[asterisk-bugs] [Asterisk 0014741]: Timed out parked calls always return to originating extension
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Mar 30 19:25:46 CDT 2009
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=14741
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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 19:25 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)
======================================================================
----------------------------------------------------------------------
(0102439) Herb (reporter) - 2009-03-30 19:25
http://bugs.digium.com/view.php?id=14741#c102439
----------------------------------------------------------------------
Here is a working output using version 1.6.1-rc3:
Audio is at 192.168.2.100 port 19332
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=z9hG4bK03c4ca34;rport
Max-Forwards: 70
From: "Herb" <sip:143 at 192.168.2.100>;tag=as79eec948
To: <sip:541 at 192.168.2.3:5060;transport=udp>
Contact: <sip:8088853143 at 192.168.2.100>
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd at 192.168.2.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.1.0-rc3
Date: Tue, 31 Mar 2009 00:21:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 316
v=0
o=root 1246613375 1246613375 IN IP4 192.168.2.100
s=Asterisk PBX 1.6.1.0-rc3
c=IN IP4 192.168.2.100
t=0 0
m=audio 19332 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 UDP://192.168.2.3:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.2.100:5060;branch=z9hG4bK03c4ca34;rport=5060;received=192.168.2.100
From: "Herb" <sip:143 at 192.168.2.100>;tag=as79eec948
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=190959364
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd 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 UDP://192.168.2.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.100:5060;branch=z9hG4bK03c4ca34;rport=5060;received=192.168.2.100
From: "Herb" <sip:143 at 192.168.2.100>;tag=as79eec948
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=190959364
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd 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)/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 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=z9hG4bK3276237e;rport
Max-Forwards: 70
From: "Herb" <sip:143 at 192.168.2.100>;tag=as79eec948
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=190959364
Contact: <sip:8088853143 at 192.168.2.100>
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd at 192.168.2.100
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.1.0-rc3
Content-Length: 0
---
Scheduling destruction of SIP dialog
'4d4db74e1f81ff3e0d1b8277392cc6cd 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=z9hG4bK0d60b2f6;rport
Max-Forwards: 70
From: "Herb" <sip:143 at 192.168.2.100>;tag=as79eec948
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=190959364
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd at 192.168.2.100
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.1.0-rc3
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
kohana*CLI>
---
kohana*CLI>
<--- SIP read from UDP://192.168.2.3:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.2.100:5060;branch=z9hG4bK0d60b2f6;rport=5060;received=192.168.2.100
From: "Herb" <sip:143 at 192.168.2.100>;tag=as79eec948
To: <sip:541 at 192.168.2.3:5060;transport=udp>;tag=190959364
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd 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
'4d4db74e1f81ff3e0d1b8277392cc6cd at 192.168.2.100' Method: INVITE
kohana*CLI>
<--- SIP read from UDP://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=z9hG4bKd716aae3a554726c0.81743cedeea19a932
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: 13462 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) ---
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=z9hG4bKd716aae3a554726c0.81743cedeea19a932;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=as148c4e2a
Call-ID: b6c364c12bbebd6a
CSeq: 13462 REGISTER
Server: Asterisk PBX 1.6.1.0-rc3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 120
Contact: <sip:541 at 192.168.2.3:5060;transport=udp>;expires=120
Date: Tue, 31 Mar 2009 00:21:36 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'b6c364c12bbebd6a' in 32000 ms
(Method: REGISTER)
[Mar 30 14:21:53] WARNING[23889]: features.c:2950 manage_parkinglot: now
going to parkedcallstimeout,s,1 | ps is 701
Really destroying SIP dialog 'b6c364c12bbebd6a' Method: REGISTER
Issue History
Date Modified Username Field Change
======================================================================
2009-03-30 19:25 Herb Note Added: 0102439
======================================================================
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