IRC log for #asterisk-dev on 20120911

00:07.00*** join/#asterisk-dev zerohalo (~zerohalo@74.61.196.236)
01:12.10JunK-Ylo guys
01:13.10*** join/#asterisk-dev mjordan (~mjordan@user-69-1-6-49.knology.net)
01:13.11*** mode/#asterisk-dev [+o mjordan] by ChanServ
01:22.21JunK-Y*CLI> !cat /tmp/refs
01:22.21JunK-Y0x9980320 -1   astobj2.c:948:cd_cb_debug (deref object via container destroy) [@1]
01:22.35JunK-Yhow can I know which container's name was it?
01:35.50mjordanJunK-Y: the first item in the line is the memory address of the ref counted object.  Does it show up in the rest of your ref count log?
01:38.42JunK-Ymjordan: no, all lines are like that
01:42.07mjordanhm.
01:42.27mjordanyou could define REF_DEBUG in a header and recompile.  That should enable it globally in all files that support ref debugging.
01:42.31mjordanyou'll get a lot of output however
01:45.37JunK-Ythat would be great to have it via compilers flags (after doing --enable-dev-mode
01:52.52*** join/#asterisk-dev jmls1 (~julian@host217-36-208-155.in-addr.btopenworld.com)
03:12.15*** join/#asterisk-dev mjordan (~mjordan@user-69-1-6-49.knology.net)
03:12.15*** mode/#asterisk-dev [+o mjordan] by ChanServ
06:07.48*** join/#asterisk-dev schmidts (~schmidts@Lo20-rt01.lm33.sil.at)
06:07.50schmidtsgood morning
06:11.43*** join/#asterisk-dev hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e)
08:30.45*** join/#asterisk-dev alsuren (~dlaban@80.169.133.251)
09:27.53wdoekeswho/what is using nested functions?
10:04.14*** join/#asterisk-dev tzafrir (~tzafrir@local.xorcom.com)
10:04.14*** mode/#asterisk-dev [+o tzafrir] by ChanServ
10:09.34*** join/#asterisk-dev pnlarsson (~niklas@fw1.gml.g.icnet.infracom.se)
10:12.11*** join/#asterisk-dev snuff-work (~snuffy@210.9.82.197)
10:12.11*** mode/#asterisk-dev [+o snuff-work] by ChanServ
10:41.20*** join/#asterisk-dev kpfleming (~kpfleming@unaffiliated/kpfleming)
10:58.20*** join/#asterisk-dev micols (~t@rlogin.dk)
11:14.19*** join/#asterisk-dev micols (~t@rlogin.dk)
11:16.37*** join/#asterisk-dev pnlarsson (~niklas@fw1.gml.g.icnet.infracom.se)
11:20.41Kobaz.
11:20.51Kobaz[2012-09-11 02:09:35] WARNING[10811] pbx.c: PBX requires Asterisk to be fully booted
11:20.54Kobaz[2012-09-11 02:09:35] WARNING[10811] chan_sip.c: Failed to start PBX :(
11:20.55Kobazso is that bad
11:24.18kpflemingno, it means that chan_sip processed an INVITE before the PBX was ready
11:24.28kpflemingso that incoming call failed, same as it would if asterisk had not been running
11:25.18Kobazah okay
11:25.31Kobazbecause asterisk was also unusable when it finished starting
11:25.37Kobazdialplan wasn't loaded, not sure why
11:26.37Kobazmight be a module load order issue
11:27.10kpflemingweird
11:31.30*** join/#asterisk-dev caio1982 (~caio1982@189.125.242.3)
11:36.16*** join/#asterisk-dev elred_ (~elred_@unaffiliated/elred-/x-5010831)
12:18.49*** join/#asterisk-dev pabelanger (~pabelange@asterisk/contributor-and-bug-marshal/pabelanger)
12:18.49*** mode/#asterisk-dev [+o pabelanger] by ChanServ
12:41.03*** join/#asterisk-dev micols (~t@rlogin.dk)
12:46.18*** join/#asterisk-dev hehol (~Adium@2a01:198:71d:0:21f:d0ff:fea1:568e)
12:59.09*** join/#asterisk-dev mjordan (~mjordan@nat/digium/x-mjzurswhrrcgtqsn)
12:59.09*** mode/#asterisk-dev [+o mjordan] by ChanServ
13:00.14*** join/#asterisk-dev serafie (~erin@nat/digium/x-habbnjulgjlkghdi)
13:29.19*** join/#asterisk-dev sruffell (~sruffell@asterisk/the-kernel-guy/sruffell)
13:29.19*** mode/#asterisk-dev [+o sruffell] by ChanServ
13:33.39*** join/#asterisk-dev wedhorn (~wedhorn@123-243-209-34.static.tpgi.com.au)
13:52.09*** join/#asterisk-dev zerohalo (~zerohalo@74.61.196.236)
14:02.09*** join/#asterisk-dev putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson)
14:02.09*** mode/#asterisk-dev [+o putnopvut] by ChanServ
14:07.54*** join/#asterisk-dev leedm777 (~Adium@nat/digium/x-orkqxidnokgenlbm)
14:32.19*** join/#asterisk-dev wudles (~wudles@gateway.secureinstrument.com)
14:59.53*** part/#asterisk-dev jsmith (~jsmith@fedora/jsmith)
15:13.07*** join/#asterisk-dev acidfu (~inetrio.c@75-119-242-110.dsl.teksavvy.com)
15:42.11*** join/#asterisk-dev marv (~Adium@173-14-206-193-huntsville.hfc.comcastbusiness.net)
15:53.27*** join/#asterisk-dev anthm (~anthm@freeswitch/developer/anthm)
16:14.30*** join/#asterisk-dev wudles (~wudles@gateway.secureinstrument.com)
16:45.30*** join/#asterisk-dev rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com)
16:45.33*** part/#asterisk-dev rgagnon (~rgagnon@rrcs-71-42-183-54.sw.biz.rr.com)
17:04.26*** part/#asterisk-dev jmls1 (~julian@host217-36-208-155.in-addr.btopenworld.com)
17:51.51*** join/#asterisk-dev anonymouz666 (~anonymouz@189-25-73-113.user.veloxzone.com.br)
18:11.34pabelangerSo what did we decide with the configure filename for res_xmpp and res_corosync for asterisk 11?
18:12.32mjordanpabelanger: nothing was decided
18:12.57pabelangerSo we going to default to no change?
18:13.31mjordanthere's no patch on review board as no one submitted anything, and there was hardly a concurrence on what direction to take :-)
18:13.52mjordansince no consensus was reached, the default stands, for better or worse
18:14.12pabelangerI'll get a patch up on reviewboard renaming xmpp.conf to res_xmpp.conf
18:14.16pabelangerthat should be a easy sell
18:14.27pabelangersince all the other res modules are like that
18:18.34QwellIt's a bit late to be making those kinds of changes...
18:21.48pabelangerfor some reason, it was not caught in the code review process
18:23.24mjordanI'm okay with the change being proposed, but I'd hope to get concurrence on the patch this week
18:53.03*** join/#asterisk-dev ZX81 (~ZX81@97.104.212.126)
19:00.45*** join/#asterisk-dev infobot (~infobot@rikers.org)
19:00.45*** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Many thanks to kpfleming for 7 years of awesome.
19:06.53ZX81hi, anyone able to take a look at https://issues.asterisk.org/jira/browse/ASTERISK-20402
19:07.53pabelangerZX81: If you can attach a debug log, it will help a developer
19:08.23ZX81there's not really much in the debug - it just shows what I described occurring, but ok
19:08.27ZX81:-)
19:08.43pabelangerhttps://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
19:08.55pabelangerSave time and provide as much information about it
19:11.05ZX81will have to run after hours because it's a busy call center and finding the right call in a debug log is somewhat impossible
19:11.34pabelangerZX81: checkout features.conf
19:11.37ZX81I had a look in res_features but all the masquerading and swapping round of channels etc made my head hurt
19:11.45pabelangerthey say to use option h for queue()
19:11.52ZX81no friggin way
19:11.57ZX81then I can't read...
19:12.08pabelangeror H
19:12.19ZX81yeah
19:12.22ZX81just seen now
19:12.58ZX81hmm
19:13.13ZX81that's actually what's happening
19:13.15pabelangerBut, i don't think you want to disconnect
19:13.18ZX81yeah
19:13.19pabelangeryou want to cancel the call
19:13.31ZX81yeah which is what it should be doing without the h option
19:13.32pabelangermight have to do some dialplan majic
19:13.39pabelangermagic*
19:13.43ZX81the h option is basically happening without specifying
19:13.58ZX81how so?
19:14.39pabelangerActually
19:14.47pabelangeryou would have to add H / h to the dial command
19:14.48pabelangerI think
19:14.56pabelangersince you want to hangup the transfer
19:15.16ZX81hmm
19:15.16ZX81yeah
19:15.21ZX81that looks possible
19:15.41ZX81will give it a whirl
19:21.18ZX81weird
19:21.22ZX81that does the same thing still
19:21.35ZX81let me try with h instead of H
19:23.25ZX81lol
19:23.27ZX81same both ways
19:23.40ZX81<PROTECTED>
19:23.40ZX81and
19:24.16ZX81<PROTECTED>
19:28.20ZX81tried h and H in queue app too
19:28.35ZX81always completes transfer on * regardless of settings
19:28.47*** join/#asterisk-dev kpfleming (~kpfleming@unaffiliated/kpfleming)
19:37.56ZX81I've added a debug log
19:47.14pabelangerZX81: *CLI> core show application Dial
19:47.34pabelangerRead H
19:47.36pabelangerthe note
19:48.26ZX81?
19:48.29ZX81<PROTECTED>
19:48.29ZX81<PROTECTED>
19:48.31ZX81<PROTECTED>
19:48.32ZX81<PROTECTED>
19:48.33ZX81<PROTECTED>
19:48.34ZX81<PROTECTED>
19:49.01pabelangerTry Answer() before Dial()
19:49.04pabelangersee if that happens
19:49.10pabelangeralso, use pastebin
19:49.12pabelangerand don't flood
19:49.17pabelangerpeople will get angry
19:49.17ZX81crmvoice*CLI> features show
19:49.18ZX81Disconnect Call           *       *
19:49.26ZX81even on a live call?
19:49.27ZX81ok
19:49.30ZX81will give it a go
19:51.52pabelangerya
19:51.57pabelangerjust giving you things to try
19:52.00ZX81nah
19:52.01ZX81same deal
19:52.12pabelangerk
19:52.18pabelangerthen sounds like it might be a bug
19:52.46ZX81lol, get sick of that hold music pretty quickly - hate to think what it's like for real agents :-)
19:54.04russellbwell they may still prefer that music over talking to people
19:54.16ZX81yeah true :-)
19:54.26russellbwould
19:54.27russellb:-p
19:55.37pabelangerFeature detected: fname=Attended Transfer sname=atxfer exten=##
19:55.41pabelangerwhich starts
19:55.49pabelangerbut I don't see the Feature detect for the disconnect
19:56.57ZX81## is what I have defined for attended transfer
19:57.02ZX81it then says "Transfer'
19:57.11ZX81then I send dtmf for dest
19:57.55pabelangerRight
19:58.01pabelangerbut when you press * to disconnect
19:58.05pabelangerasterisk should detect it
19:59.17ZX81[Sep 11 14:30:42] DTMF[6996] channel.c: DTMF begin '*' received on SIP/phono4-ext.voxeolabs.net-000000b2
19:59.17ZX81[Sep 11 14:30:42] DTMF[6996] channel.c: DTMF begin passthrough '*' on SIP/phono4-ext.voxeolabs.net-000000b2
19:59.28ZX81[Sep 11 14:30:42] DTMF[6996] channel.c: DTMF end emulation of '*' queued on SIP/phono4-ext.voxeolabs.net-000000b2
20:00.03pabelangerdirectmedia=no?
20:00.08pabelangerfor your sip peer
20:00.26ZX81directmedia=yes    in general
20:00.30ZX81sec while I check peer
20:00.47ZX81nothing in peer
20:00.53pabelangerset it to no
20:00.55pabelangerand retry
20:01.34pabelanger*shrugs*
20:02.25ZX81same
20:02.27ZX81:-)
20:02.50pabelangerWelp
20:03.00pabelangerGuess we need to triage it and see if anybody else can reproduce it
20:03.08ZX81I posted all my configs
20:03.13ZX81it's pretty easy to cause
20:03.28ZX81went back through like 10k revisions of Asterisk
20:03.54ZX81same thing happens with just agentlogin, queue, then features and dial
20:43.58*** join/#asterisk-dev tzafrir (~tzafrir@212.179.75.202)
20:43.59*** mode/#asterisk-dev [+o tzafrir] by ChanServ
20:54.40*** part/#asterisk-dev rillian (~giles@mf4-xiph.osuosl.org)
21:06.54*** join/#asterisk-dev malcolmd (~malcolmd@pdpc/sponsor/digium/malcolmd)
21:06.55*** mode/#asterisk-dev [+o malcolmd] by ChanServ
21:42.40*** join/#asterisk-dev ZX81 (~ZX81@97.104.212.126)
22:03.16*** join/#asterisk-dev ZX81 (~ZX81@97.104.212.126)
23:14.55*** join/#asterisk-dev snuff-work (~snuffy@210.9.82.197)
23:14.55*** mode/#asterisk-dev [+o snuff-work] by ChanServ
23:38.40*** part/#asterisk-dev mjordan (~mjordan@nat/digium/x-mjzurswhrrcgtqsn)

Generated by irclog2html.pl Modified by Tim Riker to work with infobot.