[asterisk-bugs] [Asterisk 0016763]: macro-hangup executes after 15 minutes and 30 seconds on outbound calls

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Feb 3 15:21:07 CST 2010 

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16763 
====================================================================== 
Reported By:                cjonesmo
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16763
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.1.14 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-02-03 13:17 CST
Last Modified:              2010-02-03 15:21 CST
====================================================================== 
Summary:                    macro-hangup executes after 15 minutes and 30
seconds on outbound calls
Description: 
I'm on Asterisk 1.6.1.10 and outgoing calls disconnect (the macro-hangup
executes) consistently after 15 minutes and 30 seconds. 
====================================================================== 

---------------------------------------------------------------------- 
 (0117665) cjonesmo (reporter) - 2010-02-03 15:21
 https://issues.asterisk.org/view.php?id=16763#c117665 
---------------------------------------------------------------------- 
Here is the SIP Trace with debug turned on - all that seems to show up is
the hangup call macro executing though - these are just the last few events
before  (and including) the hangup - odd that only the caller (PBX Phone)
side of the call gets disconnected though. If I hangup the called leg of
the call (even much later) then I see hangup call macro execute again:

papayapbx01*CLI> 
    -- Registered SIP '2100' at XXX.XXX.XXX.XXX port 56822
Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:56822:
OPTIONS sip:2100 at XXX.XXX.XXX.XXX:56822;rinstance=4f9d0fac7004c18e SIP/2.0

Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK43f3f472;rport

Max-Forwards: 70

From: "Unknown" <sip:Unknown at XXX.XXX.XXX.XXX>;tag=as6b0b5de6

To: <sip:2100 at XXX.XXX.XXX.XXX:56822;rinstance=4f9d0fac7004c18e>

Contact: <sip:Unknown at XXX.XXX.XXX.XXX>

Call-ID: 02aa68ad0a490a0b5fac855455b88620 at XXX.XXX.XXX.XXX

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.6.1.10

Date: Wed, 03 Feb 2010 21:05:12 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Content-Length: 0




---
       > Saved useragent "X-Lite release 1103k stamp 53621" for peer 2100

papayapbx01*CLI> 

<--- Transmitting (NAT) to XXX.XXX.XXX.XXX:56822 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP
192.168.1.102:56822;branch=z9hG4bK-d8754z-ab2d646fba73722e-1---d8754z-;received=XXX.XXX.XXX.XXX;rport=56822

From: "Joe User"<sip:2100 at mysipserver.mydomain.net>;tag=28205421

To: "Joe User"<sip:2100 at mysipserver.mydomain.net>;tag=as00ca17cd

Call-ID: Mjk0NTA2MTFkNTMyY2IyNGI4NzRhMmQwZjg3Y2ZmYzE.

CSeq: 851 REGISTER

Server: Asterisk PBX 1.6.1.10

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO

Supported: replaces, timer

Expires: 60

Contact:
sip:2100 at XXX.XXX.XXX.XXX:56822;rinstance=4f9d0fac7004c18e;expires=60

Date: Wed, 03 Feb 2010 21:05:12 GMT

Content-Length: 0




<------------>

papayapbx01*CLI> 
Scheduling destruction of SIP dialog
'Mjk0NTA2MTFkNTMyY2IyNGI4NzRhMmQwZjg3Y2ZmYzE.' in 32000 ms (Method:
REGISTER)

papayapbx01*CLI> 

<--- SIP read from UDP://XXX.XXX.XXX.XXX:56822 --->
SIP/2.0 200 OK

Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5060;branch=z9hG4bK43f3f472;rport=5060

Contact: <sip:192.168.1.102:56822>

To:
<sip:2100 at XXX.XXX.XXX.XXX:56822;rinstance=4f9d0fac7004c18e>;tag=fe62b954

From: "Unknown"<sip:Unknown at XXX.XXX.XXX.XXX>;tag=as6b0b5de6

Call-ID: 02aa68ad0a490a0b5fac855455b88620 at XXX.XXX.XXX.XXX

CSeq: 102 OPTIONS

Accept: application/sdp

Accept-Language: en

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO

User-Agent: X-Lite release 1103k stamp 53621

Content-Length: 0




<------------->

papayapbx01*CLI> 
--- (12 headers 0 lines) ---

papayapbx01*CLI> 
Really destroying SIP dialog
'02aa68ad0a490a0b5fac855455b88620 at XXX.XXX.XXX.XXX' Method: OPTIONS

papayapbx01*CLI> 

<--- SIP read from UDP://XXX.XXX.XXX.XXX:56822 --->





<------------->

papayapbx01*CLI> 
    -- Registered SIP '2100' at XXX.XXX.XXX.XXX port 1202

papayapbx01*CLI> 
       > Saved useragent "Grandstream GXP2000 1.2.2.26" for peer 2100

papayapbx01*CLI> 
    -- Executing [h at macro-dialout-trunk:1]
Macro("SIP/2100-00000a33",
"hangupcall,") in new stack
    -- Executing [s at macro-hangupcall:1]
GotoIf("SIP/2100-00000a33",
"1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,4)
    -- Executing [s at macro-hangupcall:4]
GotoIf("SIP/2100-00000a33",
"1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s at macro-hangupcall:7]
GotoIf("SIP/2100-00000a33",
"1?theend") in new stack
    -- Goto (macro-hangupcall,s,9)
    -- Executing [s at macro-hangupcall:9]
Hangup("SIP/2100-00000a33",
"") in new stack
  == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
'SIP/2100-00000a33' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on
'SIP/2100-00000a33'

papayapbx01*CLI> 
  == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on
'SIP/2100-00000a33' in macro 'dialout-trunk'
  == Spawn extension (from-internal, 2815551212, 4) exited non-zero on
'SIP/2100-00000a33' 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-02-03 15:21 cjonesmo       Note Added: 0117665                          
======================================================================

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