[asterisk-bugs] [Asterisk 0014741]: Timed out parked calls always return to originating extension
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Mar 30 19:29:30 CDT 2009
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=14741
======================================================================
Reported By: Herb
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 14741
Category: Resources/res_features
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.4.24
Regression: No
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2009-03-24 21:11 CDT
Last Modified: 2009-03-30 19:29 CDT
======================================================================
Summary: Timed out parked calls always return to originating
extension
Description:
I have been trying to get this to work on versions 1.4.22.2 and 1.4.24 and
no matter what I do, timed out parked calls always ring back the
originating extension.
However, I have downloaded and installed 1.6.1-rc3 and this feature works
correctly. I would like to stick with version 1.4 if possible due to
stability (for the most part :) ) but will just migrate if I have to, since
this is an required feature for my installation.
I have included parts of my features.conf file.
(features.conf)
[general]
comebacktoorigin = no
{extensions.conf}
[parkedcallstimeout]
exten => s,1,NoOp(user was parked on parkingslot #${PARKINGSLOT})
exten => s,2,Playback(tt-monkeys)
======================================================================
----------------------------------------------------------------------
(0102440) Herb (reporter) - 2009-03-30 19:29
http://bugs.digium.com/view.php?id=14741#c102440
----------------------------------------------------------------------
Here is the output from version 1.4.24 which always sends the timed out
parked calls back to the originating extension.
Scheduling destruction of SIP dialog
'102963404eb9d49051ff29681a161938 at 192.168.2.100' in 32000 ms (Method:
INVITE)
set_destination: Parsing <sip:541 at 192.168.2.3:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.2.3, port 5060
Reliably Transmitting (no NAT) to 192.168.2.3:5060:
BYE sip:541 at 192.168.2.3:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK41fbdcfd;rport
From: "Herb" <sip:143 at 192.168.2.100>;tag=as50162f53
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=414947709
Call-ID: 102963404eb9d49051ff29681a161938 at 192.168.2.100
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
kohana*CLI>
<--- SIP read from 192.168.2.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.100:5060;branch=z9hG4bK41fbdcfd;rport=5060;received=192.168.2.100
From: "Herb" <sip:143 at 192.168.2.100>;tag=as50162f53
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=414947709
Call-ID: 102963404eb9d49051ff29681a161938 at 192.168.2.100
CSeq: 103 BYE
Server: Aastra 9480iCT/2.5.0.82
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog
'102963404eb9d49051ff29681a161938 at 192.168.2.100' Method: INVITE
kohana*CLI>
kohana*CLI>
kohana*CLI>
kohana*CLI>
kohana*CLI> on hold
No such command 'on hold' (type 'help on hold' for other possible
commands)
Audio is at 192.168.2.100 port 19778
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.3:5060:
INVITE sip:541 at 192.168.2.3:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK5cce3665;rport
From: "Herb" <sip:143 at 192.168.2.100>;tag=as398e24c9
To: <sip:541 at 192.168.2.3:5060;transport=udp>
Contact: <sip:143 at 192.168.2.100>
Call-ID: 515f077817bb964c42ad4f8666ec4faa at 192.168.2.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 31 Mar 2009 00:16:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 289
v=0
o=root 23665 23665 IN IP4 192.168.2.100
s=session
c=IN IP4 192.168.2.100
t=0 0
m=audio 19778 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
kohana*CLI>
<--- SIP read from 192.168.2.3:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.2.100:5060;branch=z9hG4bK5cce3665;rport=5060;received=192.168.2.100
From: "Herb" <sip:143 at 192.168.2.100>;tag=as398e24c9
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=2301220826
Call-ID: 515f077817bb964c42ad4f8666ec4faa at 192.168.2.100
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "Cordless @ 541"
<sip:541 at 192.168.2.3:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D10BD9C>"
Server: Aastra 9480iCT/2.5.0.82
Supported: gruu, path
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
kohana*CLI>
<--- SIP read from 192.168.2.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.100:5060;branch=z9hG4bK5cce3665;rport=5060;received=192.168.2.100
From: "Herb" <sip:143 at 192.168.2.100>;tag=as398e24c9
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=2301220826
Call-ID: 515f077817bb964c42ad4f8666ec4faa at 192.168.2.100
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "Cordless @ 541"
<sip:541 at 192.168.2.3:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D10BD9C>"
Server: Aastra 9480iCT/2.5.0.82
Supported: gruu, path, timer, replaces
Content-Type: application/sdp
Content-Length: 255
v=0
o=MxSIP 0 0 IN IP4 192.168.2.3
s=SIP Call
c=IN IP4 192.168.2.3
t=0 0
m=audio 3000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 13 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.3:3000
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc
(ulaw|alaw)/video=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 192.168.2.3:3000
list_route: hop: <sip:541 at 192.168.2.3:5060;transport=udp>
set_destination: Parsing <sip:541 at 192.168.2.3:5060;transport=udp> for
address/port to send to
set_destination: set destination to 192.168.2.3, port 5060
Transmitting (no NAT) to 192.168.2.3:5060:
ACK sip:541 at 192.168.2.3:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK53db65ee;rport
From: "Herb" <sip:143 at 192.168.2.100>;tag=as398e24c9
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=2301220826
Contact: <sip:143 at 192.168.2.100>
Call-ID: 515f077817bb964c42ad4f8666ec4faa at 192.168.2.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
kohana*CLI>
<--- SIP read from 192.168.2.3:5060 --->
BYE sip:8088853143 at 192.168.2.100 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.3:5060;branch=z9hG4bK4adc9820f774cdeda.3ebd948dac25b7cbd
Max-Forwards: 70
From: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=2301220826
To: "Herb" <sip:143 at 192.168.2.100>;tag=as398e24c9
Call-ID: 515f077817bb964c42ad4f8666ec4faa at 192.168.2.100
CSeq: 25787 BYE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Supported: gruu, path, timer
User-Agent: Aastra 9480iCT/2.5.0.82
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.2.3 : 5060 (no NAT)
kohana*CLI>
<--- Transmitting (no NAT) to 192.168.2.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.3:5060;branch=z9hG4bK4adc9820f774cdeda.3ebd948dac25b7cbd;received=192.168.2.3
From: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=2301220826
To: "Herb" <sip:143 at 192.168.2.100>;tag=as398e24c9
Call-ID: 515f077817bb964c42ad4f8666ec4faa at 192.168.2.100
CSeq: 25787 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Really destroying SIP dialog
'515f077817bb964c42ad4f8666ec4faa at 192.168.2.100' Method: BYE
kohana*CLI>
<--- SIP read from 192.168.2.3:5060 --->
REGISTER sip:192.168.2.100:5060 SIP/2.0
Via: SIP/2.0/UDP
192.168.2.3:5060;branch=z9hG4bK0ac55614520eebfb8.44f42ec8dfe39276e
Max-Forwards: 70
From: <sip:541 at 192.168.2.100:5060>;tag=4e6fbc7013
To: <sip:541 at 192.168.2.100:5060>
Call-ID: b6c364c12bbebd6a
CSeq: 13459 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK,
SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "Cordless @ 541"
<sip:541 at 192.168.2.3:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D10BD9C>"
Supported: gruu, path
User-Agent: Aastra 9480iCT/2.5.0.82
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 192.168.2.3 : 5060 (no NAT)
kohana*CLI>
<--- Transmitting (no NAT) to 192.168.2.3:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.3:5060;branch=z9hG4bK0ac55614520eebfb8.44f42ec8dfe39276e;received=192.168.2.3
From: <sip:541 at 192.168.2.100:5060>;tag=4e6fbc7013
To: <sip:541 at 192.168.2.100:5060>
Call-ID: b6c364c12bbebd6a
CSeq: 13459 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
kohana*CLI>
<--- Transmitting (no NAT) to 192.168.2.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.3:5060;branch=z9hG4bK0ac55614520eebfb8.44f42ec8dfe39276e;received=192.168.2.3
From: <sip:541 at 192.168.2.100:5060>;tag=4e6fbc7013
To: <sip:541 at 192.168.2.100:5060>;tag=as5b89ea0e
Call-ID: b6c364c12bbebd6a
CSeq: 13459 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 120
Contact: <sip:541 at 192.168.2.3:5060;transport=udp>;expires=120
Date: Tue, 31 Mar 2009 00:16:21 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'b6c364c12bbebd6a' in 32000 ms
(Method: REGISTER)
Scheduling destruction of SIP dialog
'0d5a326c417193e519f702946acfc94b at 192.168.2.100' in 32000 ms (Method:
NOTIFY)
Reliably Transmitting (no NAT) to 192.168.2.3:5060:
NOTIFY sip:541 at 192.168.2.3:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.100:5060;branch=z9hG4bK463d7f2e;rport
From: "asterisk" <sip:asterisk at 192.168.2.100>;tag=as6844ae67
To: <sip:541 at 192.168.2.3:5060;transport=udp>
Contact: <sip:asterisk at 192.168.2.100>
Call-ID: 0d5a326c417193e519f702946acfc94b at 192.168.2.100
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 93
Messages-Waiting: no
Message-Account: sip:asterisk at 192.168.2.100
Voice-Message: 0/0 (0/0)
---
kohana*CLI>
<--- SIP read from 192.168.2.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.100:5060;branch=z9hG4bK463d7f2e;rport=5060;received=192.168.2.100
From: "asterisk" <sip:asterisk at 192.168.2.100>;tag=as6844ae67
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=2673226879
Call-ID: 0d5a326c417193e519f702946acfc94b at 192.168.2.100
CSeq: 102 NOTIFY
Contact: "Cordless @ 541"
<sip:541 at 192.168.2.3:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D10BD9C>"
Server: Aastra 9480iCT/2.5.0.82
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog
'0d5a326c417193e519f702946acfc94b at 192.168.2.100' Method: NOTIFY
Issue History
Date Modified Username Field Change
======================================================================
2009-03-30 19:29 Herb Note Added: 0102440
======================================================================
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