Asterisk Information


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@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@192.168.0.1>;tag=as0cd1aab0
To: <sip:1@192.168.0.70:5060;user=phone;transport=udp>
Contact: <sip:123@192.168.0.1>
Call-ID: 3daa9e730e767bf932a9196a35200e36@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@192.168.0.1>;tag=as0cd1aab0
To: <sip:1@192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 3daa9e730e767bf932a9196a35200e36@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@192.168.0.1>;tag=as0cd1aab0
To: <sip:1@192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 3daa9e730e767bf932a9196a35200e36@192.168.0.1
CSeq: 102 INVITE
Contact: 1 <sip:1@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@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@192.168.0.1>;tag=as0916f4ed
To: <sip:2@192.168.0.70:5060;user=phone;transport=udp>
Contact: <sip:asterisk@192.168.0.1>
Call-ID: 04d899b45a7c51130dea88261b4db31a@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@192.168.0.1>;tag=as0916f4ed
To: <sip:2@192.168.0.70:5060;user=phone;transport=udp>;tag=3724167432
Call-ID: 04d899b45a7c51130dea88261b4db31a@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@192.168.0.1' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.0.70:5060:
OPTIONS sip:1@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@192.168.0.1>;tag=as6ba5f9aa
To: <sip:1@192.168.0.70:5060;user=phone;transport=udp>
Contact: <sip:asterisk@192.168.0.1>
Call-ID: 1d2fdf042629f7ad54790ccc1002d60f@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@192.168.0.1>;tag=as6ba5f9aa
To: <sip:1@192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 1d2fdf042629f7ad54790ccc1002d60f@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@192.168.0.1' Method: OPTIONS
Scheduling destruction of SIP dialog '3daa9e730e767bf932a9196a35200e36@192.168.0.1' in 6400 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 192.168.0.70:5060:
CANCEL sip:1@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@192.168.0.1>;tag=as0cd1aab0
To: <sip:1@192.168.0.70:5060;user=phone;transport=udp>
Call-ID: 3daa9e730e767bf932a9196a35200e36@192.168.0.1
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
Scheduling destruction of SIP dialog '3daa9e730e767bf932a9196a35200e36@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@192.168.0.1>;tag=as0cd1aab0
To: <sip:1@192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 3daa9e730e767bf932a9196a35200e36@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@192.168.0.1>;tag=as0cd1aab0
To: <sip:1@192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Call-ID: 3daa9e730e767bf932a9196a35200e36@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@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@192.168.0.1>;tag=as0cd1aab0
To: <sip:1@192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099
Contact: <sip:123@192.168.0.1>
Call-ID: 3daa9e730e767bf932a9196a35200e36@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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Asterisk Information

Sunday, August 26, 2007

[asterisk-users] Bad hangup event cause

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