SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile phone) using a Dial() command.
Because both parties are external, I don't want the media to pass through my Asterisk box once the two parties connect. This is the default behaviour for the Dial command if you avoid the tT options and the codec is supported all the way through - this is true in my case. I have this working great with IAX - I can even disconnect the Ethernet cable from my Asterisk box once the call is established and the call is not affected. Unfortunately I cannot get it to work with SIP and my ITSP is dropping support for IAX. Can you help me? I have canreinvite=yes in sip.conf and I've tried with nat=yes and nat=no, but no luck either way - the call goes through in each case but the media is passing through my Asterisk box and i'd like to avoid that. I have tried without the attendant too. I've tried answering, waiting 5 seconds then direct to a Dial command. I've also tried, not answering, and go straight to the Dial command. No luck either way. The sip.conf and extensions.conf for the former are below. Anxiously awaiting a reply. Thanks, H ;************************ SIP.conf ********************************** [general] ; context=incoming-bogus-calls bindport=5060 ; Port to bind to (SIP is 5060) bindaddr=0.0.0.0 ; Address to bind to (all addresses on machine) maxexpirey=3600 ; Must be larger than the re-register timeout on the router defaultexpirey=3600 notifymimetype=text/plain rtptimeout=60 rtpholdtimeout=300 disallow=all allow=ulaw ; ; This section is because i'm behind nat ; externip=999.99.999.99 ;Outside address localnet=192.168.0.148/255.255.255.0 ;Inside Network ; ;===================== myITSP ============================ register => myuserid:[EMAIL PROTECTED]:5060 register => myuserid:[EMAIL PROTECTED]:5060 ; [myITSP-sw1] context=incoming-sip type=friend host=sw1.myITSP.com username=myuserid secret=mypassword nat=yes canreinvite=yes insecure=port,invite ; do NOT remove this qualify=yes ; do NOT remove this dtmfmode=rfc2833 ; should match what is set on your account disallow=all allow=ulaw ; set in/out codec here ; [myITSP-sw2] context=incoming-sip type=friend host=sw2.myITSP.com username=myuserid secret=mypassword nat=yes canreinvite=yes insecure=port,invite ; do NOT remove this qualify=yes ; do NOT remove this dtmfmode=rfc2833 ; should match what is set on your account disallow=all allow=ulaw ; set in/out codec here ;************************ EXTENSIONS.conf ********************************** ; ----------------------------------------------------------- ; /etc/asterisk/extensions.conf ; ;--------------------------------------------------------------------- ;============================== GENERAL ============================== ;--------------------------------------------------------------------- [general] ; static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here. ; ;--------------------------------------------------------------------- ;============================== GLOBALS ============================== ;--------------------------------------------------------------------- ; [globals] ; ;Dial Patterns LOCAL_PATTERN=_NXXXXXX LD_PATTERN=_1XXXXXXXXXX INT_PATTERN=_011. FORCE_SIP_LOCAL_PATTERN=_8NXXXXXX FORCE_SIP_LD_PATTERN=_81XXXXXXXXXX FORCE_SIP_INT_PATTERN=_8011. FORCE_FXO_LOCAL_PATTERN=_9NXXXXXX FORCE_FXO_LD_PATTERN=_91XXXXXXXXXX FORCE_FXO_INT_PATTERN=_9011. ; ;--------------------------------------------------------------------- ;============================= MACRO ============================== ;--------------------------------------------------------------------- [macro-dialmyITSP] exten => s,1,Dial(SIP/[EMAIL PROTECTED],60) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-ANSWER,1,Hangup exten => s-CONGESTION,1,Dial(SIP/[EMAIL PROTECTED],60) exten => s-CONGESTION,2,Goto(ss-${DIALSTATUS},1) exten => s-CANCEL,1,Hangup exten => s-BUSY,1,Busy(30) exten => s-CHANUNAVAIL,1,Dial(SIP/[EMAIL PROTECTED],60) exten => s-CHANUNAVAIL,2,Goto(ss-${DIALSTATUS},1) exten => ss-ANSWER,1,Hangup exten => ss-CONGESTION,1,Congestion(30) exten => ss-CANCEL,1,Hangup exten => ss-BUSY,1,Busy(30) exten => ss-CHANUNAVAIL,1,Congestion(30) ;--------------------------------------------------------------------- ;============================= CONTEXTS ============================== ;--------------------------------------------------------------------- [cleanup] ; ; If the line hangs up, it's always good to have the "h" ; extension in each context that is the "master" handler ; for calls. This cleanly exits and closes dial path routes. ; exten => t,1,NoOp("Timeout extension in context "${CONTEXT}".") exten => t,n,Hangup ; exten => T,1,NoOp("Timeout on AbsoluteTimeout "${CONTEXT}".") exten => T,n,Hangup ; exten => h,1,Hangup ; ; The user has dialed an "i"nvalid number, which means that ; there was no match by any other matching routines. Set an ; absolute timeout on the call (15 seconds), play a Congestion ; tone, and hangup. We set the absolute timeout to prevent easy ; DoS attacks from consuming too much bandwidth. However, it ; is possible that we could still be attacked in some fashion ; by someone making many calls to bogus numbers on our server. ; We could reduce this threat by removing the Congestion ; playback and going straight to hangup, but that is very ; difficult to debug at the remote end, so we are good VoIP ; citizens and we create some audio if the call reaches us. ; exten => i,1,AbsoluteTimeout(15) exten => i,n,Playtones(congestion) exten => i,n,Congestion exten => i,n,Hangup ; ;--------------------------------------------------------------------- ; [Local-Calls-PBX] ;Contains dial patterns for calls within the same area code as the Asterisk server. ; ;This particular context [*-PBX] is used by clients and/or devices that are directly ;connected to the Asterisk PBX. We allow these calls to be transferred. ; ;Default ; exten => ${LOCAL_PATTERN},1,Macro(dialmyITSP,1613${EXTEN}) ; ;--------------------------------------------------------------------- ; [Toll-Free-Calls-PBX] ;Contains dial patterns for toll free calls. ; ;This particular context [*-PBX] is used by clients and/or devices that are directly ;connected to the Asterisk PBX. We allow these calls to be transferred. ; ;Default exten => _1888.,1,Macro(dialmyITSP,${EXTEN}) exten => _1877.,1,Macro(dialmyITSP,${EXTEN}) exten => _1866.,1,Macro(dialmyITSP,${EXTEN}) exten => _1855.,1,Macro(dialmyITSP,${EXTEN}) exten => _1800.,1,Macro(dialmyITSP,${EXTEN}) ; ;--------------------------------------------------------------------- ; [NA-Long-Distance-Calls-PBX] ;Contains dial patterns for North American long distance calls. ; ;This particular context [*-PBX] is used by clients and/or devices that are directly ;connected to the Asterisk PBX. We allow these calls to be transferred. ; ;Default exten => ${LD_PATTERN},1,Macro(dialmyITSP,${EXTEN}) ; ;--------------------------------------------------------------------- ; [INT-Long-Distance-Calls-PBX] ;Contains dial patterns for International (from a North American view point) long ;distance calls. ; ;This particular context [*-PBX] is used by clients and/or devices that are directly ;connected to the Asterisk PBX. We allow these calls to be transferred. ; ;Default exten => ${INT_PATTERN},1,Macro(TrunkDial,ANY_INT_TRUNK,${EXTEN},0,t) ; ;--------------------------------------------------------------------- ;======================= INCOMING CONTEXTS =========================== ;--------------------------------------------------------------------- ; [incoming-sip] ; ;This context is used for incoming calls on SIP channels. ; exten => _16132885759,1,NoOp(${CONTEXT}) exten => _16132885759,n,Answer() exten => _16132885759,n,Ringing() exten => _16132885759,n,Wait(5) exten => _16132885759,n,Dial(SIP/[EMAIL PROTECTED]) exten => _16132885759,n,Hangup() ; ;TODO - Clean this up: ;Play tones for bogus calls: exten => i,1,Answer exten => i,2,Playtones(congestion) exten => i,3,Congestion ; exten => h,1,Hangup ; [incoming-bogus-calls] exten => s,1,NoOp(${CONTEXT}) exten => s,n,Answer exten => s,n,Playtones(congestion) exten => s,n,Congestion ; ;--------------------------------------------------------------------- ;======================= OUTGOING CONTEXTS =========================== ;--------------------------------------------------------------------- [outgoing-PBX] ; ; This is the context used when calls are being made by internal ; devices. By definition these devices are on the same LAN as ; Asterisk so leaving Asterisk in the media path is OK. Use the ; t option as these extensions will want to transfer the call. ; exten => s,1,NoOp(${CONTEXT}) ; ;-------- PBX -------- include => Extensions-PBX ; ;-------- Local -------- include => Local-Calls-PBX ; ;-------- Toll-free -------- include => Toll-Free-Calls-PBX ; ;-------- NA-long-distance-calls -------- include => NA-Long-Distance-Calls-PBX ; ;-------- INT-long-distance-calls -------- include => INT-Long-Distance-Calls-PBX ; include => cleanup -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.411 / Virus Database: 268.17.36/681 - Release Date: 11/02/2007 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users