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[a_villacis at palosant...]

I am trying to collect enough information about an problem a client is having with its asterisk 11.17.0 x86_64. This issue was observed before in 1.8.20, and we upgraded to 11.15.0 and then to 11.17.0 with no solution.

Background: this client is a telemarketing call-center that generates outgoing calls with close to a hundred agents operating simultaneously in peak hours. The system uses asterisk with FreePBX 2.8. In order to generate the calls, I wrote a program that
connects to Asterisk using the AMI protocol. This program expects the SIP agent extensions to be assigned as members of queues, of which there are about 20, as shown below:

9007 has 0 calls (max unlimited) in 'random' strategy (5s holdtime, 68s talktime), W:0, C:581, A:260, SL:82.6% within 60s
Members:
SIP/147 (ringinuse disabled) (dynamic) (On Hold) has taken 21 calls (last was 800 secs ago)
SIP/417 (ringinuse disabled) (dynamic) (In use) has taken 77 calls (last was 708 secs ago)
SIP/416 (ringinuse disabled) (dynamic) (In use) has taken 41 calls (last was 656 secs ago)
SIP/408 (ringinuse disabled) (dynamic) (In use) has taken 50 calls (last was 789 secs ago)
No Callers

The program runs "queue show" through AMI every few seconds. For each queue to be used in telemarketing, the program counts the number of members that are "Not In Use". If at least one is found, it reads that many phone numbers from the database and uses
the AMI Originate command on each one, as follows:

Action: Originate
Channel: Local/NNNNNNNNNN@from-internal
Exten: CCCC
Context: from-internal
Priority: 1
Async: true
ActionID: xxx

Here, NNNNNNNNNN is the number read from the database and CCCC is the queue extension in the FreePBX-created context that eventually runs the Queue() dialplan application for the corresponding queue. This causes the call to be connected between the
outgoing number and the queue, and is then assigned to a queue member by Asterisk. The dialplan is configured to route NNNNNNNNNN through one of a series of SIP trunks using the outbound routes as configured by FreePBX.

The issue is that although this strategy works correctly on the user's machine for a few days, we have been observing that eventually the application stops placing calls. The agents are all idle (all 90 to 100 of them), but the "queue show" command shows
them to be "In Use" on all queues. Furthermore, in normal operation, the "core show channels" command shows at most one channel for each configured SIP client in the "Up" state, but when calls stop being placed, the same command reports multiple channels
in the "Up" state, as follows (after sorting):

Local/9757007441@from-internal-0000a447;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5557007441,300,!47740412!!!3!572!(None)!1428426012.169192
Local/9759315789@from-internal-0000a456;1<ZOMBIE>!from-trunk-sip-5547740412!!1!Up!AppDial!(Outgoing Line)!9759315789!!!3!500!(None)!1428426084.169326
Local/9759315789@from-internal-0000a456;2!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740412/5559315789,300,!47740412!!!3!500!(None)!1428426084.169323
SIP/104-00014c61!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/0453511309468,300,!47740413!!!3!562!SIP/5547740413-00014c62!1428426022.169224
SIP/110-00014c2b!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing Line)!110!!!3!590!(None)!1428425994.169124
SIP/110-00014e4e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552114757,300,!47740413!!!3!92!(None)!1428426491.169760
SIP/113-00014c8c!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740420/0016499465293,300,!47740420!!!3!532!(None)!1428426052.169273
SIP/114-00014ce6!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/040,300,!47740410!!!3!430!(None)!1428426154.169384
SIP/115-00014ea4!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740400/0059144681666,300,!47740400!!!3!10!(None)!1428426574.169850
SIP/119-00014c26!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/016255863252,300,!47740413!!!3!593!(None)!1428425991.169113
SIP/119-00014d1a!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5552716011,300,!47740413!!!3!383!(None)!1428426201.169436
SIP/119-00014d4d!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740413/5556802622,300,!47740413!!!3!327!(None)!1428426257.169493
SIP/120-00014d5e!macro-dial-one!s!37!Up!Dial!SIP/230,"",trT!120!!!3!314!SIP/230-00014d5f!1428426270.169510
SIP/121-00014c24!EjecutivoROLLRATE!9014!1!Up!AppQueue!(Outgoing Line)!121!!!3!596!(None)!1428425988.169111
SIP/122-00014b56!EjecutivoQUADS!93000!1!Up!AppQueue!(Outgoing Line)!122!!!3!677!(None)!1428425906.168693
SIP/123-00014d53!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!320!(None)!1428426264.169499
SIP/123-00014e35!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/017222497260,300,!47740410!!!3!121!(None)!1428426463.169735
SIP/123-00014e9e!macro-dialout-trunk!s!19!Ring!Dial!SIP/5547740410/5556350254,300,!47740410!!!3!13!(None)!1428426570.169844
SIP/125-00014bc7!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740435/0445549261961,300,!47740435!!!3!626!(None)!1428425958.168938
SIP/125-00014cba!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740400/5521634648,300,!47740400!!!3!493!(None)!1428426091.169328
SIP/129-00014d8f!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740400/0050687069324,300,!47740400!!!3!277!(None)!1428426307.169565
SIP/130-00014a57!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740414/014777167359,300,!47740414!!!3!777!(None)!1428425807.168130
SIP/130-00014d08!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740414/5559502494,300,!47740414!!!3!402!(None)!1428426181.169418
SIP/130-00014d56!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740414/5559502494,300,!47740414!!!3!317!(None)!1428426267.169502
SIP/130-00014db0!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740414/5559502470,300,!47740414!!!3!253!(None)!1428426331.169598
SIP/130-00014e16!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740414/5559502494,300,!47740414!!!3!156!(None)!1428426428.169702
SIP/131-00014b8c!EjecutivoSEGUNDAS!99000!1!Up!AppQueue!(Outgoing Line)!131!!!3!651!(None)!1428425932.168799
SIP/131-00014d7e!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/5553625501,300,!47740410!!!3!294!(None)!1428426290.169548
SIP/131-00014e74!macro-dialout-trunk!s!19!Up!Dial!SIP/5547740410/5553526428,300,!47740410!!!3!45!SIP/5547740410-00014e75!1428426539.169798

In the example shown above, SIP/119, SIP/123, SIP/131 have three channels, and SIP/130 has five. Even assuming a bug in my program, a single SIP extension cannot be handling five simultaneous calls.

Now, here comes my speculation. I believe what is happening is that somehow, channels are not being cleaned on hangup, but somehow leave the extension capable of accepting a new call. My program sees queues with a mixture of non-affected idle members and
affected members (which appear in "queue show" as "In Use" but are actually idle and will accept a new call). As long as there is one non-affected member that can appear as "Not In Use", my program keeps placing calls on that queue, which are handled by
both affected an unaffected members. This would explain the multiple channels for a single SIP phone. As soon as all the members become affected in one queue, the dialing stalls.

Is this speculation sound, or even possible, given the internal architecture of Asterisk?
What tools can be used to debug this? Is this a channel leak? I know of DEBUG_THREADS for debugging deadlocks, but this does not seem like one.
What extra information can be provided to make a bug report on this?




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[a_villacis at palosant...]

El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió:

[p.beaumont at hatsoffs...]

I have seen a similar problem occasionally. We will be doing maintenance on a customer's server and they will have one or two "ghost" channels on their machine hundreds of hours old but with no call associated with them. So far we have just been rebooting their server or issuing a hangup command to the channels.
________________________________________
From: asterisk-users-bounces@lists.digium.com <asterisk-users-bounces@lists.digium.com> on behalf of Alex Villací­s Lasso <a_villacis@palosanto.com>
Sent: 08 April 2015 00:33
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Help debugging a possible SIP channel leak in 11.17.0, possible race condition

El 07/04/15 a las 17:38, Alex Villací­s Lasso escribió:

[vinicius at aittelecom...]

Have you tried Asterisk 13? The bridging mechanism has been completely rewritten on Asterisk 12, so there's no longer channel masquerading and zombie channels. Might be worth a try.

2015-04-07 20:33 GMT-03:00 Alex Villací­s Lasso <a_villacis@palosanto.com (a_villacis@palosanto.com)>:

[a_villacis at palosant...]

El 08/04/15 a las 08:22, Vinicius Fontes escribió: