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Summary:ASTERISK-15458: macro-hangup executes after 15 minutes and 30 seconds on outbound calls
Reporter:CJ (cjonesmo)Labels:
Date Opened:2010-02-03 13:17:14.000-0600Date Closed:2011-06-07 14:08:04
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
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Environment:Attachments:
Description:I'm on Asterisk 1.6.1.10 and outgoing calls disconnect (the macro-hangup executes) consistently after 15 minutes and 30 seconds.

****** ADDITIONAL INFORMATION ******

I had been seeing an invite from the phone when this happened and a 488 "Not Acceptable Here", so I set "ingnoredspversion=yes" on my carrier. Now, I don't see the 488 message (or the register) but the calls still disconnect. I have also tried setting registrytimeout=60 and maxexpiry=60 to no avail.

I do have Queuemetrics loaded on the system for a couple of call center agents, but all of my phones have this issue. I did try changing the maxtalktime and maxqueuetime in queuemetrics to 0, but that doesn't seem to fix it either.

I don't see 1.6.1.10 as an option for this message, so perhaps I need to upgrade. Please advise if you have any help for me.

Comments:By: CJ (cjonesmo) 2010-02-03 13:21:13.000-0600

Oh - and I forgot to mention - the CALLER (SIP Extension on Asterisk) hears a fast busy when the hang-up occurs, but the CALLEE (person dialed - outside) - can still hear the PBX phone.
By: Leif Madsen (lmadsen) 2010-02-03 14:17:17.000-0600

Additional information is required here:

* You're using SIP, so per the bug guidelines, you need to include:

 * SIP trace from the Asterisk console
 * SIP history, enabled via sip.conf
 * console trace of the issue happening with 'core set verbose 10' enabled
 * console trace with debug level logging enabled via logger.conf and 'core set debug 10'

Please upload these files as text files to the issue.
By: CJ (cjonesmo) 2010-02-03 15:21:07.000-0600

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:

�[Kpapayapbx01*CLI>
   -- Registered SIP '2100' at XXX.XXX.XXX.XXX port 56822
Reliably Transmitting (NAT) to XXX.XXX.XXX.XXX:56822:
OPTIONS sip:2100@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@XXX.XXX.XXX.XXX>;tag=as6b0b5de6

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

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

Call-ID: 02aa68ad0a490a0b5fac855455b88620@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

�[Kpapayapbx01*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@mysipserver.mydomain.net>;tag=28205421

To: "Joe User"<sip:2100@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@XXX.XXX.XXX.XXX:56822;rinstance=4f9d0fac7004c18e;expires=60

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

Content-Length: 0




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

�[Kpapayapbx01*CLI>
Scheduling destruction of SIP dialog 'Mjk0NTA2MTFkNTMyY2IyNGI4NzRhMmQwZjg3Y2ZmYzE.' in 32000 ms (Method: REGISTER)

�[Kpapayapbx01*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@XXX.XXX.XXX.XXX:56822;rinstance=4f9d0fac7004c18e>;tag=fe62b954

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

Call-ID: 02aa68ad0a490a0b5fac855455b88620@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




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

�[Kpapayapbx01*CLI>
--- (12 headers 0 lines) ---

�[Kpapayapbx01*CLI>
Really destroying SIP dialog '02aa68ad0a490a0b5fac855455b88620@XXX.XXX.XXX.XXX' Method: OPTIONS

�[Kpapayapbx01*CLI>

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





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

�[Kpapayapbx01*CLI>
   -- Registered SIP '2100' at XXX.XXX.XXX.XXX port 1202

�[Kpapayapbx01*CLI>
      > Saved useragent "Grandstream GXP2000 1.2.2.26" for peer 2100

�[Kpapayapbx01*CLI>
   -- Executing [h@macro-dialout-trunk:1] �[1;36;40mMacro�[0;37;40m("�[1;35;40mSIP/2100-00000a33�[0;37;40m", "�[1;35;40mhangupcall,�[0;37;40m") in new stack
   -- Executing [s@macro-hangupcall:1] �[1;36;40mGotoIf�[0;37;40m("�[1;35;40mSIP/2100-00000a33�[0;37;40m", "�[1;35;40m1?skiprg�[0;37;40m") in new stack
   -- Goto (macro-hangupcall,s,4)
   -- Executing [s@macro-hangupcall:4] �[1;36;40mGotoIf�[0;37;40m("�[1;35;40mSIP/2100-00000a33�[0;37;40m", "�[1;35;40m1?skipblkvm�[0;37;40m") in new stack
   -- Goto (macro-hangupcall,s,7)
   -- Executing [s@macro-hangupcall:7] �[1;36;40mGotoIf�[0;37;40m("�[1;35;40mSIP/2100-00000a33�[0;37;40m", "�[1;35;40m1?theend�[0;37;40m") in new stack
   -- Goto (macro-hangupcall,s,9)
   -- Executing [s@macro-hangupcall:9] �[1;36;40mHangup�[0;37;40m("�[1;35;40mSIP/2100-00000a33�[0;37;40m", "�[1;35;40m�[0;37;40m") 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'

�[Kpapayapbx01*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'
By: CJ (cjonesmo) 2010-02-04 10:08:01.000-0600

Another note - this is apparently only happening with our Grandstream GXP2000 phones. I know they're not very good, but they DO work on our 1.4 based Asterisk system. They are running firmware version 1.2.19 and 1.2.26. I will also post this information to Grandstream and try upgrading the PBX to 1.6.2 tonight.
By: CJ (cjonesmo) 2010-02-04 12:12:03.000-0600

You can close this if you want to - setting session-timers=refuse solved the problem for the GXP2000. Someone may need to know this if they're running 1.6.1.10 and GXP2000's.
By: Leif Madsen (lmadsen) 2010-02-04 14:45:31.000-0600

Closed per the reporter.