Re: [Asterisk-java-users] Inbound calls get cancelled - 'Subscriber Absent' warning

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Hi,

this question is more related to freepbx than asterisk-java... but anyway:

it looks like your inbound route tries to make a call to your extension 
1001 as a SIP call, not as an AGI call...
check your freepbx-settings... the inbound route shout not point to an 
extension, but to a custom destination instead.
you have to declare the custom destination via freepbx webfrontend and 
afterward write this custom destination
in the appropriate config file where in turn you call your java-AGI 
(like the way you did in your context written in your question)..

regards,
yves

Am 25.03.2013 03:12, schrieb Salman Jamali:
> Hi,
>
> I am having difficulty in configuring an inbound route for asterisk-java:
> callcentric sip did -> freepbx distro/asterisk -> asterisk-java (this 
> fails)
> callcentric sip did -> freepbx distro/asterisk -> X-Lite (works smoothly)
>
> My configuration is FreePBX Distro 2.210 w/ Asterisk 1.8.20 on CentOS. 
> I successfully configured my callcentric sip trunk, an extension, and 
> an inbound route. I can receive calls on X-lite. Now, I am trying to 
> work my way through the Asterisk-java FastAGI tutorial to be able to 
> handle inbound calls in my subclass of BaseAgiScript. But before my 
> java code gets invoked, asterisk aborts the call. Following is the 
> relevant information (please ask for anything else if required):
> --------------------------------------------------------------------------------------------------------------------------------------------
> Asterisk-java Log: (no activity here)
>
> prompt$ java -jar asterisk-java.jar
> Mar 24, 2013 9:05:58 PM org.asteriskjava.fastagi.DefaultAgiServer startup
> INFO: Listening on *:4573.
> ..
> --------------------------------------------------------------------------------------------------------------------------------------------
> Asterisk Log:
>
>     -- Executing [s@macro-dial-one:33] 
> ExecIf("SIP/callcentric-00000007", "0?Set(CHANNEL(musicclass)=)") in 
> new stack
>     -- Executing [s@macro-dial-one:34] 
> GosubIf("SIP/callcentric-00000007", "0?qwait,1()") in new stack
>     -- Executing [s@macro-dial-one:35] Set("SIP/callcentric-00000007", 
> "__CWIGNORE=") in new stack
>     -- Executing [s@macro-dial-one:36] Set("SIP/callcentric-00000007", 
> "__KEEPCID=TRUE") in new stack
>     -- Executing [s@macro-dial-one:37] 
> GotoIf("SIP/callcentric-00000007", "0?usegoto,1") in new stack
>     -- Executing [s@macro-dial-one:38] 
> GotoIf("SIP/callcentric-00000007", "1?godial") in new stack
>     -- Goto (macro-dial-one,s,42)
>     -- Executing [s@macro-dial-one:42] 
> Dial("SIP/callcentric-00000007", "SIP/1001,,tr") in new stack
> [2013-03-24 21:06:19] WARNING[3195]: app_dial.c:2345 dial_exec_full: 
> Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
>   == Everyone is busy/congested at this time (1:0/0/1)
>     -- Executing [s@macro-dial-one:43] 
> ExecIf("SIP/callcentric-00000007", "0?Set(DIALSTATUS=)") in new stack
>     -- Executing [s@macro-dial-one:44] 
> GosubIf("SIP/callcentric-00000007", "0?s-CHANUNAVAIL,1()") in new stack
>     -- Executing [s@macro-dial-one:45] 
> MacroExit("SIP/callcentric-00000007", "") in new stack
>     -- Executing [s@macro-exten-vm:15] Set("SIP/callcentric-00000007", 
> "SV_DIALSTATUS=CHANUNAVAIL") in new stack
>     -- Executing [s@macro-exten-vm:16] 
> GosubIf("SIP/callcentric-00000007", "0?docfu,1()") in new stack
>     -- Executing [s@macro-exten-vm:17] 
> GosubIf("SIP/callcentric-00000007", "0?docfb,1()") in new stack
>     -- Executing [s@macro-exten-vm:18] Set("SIP/callcentric-00000007", 
> "DIALSTATUS=CHANUNAVAIL") in new stack
>     -- Executing [s@macro-exten-vm:19] 
> ExecIf("SIP/callcentric-00000007", "0?MacroExit()") in new stack
>     -- Executing [s@macro-exten-vm:20] 
> GotoIf("SIP/callcentric-00000007", "1?s-CHANUNAVAIL,1") in new stack
>     -- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
>     -- Executing [s-CHANUNAVAIL@macro-exten-vm:1] 
> GotoIf("SIP/callcentric-00000007", "0?exit,1") in new stack
>     -- Executing [s-CHANUNAVAIL@macro-exten-vm:2] 
> PlayTones("SIP/callcentric-00000007", "congestion") in new stack
>     -- Executing [s-CHANUNAVAIL@macro-exten-vm:3] 
> Congestion("SIP/callcentric-00000007", "10") in new stack
>   == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 3) exited 
> non-zero on 'SIP/callcentric-00000007' in macro 'exten-vm'
>   == Spawn extension (from-did-direct, 1001, 2) exited non-zero on 
> 'SIP/callcentric-00000007'
>     -- Executing [h@from-did-direct:1] 
> Macro("SIP/callcentric-00000007", "hangupcall,") in new stack
>     -- Executing [s@macro-hangupcall:1] 
> GotoIf("SIP/callcentric-00000007", "1?theend") in new stack
>     -- Goto (macro-hangupcall,s,3)
>     -- Executing [s@macro-hangupcall:3] 
> ExecIf("SIP/callcentric-00000007", "0?Set(CDR(recordingfile)=)") in 
> new stack
>     -- Executing [s@macro-hangupcall:4] 
> Hangup("SIP/callcentric-00000007", "") in new stack
>   == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
> 'SIP/callcentric-00000007' in macro 'hangupcall'
>   == Spawn extension (from-did-direct, h, 1) exited non-zero on 
> 'SIP/callcentric-00000007'[/code][/size]
> --------------------------------------------------------------------------------------------------------------------------------------------
> extensions_custom.conf (contexts tried: default, from-pstn, from-trunk 
> .. none worked)
>
> [from-pstn-custom]
> exten => 1001,1,AGI(agi://192.168.1.106:4573/hello.agi)[/code] 
> <http://192.168.1.106:4573/hello.agi%29[/code]>
> --------------------------------------------------------------------------------------------------------------------------------------------
> SIP trunk peer details (From Callcentric's configuration instructions 
> for FreePBX)
>
> context=from-pstn
> fromdomain=callcentric.com <http://callcentric.com>
> fromuser=177723XXXXX
> host=callcentric.com <http://callcentric.com>
> insecure=port,invite
> secret=password
> type=peer
> defaultuser=177723XXXXX
> disallowed_methods=UPDATE
> directmedia=no
> videosupport=no
> disallow=all
> allow=ulaw
> --------------------------------------------------------------------------------------------------------------------------------------------
> sip_general_custom.conf (From Callcentric's configuration instructions 
> for FreePBX)
>
> context=from-pstn
> srvlookup=yes
> session-timers=refuse
> session-expires=180
> session-minse=90
> session-refresher=uas
> --------------------------------------------------------------------------------------------------------------------------------------------
> SIP detailed logs:
>
> localhost*CLI> sip set debug on
> SIP Debugging enabled
> Really destroying SIP dialog '05b1a0c008559d24301bde7a0879ed4f@[::1]' 
> Method: REGISTER
>
> <--- SIP read from UDP:204.11.192.161:5060 
> <http://204.11.192.161:5060> --->
> INVITE sip:17772380367@71.62.43.250:5060 
> <http://sip:17772380367@71.62.43.250:5060> SIP/2.0
> v: SIP/2.0/UDP 
> 204.11.192.161:5060;branch=z9hG4bK-065a4117c11d04df12a8cf3dafa2af77
> f: <sip:16617480240@66.193.176.35 
> <mailto:sip%3A16617480240@66.193.176.35>>;tag=3573164592-373269
> t: <sip:163...@ss... 
> <mailto:sip%3A1...@ss...>>
> i: 238...@ms... 
> <mailto:238...@ms...>
> CSeq: 1 INVITE
> Max-Forwards: 8
> m: 
> <sip:6e1ae09b353f7d72cc21a7cf0cd89f6e@204.11.192.161:5060;transport=udp>
> Supported: timer
> c: application/sdp
> l: 350
>
> v=0
> o=NexTone-MSW 2147483647 2147483647 IN IP4 204.11.192.161
> s=sip call
> c=IN IP4 204.11.192.161
> t=0 0
> m=audio 54366 RTP/AVP 18 0 8 101
> a=fmtp:18 annexb=no
> a=fmtp:101 0-15
> a=rtpmap:101 telephone-event/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=ptime:20
> a=sendrecv
> a=silenceSupp:off - - - -
> a=setup:actpass
> <------------->
> --- (11 headers 16 lines) ---
> Sending to 204.11.192.161:5060 <http://204.11.192.161:5060> (NAT)
> Using INVITE request as basis request - 
> 238...@ms... 
> <mailto:238...@ms...>
> No matching peer for '16617480240' from '204.11.192.161:5060 
> <http://204.11.192.161:5060>'
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
> Found RTP audio format 18
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 101
> Found audio description format telephone-event for ID 101
> Found audio description format PCMA for ID 8
> Found audio description format PCMU for ID 0
> Found audio description format G729 for ID 18
> Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c 
> (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 
> 0xc (ulaw|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 
> (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port 204.11.192.161:54366 
> <http://204.11.192.161:54366>
> Looking for 17772380367 in from-pstn (domain 71.62.43.250)
> list_route: hop: 
> <sip:6e1ae09b353f7d72cc21a7cf0cd89f6e@204.11.192.161:5060;transport=udp>
>
> <--- Transmitting (NAT) to 204.11.192.161:5060 
> <http://204.11.192.161:5060> --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 
> 204.11.192.161:5060;branch=z9hG4bK-065a4117c11d04df12a8cf3dafa2af77;received=204.11.192.161;rport=5060
> From: <sip:16617480240@66.193.176.35 
> <mailto:sip%3A16617480240@66.193.176.35>>;tag=3573164592-373269
> To: <sip:163...@ss... 
> <mailto:sip%3A1...@ss...>>
> Call-ID: 238...@ms... 
> <mailto:238...@ms...>
> CSeq: 1 INVITE
> Server: FPBX-2.10.1(1.8.20.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
> INFO, PUBLISH
> Supported: replaces
> Contact: <sip:17772380367@71.62.43.250:5060 
> <http://sip:17772380367@71.62.43.250:5060>>
> Content-Length: 0
>
> <--- Reliably Transmitting (NAT) to 204.11.192.161:5060 
> <http://204.11.192.161:5060> --->
> SIP/2.0 503 Service Unavailable
> Via: SIP/2.0/UDP 
> 204.11.192.161:5060;branch=z9hG4bK-065a4117c11d04df12a8cf3dafa2af77;received=204.11.192.161;rport=5060
> From: <sip:16617480240@66.193.176.35 
> <mailto:sip%3A16617480240@66.193.176.35>>;tag=3573164592-373269
> To: <sip:163...@ss... 
> <mailto:sip%3A1...@ss...>>;tag=as5f64e106
> Call-ID: 238...@ms... 
> <mailto:238...@ms...>
> CSeq: 1 INVITE
> Server: FPBX-2.10.1(1.8.20.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
> INFO, PUBLISH
> Supported: replaces
> X-Asterisk-HangupCause: Subscriber absent
> X-Asterisk-HangupCauseCode: 20
> Content-Length: 0
> <------------>
> --------------------------------------------------------------------------------------------------------------------------------------------
> Any help will be highly appreciated, thanks!
>
>
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