/wiki/view/Asterisk+sip+type) Am 15.10.2013 14:27, schrieb Vick Khera: > > On Tue, Oct 15, 2013 at 7:48 AM, Claudio Thomas <[email protected] > <mailto:[email protected]>> wrote: > > So my guess is that NAT+Portforwarding is not working correctly. Can > anyone help? > > Thanks, Claudio > > PS: annexed some details... > > asterisk <-> siproxd 0.8.0_2/pfSense 2.1(i386) <-> sipgate > 10.150.0.14 <-> 10.150.0.158/(pub-ip > <http://10.150.0.158/%28pub-ip> censored) <-> 217.10.68.150 > > > Our asterisk server is connected as a client to both Vitelity and > Skype for Business. Calls work both ways just fine. No siproxyd > involved at all. > > I do not connect as a peer. > > > > _______________________________________________ > List mailing list > [email protected] > http://lists.pfsense.org/mailman/listinfo/list
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