ROuting anonymous calls

Hi

We have a Scenario where in we have two asterisk Servers , Asterisk1 and Asterisk 2 ,
When a call comes from outside (from anywhere) to asterisk1- We want Asterisk1 to simply forward the call to Asterisk2 which then would handle the call as per the dialplans it has .

We have setup a Trunk at Asterisk1 which will direct the calls to Asterisk2 , however it does not seem to work .
SIP/PbxUS is the Trunk on Asterisk1 pointing to Asterisk2
Please Help ?

Below is the Asterisk call log :

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [600@from-sip-external:1] NoOp(“SIP/172.16.0.97-00000027”, “Received incoming SIP connection from unknown peer to 600”) in new stack
– Executing [600@from-sip-external:2] Set(“SIP/172.16.0.97-00000027”, “DID=600”) in new stack
– Executing [600@from-sip-external:3] Goto(“SIP/172.16.0.97-00000027”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“SIP/172.16.0.97-00000027”, “1?checklang:noanonymous”) in new stack
– Goto (from-sip-external,s,2)
– Executing [s@from-sip-external:2] GotoIf(“SIP/172.16.0.97-00000027”, “0?setlanguage:from-trunk,600,1”) in new stack
– Goto (from-trunk,600,1)
– Executing [600@from-trunk:1] NoOp(“SIP/172.16.0.97-00000027”, “Catch-All DID Match - Found 600 - You probably want a DID for this.”) in new stack
– Executing [600@from-trunk:2] Goto(“SIP/172.16.0.97-00000027”, “ext-did,s,1”) in new stack
– Goto (ext-did,s,1)
– Executing [s@ext-did:1] ExecIf(“SIP/172.16.0.97-00000027”, “1?Set(__FROM_DID=s)”) in new stack
– Executing [s@ext-did:2] Gosub(“SIP/172.16.0.97-00000027”, “app-blacklist-check,s,1()”) in new stack
– Executing [s@app-blacklist-check:1] GotoIf(“SIP/172.16.0.97-00000027”, “0?blacklisted”) in new stack
– Executing [s@app-blacklist-check:2] Set(“SIP/172.16.0.97-00000027”, “CALLED_BLACKLIST=1”) in new stack
– Executing [s@app-blacklist-check:3] Return(“SIP/172.16.0.97-00000027”, “”) in new stack
– Executing [s@ext-did:3] Set(“SIP/172.16.0.97-00000027”, “CDR(did)=s”) in new stack
– Executing [s@ext-did:4] ExecIf(“SIP/172.16.0.97-00000027”, “1 ?Set(CALLERID(name)=601)”) in new stack
– Executing [s@ext-did:5] Set(“SIP/172.16.0.97-00000027”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [s@ext-did:6] Set(“SIP/172.16.0.97-00000027”, “CALLERPRES()=allowed_not_screened”) in new stack
– Executing [s@ext-did:7] Goto(“SIP/172.16.0.97-00000027”, “ext-trunk,2,1”) in new stack
– Goto (ext-trunk,2,1)
– Executing [2@ext-trunk:1] Set(“SIP/172.16.0.97-00000027”, “TDIAL_STRING=SIP/PbxUS”) in new stack
– Executing [2@ext-trunk:2] Set(“SIP/172.16.0.97-00000027”, “DIAL_TRUNK=2”) in new stack
– Executing [2@ext-trunk:3] Goto(“SIP/172.16.0.97-00000027”, “ext-trunk,tdial,1”) in new stack
– Goto (ext-trunk,tdial,1)
– Executing [tdial@ext-trunk:1] Set(“SIP/172.16.0.97-00000027”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [tdial@ext-trunk:2] GotoIf(“SIP/172.16.0.97-00000027”, “1?nomax”) in new stack
– Goto (ext-trunk,tdial,4)
– Executing [tdial@ext-trunk:4] ExecIf(“SIP/172.16.0.97-00000027”, “1?Set(CALLERPRES()=allowed_not_screened)”) in new stack
– Executing [tdial@ext-trunk:5] Set(“SIP/172.16.0.97-00000027”, “DIAL_NUMBER=s”) in new stack
– Executing [tdial@ext-trunk:6] GosubIf(“SIP/172.16.0.97-00000027”, “0?sub-flp-2,s,1()”) in new stack
– Executing [tdial@ext-trunk:7] Set(“SIP/172.16.0.97-00000027”, “OUTNUM=s”) in new stack
– Executing [tdial@ext-trunk:8] Dial(“SIP/172.16.0.97-00000027”, “SIP/PbxUS/s,300,”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/PbxUS/s
[2012-09-05 20:08:55] NOTICE[3349]: chan_sip.c:20233 handle_response_invite: Failed to authenticate on INVITE to ‘“601” sip:601@172.16.0.97;tag=as378cec9d’
– SIP/PbxUS-00000028 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [tdial@ext-trunk:9] Set(“SIP/172.16.0.97-00000027”, “CALLERID(number)=601”) in new stack
– Executing [tdial@ext-trunk:10] Set(“SIP/172.16.0.97-00000027”, “CALLERID(name)=601”) in new stack
– Executing [tdial@ext-trunk:11] Hangup(“SIP/172.16.0.97-00000027”, “”) in new stack
== Spawn extension (ext-trunk, tdial, 11) exited non-zero on ‘SIP/172.16.0.97-00000027’

You have a problem with your sip.conf’s. You didn’t provide copies of them.

Also, “trunks” are GUI concepts and this looks like a GUI dialplan; are you sure you shouldn’t be asking on a forum for the Asterisk GUI.

Also, it looks like the first Asterisk has allowguest set to yes. This is normally a bad thing from a security point of view.