JFYI, http://svn.digium.com/svn/asterisk/team/oej/codename-pineapple
Not Found The requested URL /svn/asterisk/team/oej/codename-pineapple was not found on this server. ------------------------------------------------------------------------ -------- Apache Server at svn.digium.com Port 80 -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E Johansson Sent: Sunday, June 10, 2007 5:08 PM To: Asterisk Developers Mailing List Subject: Re: [asterisk-dev] SIP over TCP/SCTP 10 jun 2007 kl. 13.49 skrev Ibrar Ahmed: > Hi, > > I have been working on asterisk since last 2/3 years. I have > implemented pay phone channel based on mgcp protocol(chan_mgcp.c) > and also worked on ss7 channel implementation. Now I feel I should > contribute on community. I am working on sip over TCP/SCTP. I have > some question about this > > 1 - Is any body working on this. Yes for TCP, I have never got any requests for SCTP though. > 2 - How much important this feature is. Very See http://www.codename-pineapple.org for more information about changes that are being worked on. /O _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
