IRC log for #asterisk-dev on 20140109

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06:56.26charewhat does google voice ending xmpp mean for asterisks?
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07:03.04chareis anyone alive here?
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12:41.50*** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Asterisk 12.0.0 now released. Do the release dance! -=- Follow on Twitter at @AsteriskDev
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12:42.59*** topic/#asterisk-dev is Asterisk Development Discussion -=- http://www.asterisk.org/developers -=- Tier 2 and 3.14159265 support is in #asterisk -=- Asterisk 12.0.0 now released. Do the release dance! -=- Follow on Twitter at @AsteriskDev
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13:08.58hexanolI need an ao2_container which is used only to link / unlink objects and apply a callback to all objects in it
13:09.07hexanolI was wondering what was the "best" parameter to create such a container
13:09.11hexanolI tought about "ao2_container_alloc(1, NULL, NULL)"
13:09.22hexanolIs it ok ? Should I use more than 1 bucket ? Should I use a hash function that hash on the pointer address ?
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14:10.32oejHello folks!
14:13.12sruffellwaves
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15:10.40opticronX-Rob, are you xrobau on jira? I've got a patch on ASTERISK-23081 that could use some testing
15:11.39Qwellopticron: he is, and he's sleeping (probably.  he's a bad Australian)
15:11.52opticronah
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15:30.20wdoekeswow, apparently my manager didn't have 5038 because of a broken SHELL() command yesterday (after which asterisk had been restarted).. sh -c grep \"^+31... holds the listening socket
15:30.50wdoekesquite unexpected, as it should've been closed on exec/fork
15:31.41mjordanweird.
15:36.57wdoekeswait, not so weird, as popen() doesn't get the 'e' flag
15:39.28wdoekesor no.. that applies to its new fd's, not all fd's of the forked child
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16:22.15fileopticron, much nicer.
16:34.08opticron:D
16:34.49opticronhopefully there's not too much to rage on
16:35.09opticronthe infrastructure is now the vast majority of that patch
16:36.23filenothing major
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17:10.54mjordanfile: looking at your multi-homed routing patch, my first thought was: gee, we really do need a new module loader.
17:14.50filemjordan, and a single apple.
17:16.03mjordanfile: what is the effect of having the multi-homed module loaded with res_pjsip_nat? In particular, say I have Asterisk behind a NAT and it has two interfaces (who knows why, but let's just pretend). On transmitted messages, what IP address gets placed in, say, the Contact header?
17:16.26fileNAT occurs after multi-homed
17:16.54mjordank, so we'd start off by replacing it with whatever route won out in multi-homed, then go through and blow them away again if we needed to
17:17.00fileyes
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18:40.24tm1000its too quiet in here….file mjordan break something please
18:40.48filebreaks reality
18:40.54opticronnooooo, I need that!
18:43.01jsmithReality is an illusion -- just look at "reality" television :-p
18:43.53sgriepentrogSo on my sip_one_legged_transfer test failure, I compared the full log with a passing log that jbigelow gave me, and on mine it just never gets the replaces invite.  I've rebuilt pjsua per puppet's procedure, no go.  I'm down to trying ubuntu and an earlier copy of pjprj
18:45.16jsmithmjordan: Did you ever approach the pjproject folks about https://github.com/asterisk/pjproject/issues/3, perchance?
18:45.27jsmithjust needed the opportunity to use the word "perchance" today
18:46.31filehe hasn't yet afaik
18:46.42jsmithOK
18:46.56jsmithMeanwhile, Linux distros just keep on rolling...
18:47.13jsmithreturns to his regularly scheduled programming
18:48.08mjordanjsmith: nope, dropped the ball on that one. I'll get something together on it this week
18:55.03sgriepentrogFinally success: answer is current svn.pjsip.org code will not work to build pjsua, have to use older version
19:00.37sgriepentrogWhen you think you've tried everything, the one thing you haven't yet tried is invariably the answer.
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19:04.28mjordansgriepentrog: weird. Hopefully we didn't screw up their build system for that.
19:04.59mjordando you know what the issue is between the two versions?
19:05.13sgriepentrogNot sure it was build - at least, the script I used was the same just switched to the tarball that puppet uses.
19:05.42sgriepentrogI.e. it has always compiled, just wouldn't issue the Replaces invite.
19:06.08sgriepentrogI can diff them and try to determine...
19:07.03sgriepentrogOr I could just rig a script to sneak backwards through git commits until it works again.
19:09.07mjordanhm.
19:09.14mjordanMight be worthwhile reporting the issue up stream
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19:15.48X-Robopticron: I'm here. Patch me up!
19:16.11opticronX-Rob, it's on the issue https://issues.asterisk.org/jira/browse/ASTERISK-23081
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19:21.24X-Robopticron: ooh yeah I saw that, thanks
19:21.44X-RobI was using our RPM, but I'm now able to play with source builds again
19:21.51mjordanX-Rob: if you're up for testing...
19:21.56X-Robmjordan: always.
19:22.01mjordanX-Rob: there's a patch up on review board right now as well for multihomed support
19:22.07X-RobOH. MY. GLOB.
19:22.12mjordanit's an entirely new module :-)
19:22.17X-RobWow.
19:22.25mjordanit adds a bit of extra processing in Asterisk, so file stuck it there so if you don't need it, you don't have to load it
19:22.47mjordanhttps://reviewboard.asterisk.org/r/3102/diff/raw/
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19:28.49mjordansgriepentrog: do you have an update on the valgrind support for the test suite? I really, really don't want to check in the PEP 8 changes until you get yours in :-)
19:29.50mjordanand before it gets too late in the day, can I get some love on https://reviewboard.asterisk.org/r/3107/ and https://reviewboard.asterisk.org/r/3099/? :-D
19:31.00opticronI need new eyes after going through the PEP8 changes...
19:31.18putnopvutopticron: more power to you
19:31.22opticronat least it went faster after the 6th page
19:31.35sgriepentrogmjordan: I was presuming to update valgrind support after resyncing it with your pep8 changes.  I don't have a *lot* of work to do on it, but was still treating this as back-burner.
19:33.05tzafrirsruffell, you need -maxdepth for find
19:33.33tzafrirThere's also 'setopt -s nullglob' if we want to assume bash
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19:36.00sruffellyikes..I opened a can of worms.  I should have just left it the way you had it originally.
19:36.37sruffellalthough, if we can stay away from bashims as much as possible I'm all for it.
19:37.10sruffellso…what's your opinion about the best way to handle this?
19:37.26sruffellstick with grep -v '*' or a variation of find?
19:39.01pabelangerjsmith, Have you had any transfer deadlocks with app_queue in the past?
19:40.58jsmithpabelanger: deadlocks, no.  Crashes, yes.  See ASTERISK-20226
19:41.24jsmithpabelanger: But thankfully, putnopvut wrote us a patch and we haven't had any crashes in the past 14 months
19:41.46jsmithpabelanger: FWIW, that patch really helped leifmadsen as well
19:42.43leifmadsenno crashes (in that area of the code) since commit
19:42.51leifmadsenwere getting them 1-3 times a day depending on server and load
19:42.51putnopvutI think I remember that one. That's the one where I adjusted the refcounting in SIP attended transfers.
19:43.19pabelangerjsmith, Ya, not running that one... yet,
19:43.22sruffelltzafrir: also, just FYI, how do you handle a system with a configuration already in a place, then adding a new device and re-running dahdi_genconf?  It seems like I have to blow away the existing configs in /etc/dahdi or hand run dahdi_assign_spans before running the generator since the presence of the /etc/dahdi/span-assignments prevents the new spans from being assigned.
19:43.25pabelangerlooks like I just ran into ASTERISK-19639
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19:43.54jsmithpabelanger: If you're not running that patch, do so now... it'll save you *all* kinds of headaches (related to memory corruption)
19:43.54pabelangerlet me backport ASTERISK-20226 and get some testing on it
19:44.02pabelangerjsmith, ack'd
19:44.04jsmithpabelanger: Backport to 1.4?
19:44.08pabelanger1.8.7.1
19:44.24jsmithpabelanger: Seriously?
19:44.33jsmithcan't figure out why people are running such old versions of 1.8
19:44.45pabelangeryup
19:44.46tzafrirsruffell, not completely blow away. The theory is: you configure it as you please, and then you run dahdi_genconf, which records the existing assignments
19:44.55pabelangerjsmith, stable for us
19:44.58tzafrirI can't think of anything better
19:45.10pabelangerhowever, we are now moving into 12 for our current development
19:46.04sruffellhmm…so it's more like dahdi_saveconf? :)
19:46.27sruffellbut yeah…I've been thinking about how to handle this…I've not had any epiphanies yet except to do what you said.
19:47.27sruffellI was thinking that dahdi_genconf might auto_assign any spans that weren't taken care of by udev?
19:48.00pabelangerjsmith, Oh, I know why I didn't backport it already. We don't use connected line updates, so we are not affected.  However, I'll set some time aside to test and merge
19:49.07jsmithpabelanger: We didn't really use connected line updates, but the patch made a world of difference for us
19:51.27putnopvutYes, the crash tended to happen during connected line operations, but the actual problem lay deeper.
19:51.46leifmadsenjsmith: that overview you gave me that one day of what the problem was might not be a bad idea to add to the wiki for historical purposes
19:52.06leifmadsenya, we don't use connected line appearances really either; the issue for us was large use of AMI and chan_sip together with app_queue
19:54.54mjordanjsmith: ASTERISK-19639 looks to be different than the memory corruption/crash, unless I'm missing something
19:55.30mjordansighs
19:55.33mjordanI really need to scroll up further
19:55.59mjordansgriepentrog: np - if you don't mind losing the commit race on that at least
19:56.22mjordanand yeah, opticron, thanks. Believe it or not, that was my holiday
19:58.45sgriepentrogmjordan: not at all, will make it easier matching up to proper pep8 coding style
20:08.10opticronmjordan, you must really hate holidays
20:08.24mjordanI do. No coal for anyone.
20:12.18X-Robmjordan: any news on the codec issue, since I'm building from source and applying patches
20:12.22X-Rob?
20:15.34mjordanX-Rob: we haven't hit it yet
20:16.22X-RobPoot.
20:16.29mjordansorry :-\
20:16.31mjordanit's high on the list
20:16.50mjordanbut there's some subtle behavior in that's going to take a bit to shake loose
20:17.04X-Robyeah, file said he knew what the cause was
20:17.22X-Robin a global overview sorta way
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20:25.34X-Robopticron: after building from current SVN, I can't duplicate my original error, either.
20:26.07opticronhah
20:26.35opticronI guess I'll just put it up on general reviewboard, then
20:26.45X-Robopticron: sorry 8-(
20:27.13mjordanwell, it's not like there weren't other problems with the CLI command tab completion
20:27.18opticronit's cool
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20:27.28opticronmjordan, I'm pretty sure that's most of what I fixed
20:29.28tzafrirsruffell, sorry: was away. saveconf to where?
20:30.13tzafrirdahdi_genconf's handling of system.conf is different: it is completely generated and not saved
20:31.03tzafrirI wonder if it's possible to have a dumpconfig for system.conf
20:31.38tzafrirWhat information is missing in sysfs?
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20:34.10mjordanpabelanger: are you going to take the patch on ASTERISK-19639 and put it up for review?
20:34.50sruffellI think the channel assignments for FXO / FXS.
20:34.56sruffellBut I think we can change that.
20:35.07pabelangermjordan, as soon as I find out how to apply it against http://svnview.digium.com/svn/asterisk/branches/1.8/apps/app_queue.c?r1=378663&r2=378662&pathrev=378663 I will
20:35.14pabelangerthat code patch changed
20:35.19pabelangerpath*
20:35.40pabelangerTrying to understand how the patch fixes the issue
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20:42.54mjordanI can guess how it fixed the issue in that version of 1.8, but I doubt it's still a problem. During the transfer, we're locking the channel, followed by the queue. Without his patch, we hold the queue lock, and have gone off to get the device state of something. This requires locking the channels container and the channel, causing an inversion.
20:43.06mjordanHowever, we now no longer look at the device state at that point - we merely clear it from the user.
20:43.36mjordanguess we already fixed our shitty code ;-)
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20:44.30pabelangerHmm...
20:44.40pabelangerthen that patch is not the fix
20:44.53pabelangerI think I am hitting the same issue
20:45.07pabelangerit locks look the same, as well as the backtrace
20:45.18pabelangerbut I have the patch from richard applied
20:45.43pabelangerlooks like, queue member does a transfer just as a new caller enters the queue and rings a member
20:45.44pabelangerlock
20:46.35pabelangermjordan, get_queue_member_status() is holding the lock right?
20:48.43pabelangerokay
20:48.46pabelangerI think I see it now
20:48.50mjordanget_queue_member_status is calling into the device state core. If the device state core doesn't have the state of the channel, it will end up finding the channel that it needs to go get the state from - which involves locking the channels container and the channel.
20:49.02pabelangerwe are just moving get_queue_member_status before we do the lock
20:49.13mjordanthe lock of the queue, correct.
20:49.17mjordanHowever, since Richard nuked out the location of that patch, whatever you're hitting sounds different.
20:49.51mjordanwithout a backtrace, however, it's hard to know for sure, since I'm not sure who called get_queue_member_status
20:50.06mjordanbut since you're running a pretty heavily patched version, it may not be in the latest 1.8 either
20:52.08X-Robmjordan: Good news!  http://pastebin.com/dzxcLrqZ
20:52.17pabelangerso, I found this pretty cool script that graphs locks! http://minus.com/i/1LBHcftDwBXu
20:52.28pabelangergenerated from: http://pastebin.com/WBq4DAQq
20:52.46X-RobI can now call between 300 and they work.  HOWEVER. When I do a Dial(PJSIP/300) only one of them rings.
20:52.51pabelangerwhich is from gdb: http://pastebin.com/qP760Qz0
20:53.16mjordanX-Rob: oooo, two devices registered to a single endpoint
20:53.48mjordanX-Rob: correct, you'll need https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_DIAL_CONTACTS
20:54.04pabelangerThread 5 and Thread 6 are waiting on each other
20:54.11mjordanit's a convenience dialplan function that does that for you. You can technically do it yourself with a lot of magicry, but it's a bit of a pain
20:55.01X-Robmjordan: thanks. Investigating.
20:55.22fileone day... the core will support it
20:55.50mjordanpabelanger: right, but that doesn't show the call stack in app_queue
20:59.22mjordanpabelanger: that's a different deadlock that you have
20:59.46pabelangerYa
20:59.49pabelangerI see that now
20:59.55X-Robmjordan: ok. That's an awesome function. Tell me why I shouldn't be using that for EVERY PJSIP Dial command? Because I think I should be?
21:00.05pabelangerlocks a little further down
21:03.00mjordanits also happening between your AMI execution and the attended transfer. Since you're patched, though, I'm not sure why. The AMI execution doesn't appear to grab a channel lock in 1.8, and locks things in the queue -> member order, which is what update_queue also does (called from the fixup). If we attempted to grab a channel lock in the AMI action I'd expect bad things, but I don't see that there.
21:03.08mjordanX-Rob: you probably should be
21:03.26mjordanunless you want the granularity of dialing only a single PJSIP contact
21:03.28mjordanfor some reason
21:04.09mjordanand it is only really needed if you have multiple devices registering to an endpoint. Otherwise, it's not needed.
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21:06.06X-Roband the only way I can tell if multiple devices are registering to an endpoint is by..checking to see if the dial string returned by PJSIP_DIAL_CONTACTS contains more than one result.. Yeeeeah I'm just gunna go ahead and use that for every PJSIP Dial string.
21:06.32mjordanpabelanger: disregard that
21:07.07mjordanpabelanger: your deadlock isn't that. It's between the masquerade fixup and chan_sip. I'm 99.9999% sure we fixed that.
21:07.20pabelangermjordan, ok
21:07.38mjordanthere were a lot of patches that went in and fixed a number of masquerade/chan_sip channel allocation issues in the high teens of 1.8
21:07.39pabelangerchecking chan_sip svn log
21:07.44mjordangod speed sir
21:07.48mjordan(upgrade!)
21:07.53pabelangerAm
21:07.54pabelanger12
21:07.59mjordanballsy
21:08.03pabelangerOnce I replace app_queue
21:08.04pabelanger:D
21:08.54pabelangermjordan, actually, 1.8.7.1 has been very good to me. Aside from a few backport jsmith and leifmadsen run, we haven't had any issues
21:09.17pabelangerthis properly is because the client changed out polycom / zoiper with an unknown sip stack we have never tested with
21:09.18mjordanX-Rob: I'd probably do that, personally
21:09.36X-Robmjordan, file -- with that patch I can now have two devices on different subnets register and both can make and receive calls. This is awesome. However. What if I want to have 300 as a desk phone, but I also want them to be able to log in via websockets and make/receive a call via a web page?  I still can't bind an endpoint to two transports, correct?
21:09.51filedon't specify a transport
21:10.04mjordanfile: how would it know the local_nets then?
21:10.40fileit'll end up using the UDP transport if it's for UDP, or it'll use the open WS connection if it's for that
21:11.00mjordanX-Rob: you guys are going to go with a single UDP transport definition, right?
21:11.24mjordanI think tm1000 might have mentioned that
21:11.26X-Robmjordan: with this patch, I'm going to rip out all the work I did for multiple per-interface transports
21:11.33mjordansorry
21:11.46fileit'll work either way, just don't specify an explicit transport
21:12.09X-Robhttp://i.imgur.com/FSY85bl.png
21:12.11fileI foresee a wiki page in my future
21:12.21X-RobI hated that
21:12.25X-Robso I'm happy to throw it away
21:13.00mjordanfile: I see that too!
21:13.08X-RobI remember reading something about endpoints picking the first transport in the file..
21:13.44tm1000change is good
21:14.02X-RobFor example, endpoint's "transport=" option, if no value is assigned then Asterisk will *DEFAULT* to the first configured transport in pjsip.conf which is valid for the URI we are trying to contact.
21:14.13X-RobAh. I read incorrectly.
21:14.28fileres_pjsip_multihomed changes things
21:14.56tm1000X-Rob: wanna change pjsip dial to dial contacts then?
21:15.06tm1000X-Rob: (really how much work is that though)
21:15.14fileACHOO
21:15.16X-Robtm1000: Trivial.
21:15.35tm1000X-Rob: lets do it, as users are expecting this feature to work in 12 as it was touted by people named ward mundy
21:15.51tm1000so if it doesnt work for a technically then people will be confused
21:16.01tm1000plus it gives us more ammo for destroying DU mode someday
21:16.13tm1000X-Rob: want a ticket for it?
21:16.18X-Robtm1000: Sure.
21:20.05tm1000file: mjordan with all of these changes when is 12.0.1?
21:20.20tm1000as I am weary if freepbx should just not work with 12.0.0 at all at this point
21:23.03pabelangermjordan, thanks for looking and the pointers. I have a few potential patches to now test
21:23.23mjordantm1000: I'd say whenever you guys want
21:24.20mjordantm1000: it's not a lot of work for us to create a tarball. I mentioned it to tony; I'd say when you guys are happy, we should roll a tarball so we have a line in the sand for all of the issues you've found
21:24.34X-Robmjordan: +1 multihome
21:24.39tm1000mjordan: maybe tomorrow then?
21:24.48X-RobI built from SVN and things seem better
21:24.50tm1000I want bryan (GameGamer43) to roll a new RPM so we can check
21:24.55X-RobI'd love for the codec issue to be fixed tho 8-(
21:24.59tm1000and then X-Rob and I can be on RPM as well
21:25.04tm1000X-Rob: g722?
21:25.07mjordanyeah, I'm with X-Rob - there are a few patches in flight
21:25.35X-Robtm1000: it's actually more than just g722, turns out it's a ball of wibbly-wobbly codecy ... stuff.
21:25.43mjordanalong with a few other things that may be worth us fixing. But if it's giving you a headache, I'll roll one
21:25.54tm1000mjordan: no we can hold off
21:25.55mjordanshakes his fist at format_cap.c
21:26.14tm1000this is another reason we are hiding the release from the public, except for digium employees and our own
21:26.19*** join/#asterisk-dev otherwiseguy (~otherwise@2605:a601:437:ce01:2677:3ff:fe8f:2730)
21:26.23tm1000I can only imagine the headaches we would have already had
21:26.25tm1000:-p
21:26.39mjordanit's been a productive couple of weeks, I'll just say that :-D
21:26.40tm1000and then the useless bug and dup bugs you'd all have
21:27.32GameGamer43tm1000: I can roll an rpm tomorrow morning if you want
21:27.58tm1000G+1
21:28.04tm1000GameGamer43: +1 as well
21:29.27pabelangermjordan, actually, if you look at http://svnview.digium.com/svn/asterisk?view=revision&revision=331774
21:29.51pabelangerthere is a commit to explicitly unload the queue before calling update_queue
21:30.02pabelangerhowever, in queue_transfer_fixup() we don't do that
21:30.14pabelangerI'm curious if that is an oversight
21:30.46pabelangers/unload/unlock
21:30.54mjordanyou can't unlock the channel there.
21:30.55mjordanso. Hm.
21:31.13pabelangerokay, maybe not
21:31.24mjordanNot saying it isn't a problem.
21:31.42pabelangerthe lock is in try_calling()
21:32.01mjordanI'm not sure what agent list lock Matt is referring to there
21:33.21mjordank. I'd file a bug for that then. But to be frank: that's going to be a beast to solve correctly. You *cannot* unlock a channel during a masquerade: that would lead to all sorts of horror shows.
21:33.32mjordanwhich means the fixup has to find another way to update the queue, which should be entertaining.
21:34.24pabelangerokay
21:35.07danjenkinsin new pjsip configuration, where are you meant to specify the mailbox? in the aor type? or the endpoint type? I'm a little confused
21:35.25danjenkinsthe file seems to suggest endpoint but the wiki seems to suggest the aor
21:35.51danjenkinsi did see a jira saying the docs werent up to scratch ;)
21:36.52putnopvutdanjenkins: it depends on if you want solicited or unsolicited MWI notifications.
21:37.12danjenkinsoh right
21:37.13putnopvutIf your device sends a SUBSCRIBE in order to get MWI, then put the mailboxes in the aor
21:37.25putnopvutIf you want to get unsolicited MWI notifications, then put the mailboxes in the endpoint.
21:37.44danjenkinsdigium phone? Think it does it as a subscribe
21:37.51putnopvutdanjenkins: I believe you're right.
21:37.57danjenkinsthanks putnopvut
21:38.37pabelangerdanjenkins, so, what timezone you have to work in?
21:38.41pabelangerCST?
21:38.54danjenkinsi work in gmt
21:39.21leifmadsenwhat offset?
21:39.33danjenkinsits 6 hours from CST
21:39.40leifmadsenso no offset :)
21:39.47danjenkins?
21:39.57pabelangerdo you start at 9am GMT or 2pm GMT
21:40.20danjenkins9 GMT, well im actually starting at 10 but you get my point
21:40.23pabelangerto match the guys in huntsville
21:40.28pabelangercool
21:40.36danjenkinsstarting at 10 to get a bigger overlap
21:40.47pabelanger.notbad
21:41.06danjenkinsyeah seems to work well
21:41.47mjordanputnopvut: ironically, we have an issue to clarify that in the documentation
21:43.31danjenkinsmjordan: i found it earlier while trying to figure out what was right and wrong, and went "well that doesnt help"
21:44.13pabelangerOn a side note, are Changelogs formatting is terrible.  Going to find that patch I started a while back to make it more human readable.
21:45.26mjordanpabelanger: I was thinking about that too recently. Are you updating svn2log?
21:48.47*** join/#asterisk-dev [TK]D-Fender (~chatzilla@64.235.216.2)
21:50.55ipengineerAre there any known issues in Asterisk 12 with bridging a forwarded call? I am having issues with no RTP being passed when taking an inbound call from PSTN and turning around and trying to send it back out to the PSTN
21:52.20ipengineerIf I look at bridge show all it says it is a basic, native_rtp bridge
21:54.35mjordanuhm. You said PSTN. Are the channels DAHDI?
21:54.57ipengineermjordan: No it is all SIP Trunks..
21:55.10mjordank. Just checking.
21:55.36ipengineerNo prob.. What is weird is I have no problems with RTP anywhere else.. Only when trying to bridge calls in this scenario
21:55.49mjordanSo the answer is no, I'm not aware of any issue. A native_rtp bridge is either going to be doing packet 2 packet bridging between two RTP capable channels drivers, or is going to also support remote bridging, i.e., direct media
21:56.16ipengineerThat may be the issue right there..
21:56.34ipengineerOn my peers I have direct media set to off.. How would I turn direct_media off on chan_sip for this scenario since there is no peer
21:56.53filedirectmedia=no
22:00.23*** join/#asterisk-dev otherwiseguy (~otherwise@2605:a601:437:ce01:2677:3ff:fe8f:2730)
22:00.59ipengineerNah.. Didn't seem to help. That is on chan_sip since for the life of me I couldn't figure out how to get Flowroute working with PJSIP. Everything else is running on PJSIP well
22:06.51*** join/#asterisk-dev vlad_starkov (~vlad_star@109.188.127.107)
22:07.53fileipengineer, do you have a SIP trace (chan_sip and PJSIP) you can pastebin? does rtp set debug on show anything as well?
22:08.15ipengineerfile: Let me see what I can dig up
22:14.30danjenkinshas anyone chucked asterisk logs into loggly before? Looking for a ready made filter...
22:17.03ipengineerfile here is a sip trace: http://gist.github.com/zconkle/3cc800d67a70e5bca67a
22:18.01fileyour chan_sip is not configured for your NAT
22:18.45mjordanipengineer: I know it isn't FlowRoute, but rnewton wrote some documentation on the Asterisk wiki about setting up a trunk with an ITSP here - https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip+to+work+through+NAT
22:19.24[TK]D-Fender<PROTECTED>
22:19.31[TK]D-Fendernot a different tech?
22:19.36filehe's using chan_sip
22:19.47[TK]D-Fenderthat's what I wa thinking
22:20.15[TK]D-FenderContact: <sip:12149601082@192.168.3.212:5069> <- and here's where we see that NAT setup was done wrong
22:21.46ipengineer[TK]D-Fender: will extenip in sip.conf fix that?
22:22.07[TK]D-Fenderif you set everything else that is part of this right... it'll do it's part
22:51.05wedhornGetting an unused variable "i" in chan_dahdi with dev-mode (only). Seems to do with "i" being declared outside ifdef in dahdi_create_channel_range (r396103)
22:51.22wedhorn^^^ ast12 and trunk
22:53.23mjordanwedhorn: interesting... what version of gcc?
22:53.32mjordanlooks
22:54.08mjordanah
22:54.12mjordanyup. HAVE_PRI.
22:54.41mjordanwedhorn: if you want to fix that's fine, otherwise I'll make a quick issue so we don't forget about it
22:55.24wedhorneasy to fix, but with out knowing the logic, not sure whether to move i to within or remove/move the ifdef's?
22:55.34mjordanmove i into the range
22:56.09mjordanhonestly, both i and x should probably be in that range. Hm.
22:56.41wedhornyep, and the for loop
22:56.50wedhornfor x
22:57.01mjordanthis is new in 12...
22:58.17mjordanah. This is part of tzafrir's patch, I think
22:58.50mjordanyeah.
22:58.58mjordanr396474
22:59.27mjordanwedhorn: I'd make a new block scope and put i and x inside of it, since the for loop is pointless without PRI. That's me, personally.
23:00.59*** join/#asterisk-dev ipengineer (~zconkle@static-71-252-134-64.dllstx.fios.verizon.net)
23:02.27wedhornokay, and coding style? Just put { on a line by itself?
23:02.30pabelangermjordan: yes, I had something going before. Just need to find it
23:03.25snuff-workwedhorn: =)
23:03.43mjordanwedhorn: pretty sure you have to for a new block scope
23:03.57wedhornokay, cool
23:04.15wedhornhey snuff-work, how's life treating you
23:04.43snuff-workbusy.. wanting to do more stuff with skinny @ work but haven't had the time
23:05.19snuff-workmidst of deploying 50 servers over 10 blade chassis.. keep me busy for the next lil while too
23:06.35wedhornsounds like you've got your work cut out
23:06.42*** join/#asterisk-dev vlad_starkov (~vlad_star@109.188.127.172)
23:07.11snuff-workyer.. lucky most of the 'hard' part is done.. golden set of images..
23:07.48snuff-workstill bits of manual to do for each
23:08.07snuff-workhow's things with u ?
23:09.20wedhornyeah pretty good. Moved to Brisbane, enjoying the weather (apart from the odd day) and being a kept man. Not sure how much longer that'll last though ;)
23:10.44snuff-work;)
23:36.39fileI'm adding "silly use of RAII_VAR" to my list of things that would take collateral damage when I'm fixing nearby code
23:36.51fileor rather, will take damage
23:37.00mjordanfile: +1
23:37.29mjordanwe have a very nice screwdriver. Some things are nails.
23:38.21fileuses RAII_VAR on mjordan, it's super effective!
23:38.41mjordancrap. I think when I leave my office, that means I'll be auto-destructed
23:38.43mjordanNOOOOO
23:38.52fileyou're welcome
23:41.12mjordanfile: http://shirt.woot.com/offers/what-are-those-foxes-saying
23:41.25fileI approve
23:44.38*** part/#asterisk-dev mjordan (~matt@nat/digium/x-zqlyksvxwtrlbllb)

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