Hello, Can someone please give me some help....
I am trying to use asteriskout.com gateway to my asterisk server but I am doing something wrong with the sip setup. as it does not place the call correct in the asteriskout.com servers. A help from the asteriskout.com tells that their side need an invite and not a register... so I coded in the sip.conf.... [ipcb] type=peer username=uuuuuuuu secret=ppppppppp host= a.b.c.d (the ip from the asteriskout.com server) fromuser= uuuuuuu port=5070 --------------------------------- and in extensions.conf.... [ipcb] exten=>_.,1,Dial (SIP/[EMAIL PROTECTED]) ------------------------------------------------------ reload the configuration.... my asterisk sends register to the asteriskout.com if I changed the extensions.conf... to.... -> [ipcb] exten=> _.,1,dial(SIP/user:[EMAIL PROTECTED]:5070/[EMAIL PROTECTED]) it sends the invitation, starts the call... but....replies with: Jul 21 19:03:52 WARNING[72115]: chan_sip.c:1048 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Critical Request) == No one is available to answer at this time (1:0/0/0) I am using the last CVS version.... everything works... Another question is: Is there a way to tell asterisk to send an invite without register??? an option in sip.conf??? Thanks for any help, Sergio _______________________________________________ Asterisk-BSD mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-bsd
