Asterisk + libss7 [3rd]
Connecting and placing calls: [X ~]# cat /etc/asterisk/sip.conf [general] context=[your context] ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=[your IP] ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos_sip=cs3 ; Sets TOS for SIP packets. tos_audio=ef ; Sets TOS for RTP audio packets. cos_sip=3 cos_audio=5 relaxdtmf=yes canreinvite=no disallow=all ; see doc/rtp-packetization for framing options allow=g729 ; add whatever codec you have and like to use ;allow=ulaw ;allow=alaw allowguest=no notransfer=yes ; part of rtp timeout rtptimeout=19 rtpholdtimeout=20 [client1] type=peer insecure=port,invite host=[client ip] notransfer=yes disallow=all allow=g729 ;allow=g723 al...