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