Thread: [Asterisk-java-users] Inbound calls get cancelled - 'Subscriber Absent' warning
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From: Salman J. <sal...@gm...> - 2013-03-25 02:12:42
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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]
--------------------------------------------------------------------------------------------------------------------------------------------
SIP trunk peer details (From Callcentric's configuration instructions for
FreePBX)
context=from-pstn
fromdomain=callcentric.com
fromuser=177723XXXXX
host=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 --->
INVITE 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>;tag=3573164592-373269
t: <sip:163...@ss...>
i: 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 (NAT)
Using INVITE request as basis request -
238...@ms...
No matching peer for '16617480240' from '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
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 --->
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>;tag=3573164592-373269
To: <sip:163...@ss...>
Call-ID: 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>
Content-Length: 0
<--- Reliably Transmitting (NAT) to 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>;tag=3573164592-373269
To: <sip:163...@ss...>;tag=as5f64e106
Call-ID: 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!
|
|
From: Yves A. <yv...@gm...> - 2013-03-25 08:41:37
|
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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> Download AppDynamics Lite for free today:
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> _______________________________________________
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> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
|
|
From: Salman J. <sal...@gm...> - 2013-03-25 13:50:26
|
Thanks a lot for clear explanation, it worked. Just want to share what I
did with the community:
In extensions_custom.conf:
[from-internal-custom]
include => invoke-fastagi
[invoke-fastagi]
exten => 1001,1,AGI(agi://192.168.1.106/hello.agi)
In FreePBX UI, created a Custom Destination:
invoke-fastagi,1001,1
Reconfigured my Inbound Route to point to the created custom destination
instead of 1001 Extension.
Called from my phone, and my Java program executed smoothly!
Thanks,
Salman
On Mon, Mar 25, 2013 at 4:41 AM, Yves A. <yv...@gm...> wrote:
> 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%5B/code%5D>
>
> --------------------------------------------------------------------------------------------------------------------------------------------
> SIP trunk peer details (From Callcentric's configuration instructions for
> FreePBX)
>
> context=from-pstn
> fromdomain=callcentric.com
> fromuser=177723XXXXX
> host=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 --->
> INVITE 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>;tag=3573164592-373269
> t: <sip:163...@ss...>
> i: 238...@ms...
> CSeq: 1 INVITE
> Max-Forwards: 8
> m:
> <sip:6e1ae09b353f7d72cc21a7cf0cd89f6e@204.11.192.161:5060;transport=udp><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 (NAT)
> Using INVITE request as basis request -
> 238...@ms...
> No matching peer for '16617480240' from '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
> Looking for 17772380367 in from-pstn (domain 71.62.43.250)
> list_route: hop:
> <sip:6e1ae09b353f7d72cc21a7cf0cd89f6e@204.11.192.161:5060;transport=udp><sip:6e1ae09b353f7d72cc21a7cf0cd89f6e@204.11.192.161:5060;transport=udp>
>
> <--- Transmitting (NAT) to 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>;tag=3573164592-373269
> To: <sip:163...@ss...>
> Call-ID: 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>
> Content-Length: 0
>
> <--- Reliably Transmitting (NAT) to 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>;tag=3573164592-373269
> To: <sip:163...@ss...>;tag=as5f64e106
> Call-ID: 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!
>
>
> ------------------------------------------------------------------------------
> Everyone hates slow websites. So do we.
> Make your web apps faster with AppDynamics
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>
>
>
> _______________________________________________
> Asterisk-java-users mailing lis...@li...https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
>
>
> ------------------------------------------------------------------------------
> Everyone hates slow websites. So do we.
> Make your web apps faster with AppDynamics
> Download AppDynamics Lite for free today:
> http://p.sf.net/sfu/appdyn_d2d_mar
> _______________________________________________
> Asterisk-java-users mailing list
> Ast...@li...
> https://lists.sourceforge.net/lists/listinfo/asterisk-java-users
>
>
--
-salman
The Messenger of Allah (sallallahu alaihi wa-sallam) said: *“The people
most severely tested are the Prophets, then the righteous, then the next
best and the next best. A man will be tested in accordance with the degree
of his religious commitment; the stronger his religious commitment, the
stronger his test.”* Tuhfat al-Ahwadhi (7:78)
|
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