When I made a call using an older version I saw, using checkpoint's user monitor that the call was indeed using RTP (somewhere between 10000 and 20000, dynamically set for each call).
After I upgraded the firmware, the entire conversation stays on the sip port. > -----Original Message----- > From: Jon Lawrence [mailto:[EMAIL PROTECTED] > Sent: Tuesday, December 14, 2004 10:30 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Asterisk to sip client behindFirewall/NAT- > cancall but cannot receive calls ? > > On Tuesday 14 December 2004 15:19, Shoval Tomer wrote: > > As far as I can remember I only opened sip and tftp ports for the phone. > > > > For some reason (didn't look into it too much) the call stays with sip > > and doesn't use RTP. > > > > SIP is what sets up the session (ie it does session handling) > RTP is the transport protocol that the audio uses. > > If you're using SIP then you're using RTP eos. > > Jon > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > MailScanner thanks transtec Computers for their support. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users