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Summary:ASTERISK-13819: Timed out parked calls always return to originating extension
Reporter:Herb (herb)Labels:
Date Opened:2009-03-24 21:11:28Date Closed:2011-06-07 14:07:59
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Resources/res_features
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
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)
Comments:By: Russell Bryant (russell) 2009-03-29 00:24:50

Can you please provide console output of a working and non-working case?
By: Herb (herb) 2009-03-30 19:25:46

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@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@192.168.2.100>;tag=as79eec948
To: <sip:541@192.168.2.3:5060;transport=udp>
Contact: <sip:143@192.168.2.100>
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd@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@192.168.2.100>;tag=as79eec948
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=190959364
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd@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@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@192.168.2.100>;tag=as79eec948
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=190959364
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd@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@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@192.168.2.3:5060;transport=udp>
set_destination: Parsing <sip:541@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@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@192.168.2.100>;tag=as79eec948
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=190959364
Contact: <sip:143@192.168.2.100>
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd@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@192.168.2.100' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:541@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@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@192.168.2.100>;tag=as79eec948
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=190959364
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd@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@192.168.2.100>;tag=as79eec948
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=190959364
Call-ID: 4d4db74e1f81ff3e0d1b8277392cc6cd@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@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@192.168.2.100:5060>;tag=4e6fbc7013
To: <sip:541@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@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@192.168.2.100:5060>;tag=4e6fbc7013
To: <sip:541@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@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


By: Herb (herb) 2009-03-30 19:29:30

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@192.168.2.100' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:541@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@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@192.168.2.100>;tag=as50162f53
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=414947709
Call-ID: 102963404eb9d49051ff29681a161938@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@192.168.2.100>;tag=as50162f53
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=414947709
Call-ID: 102963404eb9d49051ff29681a161938@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@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@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@192.168.2.100>;tag=as398e24c9
To: <sip:541@192.168.2.3:5060;transport=udp>
Contact: <sip:143@192.168.2.100>
Call-ID: 515f077817bb964c42ad4f8666ec4faa@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@192.168.2.100>;tag=as398e24c9
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=2301220826
Call-ID: 515f077817bb964c42ad4f8666ec4faa@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@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@192.168.2.100>;tag=as398e24c9
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=2301220826
Call-ID: 515f077817bb964c42ad4f8666ec4faa@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@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@192.168.2.3:5060;transport=udp>
set_destination: Parsing <sip:541@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@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@192.168.2.100>;tag=as398e24c9
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=2301220826
Contact: <sip:143@192.168.2.100>
Call-ID: 515f077817bb964c42ad4f8666ec4faa@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:143@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@192.168.2.3:5060;transport=udp>;tag=2301220826
To: "Herb" <sip:143@192.168.2.100>;tag=as398e24c9
Call-ID: 515f077817bb964c42ad4f8666ec4faa@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@192.168.2.3:5060;transport=udp>;tag=2301220826
To: "Herb" <sip:143@192.168.2.100>;tag=as398e24c9
Call-ID: 515f077817bb964c42ad4f8666ec4faa@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@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@192.168.2.100:5060>;tag=4e6fbc7013
To: <sip:541@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@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@192.168.2.100:5060>;tag=4e6fbc7013
To: <sip:541@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@192.168.2.100:5060>;tag=4e6fbc7013
To: <sip:541@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@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@192.168.2.100' in 32000 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.2.3:5060:
NOTIFY sip:541@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@192.168.2.100>;tag=as6844ae67
To: <sip:541@192.168.2.3:5060;transport=udp>
Contact: <sip:asterisk@192.168.2.100>
Call-ID: 0d5a326c417193e519f702946acfc94b@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@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@192.168.2.100>;tag=as6844ae67
To: <sip:541@192.168.2.3:5060;transport=udp>;tag=2673226879
Call-ID: 0d5a326c417193e519f702946acfc94b@192.168.2.100
CSeq: 102 NOTIFY
Contact: "Cordless @ 541" <sip:541@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@192.168.2.100' Method: NOTIFY


By: Herb (herb) 2009-03-30 19:30:42

This server is being installed tomorrow, so not sure how much more information I will be able to provide.  But I will try and help out with what I can.

Thanks again,
Herb
By: Leif Madsen (lmadsen) 2009-06-10 12:38:30

I have a feeling something like this just recently got fixed. Can you try the latest 1.4 release to see if this is still happening for you? Thanks!
By: Herb (herb) 2009-06-17 13:49:14

Aloha,

I compiled & installed version 1.4.25.1 and this problem still exists for me.  This problem does not occur with versions 1.6 or 1.6.1.

~Herb
By: Leif Madsen (lmadsen) 2009-06-22 08:53:29

Sorry, I mis-spoke -- I didn't mean latest 1.4 release, I meant latest 1.4 branch, i.e.:

svn co http://svn.asterisk.org/svn/asterisk/branches/1.4

Then compile that and try testing with it.
By: dlnoah (dlnoah) 2009-06-22 13:56:27

I am actually seeing that comebacktoorigin = no is NOT working for me on any of my four servers, running either 1.6.0.9 or 1.6.1.1.

My configuration is as follows:
(features.conf)
[general]
comebacktoorigin = no

(extensions.conf)
exten => s,1,NoOp("Parked call timed out - ${PARKEDAT} - CID: "${CALLERID(all)})
exten => s,n,... some more handling for the timed out park

The console behavior is as follows:
   -- Stopped music on hold on IAX2/dundi-directlink-7325
[Jun 22 10:32:18] WARNING[322]: features.c:2975 manage_parkinglot: Dialfeatures not found on IAX2/dundi-directlink-7325, using default!
   -- Added extension 'IAX20dundi-directlink' priority 1 to park-dial (0xb7409258)
[Jun 22 10:32:18] WARNING[322]: features.c:2984 manage_parkinglot: now going to parkedcallstimeout,s,1 | ps is 1071
 == Timeout for IAX2/dundi-directlink-7325 parked on 1071 (default). Returning to parkedcallstimeout,IAX20dundi-directlink,1
 == Starting IAX2/dundi-directlink-7325 at parkedcallstimeout,IAX20dundi-directlink,1 failed so falling back to exten 's'
   -- Executing [s@parkedcallstimeout:1] NoOp("IAX2/dundi-directlink-7325", ""Parked call timed out - 1071 - CID: ""Noah" <100>") in new stack


For that specific example, I parked myself, so the parking device was IAX2/dundi-directlink-7325.  If someone parks a call using a SIP device (e.g. SIP/1000), then Asterisk will first attempt to return to parkedcallstimeout,SIP01010,1 before failing at that and catching the parkedcallstimeout,s,1 extension.  It looks like Asterisk is still trying to return the call to the device that parked it, but the / is getting replaced with a 0, which causes the channel creation to fail.

Also, if you do not have comebacktoorigin = no set, 1.6.x will attempt to call the parking device, with the 0 substituted for the / (so it tries to call SIP01010 instead of SIP/1010, failing), and then fails into the park-dial,t,1 context.  The fall-through to park-dial,t,1 is the behavior that 1.4.x exhibited when the 30 seconds of ringing timed out and is expected behavior, but the parking extension is never successfully being rang because of Asterisk trying to call SIP01010 instead of SIP/1010.


By: Herb (herb) 2009-06-24 18:32:48

My 1.6.1.0 server is working just fine in this regard.  I have not installed 1.6.0.9 or 1.6.1.1.

I have not attempted the most recent SVN branch of 1.4 yet either, since that is our main server.  I hope to get some time soon.
By: dlnoah (dlnoah) 2009-06-24 18:40:50

Herb, I'm curious as to what the console output of a timed-out parked call looks like on your 1.6.1.0 server.  On mine, the comebacktoorigin = no "works" in the sense that the first successfully executed dialplan priority after a parked call times out is the designated parkedcallstimeout,s,1; but the call tries to go to parkedcallstimeout,SIP0{Originating Extension},1 first and fails at that, leaving the call hanging for about 5 seconds while Asterisk attempts to call the originating extension and fails because of subbing the / with a 0 in the channel name (SIP/1010 -> SIP01010).  

If yours doesn't do that, could you maybe e-mail me the pertinent parts of your parking configs so I can find out what's wrong with my setup?
By: Herb (herb) 2009-06-24 19:24:13

Hi,

The way we have it set here, is that if a parked call times out, it rings all the phones since there's no guarantee that someone will be at the originating extension by the time the parked call times out.

If after 60 secs and no one answers the call, it then rings our main number/building.

{features.conf}
[general]
comebacktoorigin = no

{extensions.conf}
[parkedcallstimeout]
exten => s,1,Goto(mainmenu,s,1)

[mainmenu]
exten => s,1,Answer
exten => s,2,Dial(${ALLPHONES},60,rtTkK)
exten => s,3,Dial(IAX2/server.ip.com/101,,rtTkK)
include => default

[Jun 24 14:01:15] WARNING[28784]: features.c:2984 manage_parkinglot: now going to parkedcallstimeout,s,1 | ps is 701

I've tried both a SIP call and IAX2 call and both are working for us.

Hope that helps.
By: Leif Madsen (lmadsen) 2009-07-20 14:01:38

As far as I can tell, this feature doesn't exist in 1.4 -- that would be why it doesn't work :)