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I've been using the wiki as a guide here... http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net I've also tried various permutations, combinations and contortions but nada. below is my extensions.conf ***** begin extensions.conf **** [general] static = yes writeprotect = no autofallthrough = yes clearglobalvars = no [globals] CONSOLE = Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO = guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK = Zap/G2 ; Trunk interface TRUNKMSD = 1 ; MSD digits to strip (usually 1 or 0) ;TRUNK=IAX2/user:pass provider [default] exten => s,1,Verbose(1|Unrouted call handler) exten => s,n,Answer() exten => s,n,Wait(1) exten => s,n,Playback(tt-weasels) exten => s,n,Hangup() [macro-voicemail] exten => s,1,Dial(${ARG1},20) exten => s,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN}) exten => s-NOANSWER,n,Hangup() exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN}) exten => s-BUSY,n,Hangup() exten => _s-.,1,Goto(s-NOANSWER,1) [incoming_calls] exten => ekiga_meyamma_in,1,Macro(voicemail,SIP/101) exten => ekiga_meyamma_in,n,Hangup() [internal_calls] exten => 101,1,Macro(voicemail,SIP/101) exten => 101,n,Hangup() exten => 102,1,Dial(SIP/102) exten => 102,n,Hangup() exten => 8,1,VoiceMailMain(s${CALLERIDNUM}) exten => 8,n,Hangup() exten => 600,1,Answer() exten => 600,n,Playback(demo-echotest) exten => 600,n,Echo() exten => 600,n,Playback(demo-echodone) exten => 600,n,Hangup() [outgoing_calls] exten => _9.,1,Dial(SIP/ekiga_meyamma_out/${EXTEN:1},20,r)) exten => _9.,n,Hangup() [home] include => internal_calls include => outgoing_calls ***** end extensions.conf **** below is my sip.conf **** begin sip.conf *** [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm allow=h261 videosupport=yes ; Register 2345 at sip provider 'sip_proxy'. Calls from this provider ; connect to local extension 1234 in extensions.conf, default context, ; unless you configure a [sip_proxy] section below, and configure a ; context. ; Tip 1: Avoid assigning hostname to a sip.conf section like [provider.com] ; Tip 2: Use separate type=peer and type=user sections for SIP providers ; (instead of type=friend) if you have calls in both directions register => meyamma:meyamma ekiga net/ekiga_meyamma_in externhost=fspublic.selfip.com externrefresh=10 ; localnet=192.168.1.0/255.255.255.0 [authentication] ;setup [101] type=friend username=101 secret=welcome qualify=yes nat=no host=dynamic canreinvite=no context=home ;port=5061 ;setup [102] type=friend username=102 secret=welcome qualify=yes nat=no host=dynamic canreinvite=no context=home ;port=5061 ;ekiga.net [ekiga_meyamma_out] type=peer username=meyamma secret=meyamma host=ekiga.net canreinvite=no qualify=300 insecure=port,invite ;ekiga.net [ekiga_meyamma_in] type=user username=meyamma secret=meyamma host=ekiga.net canreinvite=no qualify=300 context=incoming_calls insecure=port,invite **** end sip.conf **** These are the latest incantations of my sip and extensions conf files. Can anyone assist in helping me to get incoming calls from ekiga.net set up. I have the outgoing working as that was the simple part. That one is handled by [ekiga_meyamma_out] and [outgoing_calls]. Thanks a mil Explore the seven wonders of the world Learn more! |