SOLVED: Custom extension not routing properly

abostick · March 4, 2010, 4:23pm

Hi,

I am using a new install of Freepbx running:
CentOS 5.4
Asterisk 1.6.2.2
Freepbx 2.7

I have a working inbound SIP trunk but would like to route some inbound calls to remote extensions over the same trunk.

I have setup a custom extension 100 with the following dial command:

However, no call is made when trying to reach that extension. In fact, the trunk is not even hit. I turn on AGI debug and set verbose to 3 and get:

pbx*CLI> core set verbose 3
Verbosity was 0 and is now 3
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [8775551234@from-sip-external:1] NoOp("SIP/209.247.17.5-0000003d", "Received incoming SIP connection from unknown peer to 8775551234") in new stack
    -- Executing [8775551234@from-sip-external:2] Set("SIP/209.247.17.5-0000003d", "DID=8775551234") in new stack
    -- Executing [8775551234@from-sip-external:3] Goto("SIP/209.247.17.5-0000003d", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/209.247.17.5-0000003d", "1?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,2)
    -- Executing [s@from-sip-external:2] GotoIf("SIP/209.247.17.5-0000003d", "0?setlanguage:from-trunk,8775551234,1") in new stack
    -- Goto (from-trunk,8775551234,1)
    -- Executing [8775551234@from-trunk:1] Set("SIP/209.247.17.5-0000003d", "__FROM_DID=8775551234") in new stack
    -- Executing [8775551234@from-trunk:2] ExecIf("SIP/209.247.17.5-0000003d", "0 ?Set(CALLERID(name)=+15559876543)") in new stack
    -- Executing [8775551234@from-trunk:3] Set("SIP/209.247.17.5-0000003d", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [8775551234@from-trunk:4] Set("SIP/209.247.17.5-0000003d", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [8775551234@from-trunk:5] Goto("SIP/209.247.17.5-0000003d", "from-did-direct,100,1") in new stack
    -- Goto (from-did-direct,100,1)
    -- Executing [100@from-did-direct:1] Macro("SIP/209.247.17.5-0000003d", "exten-vm,novm,100") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/209.247.17.5-0000003d", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/209.247.17.5-0000003d", "AMPUSER=+15559876543") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/209.247.17.5-0000003d", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/209.247.17.5-0000003d", "1?Set(REALCALLERIDNUM=+15559876543)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/209.247.17.5-0000003d", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/209.247.17.5-0000003d", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/209.247.17.5-0000003d", "1?report") in new stack
    -- Goto (macro-user-callerid,s,9)
    -- Executing [s@macro-user-callerid:9] GotoIf("SIP/209.247.17.5-0000003d", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:10] Set("SIP/209.247.17.5-0000003d", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:11] GotoIf("SIP/209.247.17.5-0000003d", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] NoOp("SIP/209.247.17.5-0000003d", "Using CallerID "COMPANY" <+15559876543>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/209.247.17.5-0000003d", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/209.247.17.5-0000003d", "VMBOX=novm") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/209.247.17.5-0000003d", "EXTTOCALL=100") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/209.247.17.5-0000003d", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/209.247.17.5-0000003d", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/209.247.17.5-0000003d", "RT=""") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/209.247.17.5-0000003d", "record-enable,100,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/209.247.17.5-0000003d", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] ExecIf("SIP/209.247.17.5-0000003d", "0?MacroExit()") in new stack
    -- Executing [s@macro-record-enable:5] GotoIf("SIP/209.247.17.5-0000003d", "0?Group:OUT") in new stack
    -- Goto (macro-record-enable,s,16)
    -- Executing [s@macro-record-enable:16] GotoIf("SIP/209.247.17.5-0000003d", "1?IN") in new stack
    -- Goto (macro-record-enable,s,21)
    -- Executing [s@macro-record-enable:21] ExecIf("SIP/209.247.17.5-0000003d", "1?MacroExit()") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/209.247.17.5-0000003d", "dial,"",tr,100") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/209.247.17.5-0000003d", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/209.247.17.5-0000003d", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_request: dialparties.agi
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_channel: SIP/209.247.17.5-0000003d
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_language: en
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_type: SIP
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_uniqueid: 1267719205.87
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_version: 1.6.2.2
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_callerid: +15559876543
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_calleridname: COMPANY
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_callingpres: 0
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_callingani2: 0
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_callington: 0
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_callingtns: 0
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_dnid: 8775551234
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_rdnis: unknown
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_context: macro-dial
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_extension: s
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_priority: 3
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_enhanced: 0.0
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_accountcode:
<SIP/209.247.17.5-0000003d>AGI Tx >> agi_threadid: -1238369392
<SIP/209.247.17.5-0000003d>AGI Tx >>
    -- <SIP/209.247.17.5-0000003d>AGI Script dialparties.agi completed, returning -1

Are custom extensions working out of the box with 2.7 and asterisk 1.6? Anyone else have this up and running? I have tried to get some debug output from the dialparties.agi itself with no luck.

Thank you,
Aaron

abostick · March 4, 2010, 7:41pm

Come to find out, this works fine out of the box. My SIP provider gave me a peer name that was not resovable via DNS, so of course out bound calls wouldn’t when you cannot contact the peer.

Once I put in the ip address of the peer instead of the DNS name, it worked fine.

Frustrating but glad I figured it out.

Aaron