[asterisk-users] Bad hangup event cause
Francisco Seratti
fseratti at yahoo.com.ar
Mon Aug 27 00:59:40 CDT 2007
Hello, i have a problem with the hangup cause received from the AMI in the Hangup events. All calls that arent answered after ringing are returning hangup cause 16 (normal clearing) instead 19.
Im running asterisk 1.4.11, the calls are generated to a SIP peer using the AMI originate command.
This is the 'sip debug' output:
Reliably Transmitting (no NAT) to 192.168.0.70:5060:
INVITE sip:1 at 192.168.0.70:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport
From: "123" <sip:123 at 192.168.0.1>;tag=as0cd1aab0
To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>
Contact: <sip:123 at 192.168.0.1>
Call-ID: 3daa9e730e767bf932a9196a35200e36 at 192.168.0.1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 27 Aug 2007 05:53:34 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 21676 21676 IN IP4 192.168.0.1
s=session
c=IN IP4 192.168.0.1
t=0 0
m=audio 15274 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
gw*CLI>
<--- SIP read from 192.168.0.70:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport
From: "123" <sip:123 at 192.168.0.1>;tag=as0cd1aab0
To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 3daa9e730e767bf932a9196a35200e36 at 192.168.0.1
CSeq: 102 INVITE
Server: Cisco ATA 186 v3.2.1 atasip (050616A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
gw*CLI>
<--- SIP read from 192.168.0.70:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport
From: "123" <sip:123 at 192.168.0.1>;tag=as0cd1aab0
To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 3daa9e730e767bf932a9196a35200e36 at 192.168.0.1
CSeq: 102 INVITE
Contact: 1 <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>
Server: Cisco ATA 186 v3.2.1 atasip (050616A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Reliably Transmitting (no NAT) to 192.168.0.70:5060:
OPTIONS sip:2 at 192.168.0.70:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK49966ec7;rport
From: "asterisk" <sip:asterisk at 192.168.0.1>;tag=as0916f4ed
To: <sip:2 at 192.168.0.70:5060;user=phone;transport=udp>
Contact: <sip:asterisk at 192.168.0.1>
Call-ID: 04d899b45a7c51130dea88261b4db31a at 192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 27 Aug 2007 05:53:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
gw*CLI>
<--- SIP read from 192.168.0.70:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK49966ec7;rport
From: "asterisk" <sip:asterisk at 192.168.0.1>;tag=as0916f4ed
To: <sip:2 at 192.168.0.70:5060;user=phone;transport=udp>;tag=3724167432
Call-ID: 04d899b45a7c51130dea88261b4db31a at 192.168.0.1
CSeq: 102 OPTIONS
Server: Cisco ATA 186 v3.2.1 atasip (050616A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 250
Content-Type: application/sdp
v=0
o=2 19680158 19680158 IN IP4 192.168.0.70
s=ATA186 Call
c=IN IP4 192.168.0.70
t=0 0
m=audio 16386 RTP/AVP 0 8 4 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 11 lines) ---
Really destroying SIP dialog '04d899b45a7c51130dea88261b4db31a at 192.168.0.1' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.0.70:5060:
OPTIONS sip:1 at 192.168.0.70:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK02be2790;rport
From: "asterisk" <sip:asterisk at 192.168.0.1>;tag=as6ba5f9aa
To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>
Contact: <sip:asterisk at 192.168.0.1>
Call-ID: 1d2fdf042629f7ad54790ccc1002d60f at 192.168.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 27 Aug 2007 05:53:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
gw*CLI>
<--- SIP read from 192.168.0.70:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK02be2790;rport
From: "asterisk" <sip:asterisk at 192.168.0.1>;tag=as6ba5f9aa
To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 1d2fdf042629f7ad54790ccc1002d60f at 192.168.0.1
CSeq: 102 OPTIONS
Server: Cisco ATA 186 v3.2.1 atasip (050616A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Supported: replaces
Content-Length: 250
Content-Type: application/sdp
v=0
o=1 19680166 19680166 IN IP4 192.168.0.70
s=ATA186 Call
c=IN IP4 192.168.0.70
t=0 0
m=audio 16384 RTP/AVP 0 8 4 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (11 headers 11 lines) ---
Really destroying SIP dialog '1d2fdf042629f7ad54790ccc1002d60f at 192.168.0.1' Method: OPTIONS
Scheduling destruction of SIP dialog '3daa9e730e767bf932a9196a35200e36 at 192.168.0.1' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.0.70:5060:
CANCEL sip:1 at 192.168.0.70:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport
From: "123" <sip:123 at 192.168.0.1>;tag=as0cd1aab0
To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>
Call-ID: 3daa9e730e767bf932a9196a35200e36 at 192.168.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Scheduling destruction of SIP dialog '3daa9e730e767bf932a9196a35200e36 at 192.168.0.1' in 6400 ms (Method: INVITE)
[Aug 27 02:53:44] NOTICE[21820]: pbx_spool.c:341 attempt_thread: Call failed to go through, reason (3) Remote end Ringing
gw*CLI>
<--- SIP read from 192.168.0.70:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport
From: "123" <sip:123 at 192.168.0.1>;tag=as0cd1aab0
To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 3daa9e730e767bf932a9196a35200e36 at 192.168.0.1
CSeq: 102 CANCEL
Server: Cisco ATA 186 v3.2.1 atasip (050616A)
Supported: replaces
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
gw*CLI>
<--- SIP read from 192.168.0.70:5060 --->
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport
From: "123" <sip:123 at 192.168.0.1>;tag=as0cd1aab0
To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 3daa9e730e767bf932a9196a35200e36 at 192.168.0.1
CSeq: 102 INVITE
Server: Cisco ATA 186 v3.2.1 atasip (050616A)
Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 192.168.0.70:5060:
ACK sip:1 at 192.168.0.70:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport
From: "123" <sip:123 at 192.168.0.1>;tag=as0cd1aab0
To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Contact: <sip:123 at 192.168.0.1>
Call-ID: 3daa9e730e767bf932a9196a35200e36 at 192.168.0.1
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
These are the events received from the AMI:
Event: Newchannel
Privilege: call,all
Timestamp: 1188194254.782040
Channel: SIP/1-081d3ba0
State: Down
CallerIDNum: <unknown>
CallerIDName: <unknown>
Uniqueid: 1188194254.9
Event: Newcallerid
Privilege: call,all
Timestamp: 1188194254.782548
Channel: SIP/1-081d3ba0
CallerID: 123
CallerIDName: 123
Uniqueid: 1188194254.9
CID-CallingPres: 0 (Presentation Allowed, Not Screened)
Event: Newcallerid
Privilege: call,all
Timestamp: 1188194254.782694
Channel: SIP/1-081d3ba0
CallerID: 123
CallerIDName: 123
Uniqueid: 1188194254.9
CID-CallingPres: 0 (Presentation Allowed, Not Screened)
Event: Newstate
Privilege: call,all
Timestamp: 1188194254.811535
Channel: SIP/1-081d3ba0
State: Ringing
CallerID: 123
CallerIDName: 123
Uniqueid: 1188194254.9
Event: Hangup
Privilege: call,all
Timestamp: 1188194264.781755
Channel: SIP/1-081d3ba0
Uniqueid: 1188194254.9
Cause: 16
Cause-txt: Normal Clearing
Thanks in advance, Francisco.
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