Hi, Here is what I have in my sip.con
[148] type=friend username=148 secret=something host=dynamic canreinvite=no qualify=no context=interne dtmfmode=rfc2833 mailbox=148 language=fr [spa3kphone00] type=friend host=dynamic context=interne secret=something dtmfmode=rfc2833 disallow=all allow=ulaw canreinvite=no language=fr The spa3000 is on 10.10.11.10 and the phone (Mitel 5215) is on 10.10.10.52. The spa3k can send packets to the Mitel, but not the other way. I would like this this setup to remain the same The asterisk has one nic on each network 10.10.11.1 and 10.10.10.25. The asterisk box is configured to do the ip forwarding for the sipura (the gateway) so that some pc can access the configuration panel of the sipura but the mitel does'nt have a route to the sipura. Only the FXS port is used on the SPA3K with a phone. The problem is when there is a call from or to the spa3k, asterisk try to do a Native Bridge and fail to do so. After when I hangup the FXS on the SPA3K, Asterisk do not get the end of the call. On the other side, if I hangup the Mitel, everything works ok. If I hang-up the call from the sipura or the Mitel before asterisk try the native bridge, everything is ok. Thanks a lot for your help. ___________________________ Jean-François Rousseau www.sys-tech.net [EMAIL PROTECTED] Tél. 24h (418) 520-0739 Télec. (418) 520-4554 1-877-969-tech Ouverture Technologique -----Message d'origine----- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Steve Totaro Envoyé : 14 décembre 2005 19:57 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] How to disable sip Native bridge > > Hi, > > I'm trying to disable Native bridges between two SIP Phones. This is > because they both see the asterisk box, but they can't see eachother (no > it isn't because of NAT). > > I've tried putting canreinvite=no everywhere in my config, but asterisk is > still trying a native bridge on the call. The problem is that when this > happen, the native bridge fail but one phone (Sipura 2000) think that the > bridging was done and the BYE is not received by asterisk when the call > end. > > So the question is, Is there a way to disable this behavior ? > > Thanks > Post your SIP conf. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
