I had to drop realtime with sip users. If you do a reload or a restart, you lose all the sip peer information (even with rtcachefriends=yes). That just wasn't acceptable for us.
> -----Original Message----- > From: JR Richardson [mailto:[EMAIL PROTECTED] > Sent: Tuesday, March 21, 2006 4:06 PM > To: [email protected] > Subject: [Asterisk-Users] Fw: anybody has SIP realtime working ? > > > > Message: 16 > > Date: Tue, 21 Mar 2006 18:51:29 -0300 > > From: "Frederic Jean" <[EMAIL PROTECTED]> > > Subject: [Asterisk-Users] Fw: anybody has SIP realtime working ? > > To: <[email protected]> > > Message-ID: <[EMAIL PROTECTED]> > > Content-Type: text/plain; charset="iso-8859-1" > > > > Hello, > > > > I am just asking this because I am note sure if the problem > > is on my side or not, I saw some comments on SIP realtime > > today so I was wondering, has anybody has SIP realtime working > > with a softfone ? > > > > If yes, please confirm, that would give me a light. > > My previous message to the list is below. > > > > Thanks. > > > > Frederic > > > Yes, > > I have realtime working with SIP Cisco and Polycom hard > phones, and DIAX Softphones for IAX (pretty much same config > as SIP. Sip and Iax realtime works fine for me, backing into > a MySQL database. If the softphone is giving you a problem, > try another sofphone, there are a lot if free ones to try. > There is no reason why a soft phone would not work and a hard > phone would work, except configuration or SIP stack > implementation on the soft phone. > > JR Richardson > Engineering for the Masses > > _______________________________________________ > --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
