Someone? As * is used so extensively with SIP I must've made a _glaring_ mistake in my config (!)
/Rob Robert Bielik skrev: > Tarek Sawah skrev: >> you need to post you SIP.conf and your Extensions.conf so someone can >> have a look at them and see if there is anything missing >> what are the contexts you are using with your peers? >> what is the dial plan triggered when calling your destination number? > > Machine 1 ------------------------------------------------------- > iax.conf: ====================== > [general] > bandwidth=low > disallow=lpc10 ; Icky sound quality... Mr. Roboto. > jitterbuffer=no > forcejitterbuffer=no > autokill=yes > > [2200] > type=friend > host=dynamic > context=users > username=2200 > secret=none > auth=md5 > > sip.conf ======================= > [general] > port=5060 > bindaddr=0.0.0.0 > > disallow=all > allow=alaw ; Allow codecs in order of preference > allow=ulaw > allow=gsm > allow=g726 > > dtmfmode=rfc2833 > > register => machine_1:wabo...@192.168.10.77/machine_2 > > [machine_2] > allow=alaw,ulaw,gsm,g726 > host=dynamic > secret=wabooba > type=friend > context=sip_incoming > username=machine_2 > > extensions.conf ================== > [general] > static=yes > writeprotect=no > clearglobalvars=no > > [globals] > ; The outgoing sip trunk > SIP_TRUNK=192.168.10.77 > OUTGOING_PREFIX=0 > > [default] > include => sip-incoming > include => test > > [test] > ; Create an extension, 600, for evaluating echo latency. > ; > exten => 600,1,Playback(demo-echotest) ; Let them know what's going on > exten => 600,n,Echo ; Do the echo test > exten => 600,n,Playback(demo-echodone) ; Let them know it's over > exten => 600,n,Goto(s,6) > > [users] > include => sip-incoming > include => outgoing > include => test > > [sip-incoming] > include => agi-async > include => internal > > [agi-async] > exten => _01XXXX,1,Agi(agi:async) > > [internal] > exten => _2XXX,1,NoOp() > exten => _2XXX,n,Dial(IAX2/${EXTEN}) > exten => _2XXX,n,Hangup() > > [outgoing-agi-async] > exten => _${OUTGOING_PREFIX}.,1,Dial(SIP/${ext...@${sip_trunk}) > exten => _${OUTGOING_PREFIX}.,n,Set(CALLERID(name)=reason-${DIALSTATUS}) > exten => _${OUTGOING_PREFIX}.,n,Agi(agi:async) > > [outgoing] > exten => _${OUTGOING_PREFIX}.,1,Dial(SIP/${SIP_TRUNK}/${EXTEN:1}) > exten => _${OUTGOING_PREFIX}.,n,Hangup() > > Machine 2 -------------------------------------------------------- > sip.conf ======================= > [general] > port=5060 > bindaddr=0.0.0.0 > > disallow=all > allow=alaw ; Allow codecs in order of preference > allow=ulaw > allow=gsm > allow=g726 > > dtmfmode=rfc2833 > > register => machine_2:wabo...@192.168.10.11/machine_1 > > [machine_1] > allow=alaw,ulaw,gsm,g726 > host=dynamic > secret=wabooba > type=friend > context=sip_incoming > username=machine_1 > > extensions.conf ================== > [globals] > ; The outgoing sip trunk > SIP_TRUNK=192.168.10.11 > > Rest is exactly the same. I have a zoiper connected to each machine and I'm > trying to make a call from Machine 2 to zoiper > on Machine 1: > > -- Registered IAX2 '2200' (AUTHENTICATED) at 192.168.10.113:4569 > -- Accepting AUTHENTICATED call from 192.168.10.113: > > requested format = gsm, > > requested prefs = (), > > actual format = gsm, > > host prefs = (), > > priority = mine > -- Executing [02...@users:1] Dial("IAX2/2200-1200", > "SIP/192.168.10.11/2200") in new stack > == Using SIP RTP CoS mark 5 > -- Called 192.168.10.11/2200 > [Oct 26 09:20:25] NOTICE[20248]: chan_sip.c:15031 handle_response_invite: > Failed to authenticate on INVITE to '"2200" > <sip:2...@192.168.10.77>;tag=as6173091f' > -- SIP/192.168.10.11-090c2ea8 is circuit-busy > == Everyone is busy/congested at this time (1:0/1/0) > -- Executing [02...@users:2] Hangup("IAX2/2200-1200", "") in new stack > == Spawn extension (users, 02200, 2) exited non-zero on 'IAX2/2200-1200' > -- Hungup 'IAX2/2200-1200' > > Besides that "sip show peers" on either machine shows the other one correctly > registered, and "iax2 show peers" shows the connected zoiper on each machine. > > Ideas, please ?? > > TIA > /Rob > > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users