IRC log for #asterisk on 20120406

00:02.51suedohgot it
00:04.34carrarWHAT
00:14.52*** join/#asterisk GGD (~deberle@pool-173-72-205-188.clppva.fios.verizon.net)
00:18.03*** join/#asterisk RogersJA (~RogersJA@184.70.36.178)
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00:33.02asteriskmonkeyim trying to get the sipcall id from variables in asterisk 1.10 ive tried ${SIP_HEADER(Call-ID)} and ${CALLID} .. both dont seem to work,anyone know what variable it is?
00:35.48*** part/#asterisk GGD (~deberle@pool-173-72-205-188.clppva.fios.verizon.net)
00:39.16*** join/#asterisk kayatwork (~kayfox@orca.zerda.net)
00:56.16p3nguinThere's no asterisk 1.10.
00:57.17*** join/#asterisk blizzow (~jburns@74.7.49.235)
01:23.18*** join/#asterisk blizzow (~jburns@173-8-237-25-Colorado.hfc.comcastbusiness.net)
01:46.48*** join/#asterisk greenwolf (~greenwolf@mobile-198-228-205-162.mycingular.net)
02:04.19Kobazanyone ever have a problem where audio from one phonecall is bleeding over to another, on a pri
02:04.22Kobazpri to sip phones
02:08.33din3shbleeding over?
02:11.18Kobazyeap
02:11.22Kobazlike wires crossed
02:11.44Kobazi think it's memory corruption
02:12.25Kobazlike someone gets a call, can't hear the person, but someone else sitting next to them can, like the pointer to the audio stream is bad
02:14.07din3shasterisk version?
02:19.59*** join/#asterisk greenwolf (~greenwolf@mobile-198-228-205-162.mycingular.net)
02:22.47*** join/#asterisk gonewage (~gonewage@sg720.meridiantitle.com)
02:23.40Kobaz1.8.5
02:23.48Kobazbut it's random
02:23.59Kobazdid a restart, lets see if this fixes it
02:24.12Kobazi have a daily restart but i didn't restart it today
02:27.21din3shE1 or T1?
02:36.25Kobazt1
02:37.08Kobazi seriously doubt it's a carrier issue because this has happened before and restarting asterisk or using a different version makes it go away for a while
02:37.17Kobazit tends to happen when ast has been running a while
02:37.55p3nguinEven with SIP session timers, I still have that one $@*#!%& channel that died and stuck permanently again.
02:38.21p3nguinHow the hell can I get rid of stuck channels without a restart when they do not respond to conventional method of death?
02:44.32ChannelZTry bribing them with cookies.
02:47.07*** join/#asterisk albertoandrade (~albertoan@186.206.5.67)
02:47.24MaliutaChannelZ: how are bits of data on their hard drives going to be a form of bribery? ;)
02:51.58*** join/#asterisk golikwid|mac (~chrislees@64.45.192.200)
02:52.14Kobazp3nguin: deadlock?
02:52.38*** join/#asterisk LiuYan (~liu.yan@211.154.128.135)
02:52.43Kobazp3nguin: core show locks
02:54.08ChannelZnever heard of that one
02:54.45ChannelZ(or have it)
03:01.37*** join/#asterisk jsjc (~Adium@134.147.78.188.dynamic.jazztel.es)
03:07.21*** join/#asterisk blizzow (~jburns@67.50.165.60)
03:10.03p3nguinI thought of that the other day, but asterisk apparently isn't compiled for showing locks.  It's AsteriskNOW.
03:10.39Kobazi always build with DEBUG_THREADS
03:13.20p3nguinIt just keeps happening almost every day, so there has to be something causing it, and probably a way to prevent it.
03:14.23Kobazit's a deadlock most likely
03:15.11Kobazif you can't rebuild the asteirskes, kill it with sigquit (abort and dump core)
03:15.12p3nguinBut why does it keep happening?
03:15.20Kobazand then you can backtrace it
03:15.33Kobazit happens because A is waiting for B and B is waiting for A
03:15.37Kobazand it will never finish
03:16.33p3nguinI was describing exactly what happens earlier, but it will be easier to just say it again as opposed to having you scroll back, I guess...
03:17.14Kobazmm, how much earlier
03:17.45p3nguin11 hours
03:17.49Kobazoh heh
03:18.02Kobazmy scrollback buffer doesn't go back that far
03:18.38p3nguinI make a call from my office through my asterisk to another asterisk in another office.  I can stay on the call for always a random amount of time before it eventually goes silent.  The phone never disconnects.
03:19.29p3nguinOnce the call goes silent, I can stay on for infinity and it never comes back, so I will eventually hang up my phone.
03:19.58Kobazfirst thing is to compile with DEBUG_THREADS
03:20.01p3nguinLike 30 seconds or so after I hang up my phone, the remote asterisk throws a message about auto-destroying a dialog.
03:20.10*** part/#asterisk gonewage (~gonewage@sg720.meridiantitle.com)
03:20.12Kobazand then core show locks once you have the problem
03:20.29p3nguinBut the channel remains.  Always, and forever.
03:20.52Kobazsounds like a good ole fashion deadlock
03:21.03p3nguinIt does not respond to channel request hangup.  It does not die in a fire.  It just stays there.
03:21.10Kobazdeadlock
03:21.38p3nguinI've managed to turn it into a zombie before using combinations of channel redirect and Bridge() and whatnot, but even then the zombied channel remains forever.
03:22.14Kobazthe only way you'll have any idea is by core show locks, and then a kill -sigquit to get the backtrace of every thread
03:22.24p3nguinI made sure I was using SIP session timers, thinking that might control it better, but it did not help at all.
03:22.59Kobazdoing anything other than that is wasteful and speculative
03:23.44p3nguinSo I'll have to get qwell to give me the spec file or srpm or whatever he uses to build asteirsk for asterisknow, and build a new package with thread debugging enabled.
03:24.29p3nguinI'm not about to go building my own package for it all willy-nilly.  It's a production box, so failure is not an option.
03:25.02Kobazif you can get to it off hours and break it and then sigquit it, it would get some possibly quite valuable info
03:25.27Kobazthat by itself might be able to get enough about the problem without a debug threads rebuild
03:25.38p3nguinThe channel is stuck right now, but I don't have core show locks.
03:25.52p3nguinI'm on the box.  What would you like for me to try?
03:25.52Kobazis there any traffic on it that you care about?
03:25.55p3nguinThe office is closed.
03:26.16p3nguinAs long as I don't crash the box, I don't care what else happens.
03:26.30p3nguinAsterisk can go down, as long as it is up by 9 AM.
03:27.00Kobazdo you have core dumping turned on?
03:27.19p3nguinI have no idea.  How would I be able to tell?
03:27.30Kobazif you didn't set it up, then probably not
03:27.39Kobazbut it might dump core anyway
03:27.42p3nguinHow does that get set up?
03:27.48Kobazit's in /etc/init.d/asterisk
03:27.59p3nguinLet me look at the file.
03:28.40KobazAST_DUMPCORE="xxx"
03:28.48Kobazif it's yes, then you're in good shape
03:29.37p3nguinI don't have that variable in the init script, but, however, asterisk is running with the -g option.
03:30.00Kobazoh okay
03:30.01Kobazperfect
03:30.09Kobazps aux | grep asterisk
03:30.15Kobazkill -QUIT <asteriskpid>
03:30.20p3nguinI could have told you that before if I knew that's what we were looking for.
03:30.22Kobazrestart asterisk
03:30.31p3nguin# ps -C asterisk u
03:30.32p3nguinUSER       PID %CPU %MEM    VSZ   RSS TTY      STAT START   TIME COMMAND
03:30.32p3nguinasterisk 16415  0.0  1.6  38404 20460 ?        Sl   Mar22  12:18 /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c
03:30.50Kobazoh
03:30.55Kobaz-G != -g
03:31.00Kobazit still might force it though
03:31.07Kobazi'll do a little test
03:31.08p3nguin-vvvg   <----
03:31.13Kobazoh i see that
03:31.17Kobazoh okay perfect
03:31.23p3nguinkill -QUIT 16415
03:31.24p3nguindone
03:31.25Kobazkill -QUIT it
03:31.27Kobazk
03:31.30Kobazstart it back up for tomorrow
03:31.34Kobazand then find your core file
03:31.38Kobazprobably in /var/spool/asterisk
03:31.40p3nguinIt's still running.
03:31.50Kobazrunning with the same pid?
03:31.53p3nguinYes
03:32.17p3nguinIt didn't seem to affect it at all.
03:32.22Kobazhmm
03:32.37p3nguinIf safe_asterisk would have recovered it, I would have thought it would have a new PID.
03:32.39Kobazi think maybe it arms a signal against sigquit
03:32.41Kobazyeah
03:32.44Kobazokay you know
03:32.47Kobazi generally do this from gdb
03:32.59Kobazso i've actually never tried a -QUIT
03:33.03Kobazso
03:33.11Kobazfire up gdb
03:33.17Kobazthen do
03:33.22Kobazat 16415
03:33.31Kobazyeah looks like asterisk immunes itself to sigquit
03:33.32p3nguinI have to install it first.
03:33.53p3nguinOkay, that was fast.  Let me see what it will do.
03:34.29p3nguinI ran  at 16415
03:34.51Kobazdid it say attaching
03:34.54Kobazand print out a boatload of stuff
03:35.33p3nguinYes:  http://pastebin.com/8EPJQfda
03:35.42Kobazokay
03:35.43Kobazdon't need that
03:35.54Kobazclear your screen and write this result out to pastebin
03:36.01Kobazthread apply all bt
03:36.30p3nguin---Type <return> to continue, or q <return> to quit---
03:36.30Kobazkeep hitting enter till you get it all
03:36.49p3nguinDone.
03:37.05*** join/#asterisk blizzow (~jburns@67.50.165.60)
03:37.34p3nguinpastbin the entire output?
03:38.14Kobazyeah the whole thing
03:38.24Kobazi can't believe i ate the whooooooole thing
03:38.50p3nguinhttp://pastebin.com/fLuPhNhy
03:39.11Kobazallrightey what sort of good stuff do we have
03:39.13p3nguinRight near the bottom I see the channel that is stuck.
03:39.38Kobazstuck in chanspy
03:39.48Kobazusing pthread timing i guess?
03:40.30p3nguinI'll have to check.  Can I exit gdb?
03:40.43Kobazyou can but i might want to get some var data
03:40.52Kobazbut while gdb is attached, asterisk is paused
03:41.16Kobazthis is 1.10?
03:41.31p3nguinI was going to leave gdb, reconnect to asterisk CLI, and check the timing.  It is 1.8.11.0.
03:41.39Kobazoh okay
03:41.49p3nguinIs it okay to do that?
03:41.51Kobazyeah
03:41.54Kobazyou can reattach later
03:42.04Kobazwhen it asks to detach an inferior process hit y
03:42.39Kobazso it's not a deadlock, but it's a blocking problem
03:42.49Kobazboth are just as bad
03:43.02Kobazbad blocking can cause deadlocks
03:43.30p3nguinModule                         Description                              Use Count
03:43.33p3nguinres_timing_pthread.so          pthread Timing Interface                 0
03:43.36Kobazyeah
03:43.36p3nguinres_timing_dahdi.so            DAHDI Timing Interface                   3
03:43.40Kobazoh
03:43.47Kobazadd res_timing_pthread to noload
03:43.53Kobazif you're using dahdi, you should just use dahdi
03:44.02Kobazpthread and timerfd are probmatic
03:44.10Kobazi spend like months on working on a timerfd problem
03:44.23Kobazand it's still not truely fixed
03:45.24Kobazswitching to pthread quite possibly will make the problem go away
03:45.29Kobazbut it's masking a bug in asterisk
03:45.30p3nguinOkay, noload pthread completed.
03:45.30*** join/#asterisk RogersJA (~RogersJA@S0106001d7e20d4e4.wp.shawcable.net)
03:45.49p3nguinUnloaded res_timing_pthread.so
03:46.13Kobazyou should restart before the morning though
03:47.11p3nguinI don't want to restart until you're done checking things.
03:47.18Kobazyeah
03:47.30p3nguinI may not be able to make it hang with no calls.
03:47.44p3nguinIt always does it when I am spying on the incoming calls.
03:47.53Kobazah okay
03:48.06Kobazthere's been all kinds of chanspy blocking bugs fixed
03:48.12Kobazi wouldn't be surprised that there's more
03:48.23p3nguinAs you can see, I spy on a specific prefix... and it switches channels automatically as calls come and go.
03:48.49p3nguinEventually, it goes silent.  I see calls flowing like normal on the CLI, but I no longer hear anything.
03:48.55Kobazyeah
03:49.02Kobazbecause it's waiting for something
03:49.10Kobazand that something never happens
03:49.29p3nguinSo it's a chanspy bug rather than a problem in my sip channel?
03:49.35Kobazyeah
03:49.46p3nguinI was concentrating on the sip part of it.
03:50.04p3nguinThat would explain how it knew to run an auto-destroy when I hung up the phone.
03:50.30p3nguinIt tried to kill itself, but the app was stuck.
03:50.33p3nguinI never even thought about that.
03:50.47Kobazso it probably has nothing to do with pthread timing actually
03:50.55Kobazbut it's good to only use one timing module anyway
03:51.10p3nguinThat's okay.  dahdi will do just fine without the other one hanging around.
03:51.19Kobazso this is waiting inside ast_audiohook_detach
03:51.28Kobazdahdi is the best bet for timing
03:52.36p3nguinWhen I first put up this box, I didn't even have dahdi at all.  I later installed it just for timing purposes, but never bothered to noload pthread because dahdi has priority anyway.
03:52.56Kobazi don't know the audiohook code at all, so this would be an issue for someone else, but
03:53.12Kobazpaste up this backtrace on jira
03:53.16Kobazit's definitely a problem in the code
03:53.20p3nguinWhen it comes to code, I know very little.
03:53.38p3nguinFile it as a chanspy bug?
03:53.41Kobazyeah
03:53.55Kobazyou can resume asterisk, unless you want to stick around to see if someone knows the audiohook code to help debug
03:54.00Kobazit's so great to get on a live system
03:54.09Kobazbut since it's easily reproducable it should be easy to fix
03:54.23p3nguinIt happens at least once per day.
03:54.33Kobazyeah, i saw that a lot in 1.6
03:54.38Kobazbut some of those have been fixed
03:54.54p3nguinIt happens if I use the prefix only for the ITSP, or even if i use only the SIP prefix.
03:55.08Kobazyeah
03:55.09Kobazso
03:55.11Kobazwhat this looks like
03:55.12p3nguinIt has happened on several versions of 1.8 on this AsteriskNOW box.
03:55.33*** join/#asterisk Dovid (~Dovid@213.8.121.90)
03:55.34p3nguinEach upgrade, I hope it is not going to do it again, but it always does.
03:55.37Dovid<PROTECTED>
03:55.46Kobazthere's a timed audiohook wait, which is the ast_audiohook_trigger_wait function
03:55.55Dovidmorning ev1
03:56.02Kobazand you don't have any channel data to return back when nothing is available
03:56.22Kobazso if nothing triggers your audiohook you'll wait forever
03:56.37Kobazand there's no unwind code for hangup
03:57.12p3nguinI will sometimes sit waiting on a new call for a long time, but usually the app attaches to a new channel and I get audio.
03:57.30Kobazwhat happens if you load up chanspy against a non-existant prefix
03:57.32Kobazand then hang up
03:57.33p3nguinOther times, it just stays quiet and I do not see it say anything about attaching.
03:57.47p3nguinLet me check what it does.
03:58.14Kobazthere's an option to chanspy to hangup when there's no more channels to be watched
03:58.38p3nguinWhen I am monitoring calls, it is annoying to have to keep redialing the spy extension.
03:58.41Kobaz<PROTECTED>
03:58.42Kobazwell
03:58.44Kobazwhat you can do
03:58.47Kobazis loop in dialplan
03:58.53*** join/#asterisk LiuYan (~liu.yan@211.154.128.135)
03:58.59p3nguinSounds like lots of CPU time.
03:59.06KobazChanSpy(prefix,S); Wait(1)
03:59.07Kobaznah
03:59.22Kobazi've set up worse
03:59.35Kobazbut try ChanSpy(badprefix)
03:59.44Kobazand try ChanSpy(goodprefix,S)
03:59.51Kobazand then hang up the spyee
04:00.15Kobazasterisk can take a lot of abuse in dialplan code
04:00.26Kobazjust don't have infinite recursion running MixMonitor
04:00.27Kobazthen you'
04:00.31Kobazll surely crash
04:00.45Kobazi wonder if the laundry is done
04:00.50p3nguinAttaching to aardvark
04:00.51p3nguinhaha
04:00.56p3nguinNow what?
04:01.09*** join/#asterisk gonewage (~gonewage@sg720.meridiantitle.com)
04:01.13Kobazhang up aardvark
04:01.15p3nguinI'm sure chan_aardvark is not loaded.  :)
04:01.20Kobazoh
04:01.21Kobazno no
04:01.24Kobazi mean like
04:01.31KobazSIP/somethingthatdoesn'texist
04:01.35p3nguinah
04:01.42*** join/#asterisk gonewage (~gonewage@sg720.meridiantitle.com)
04:01.43p3nguinchanges to SIP/aardvark
04:01.45Kobazlike something that could be valid
04:01.51Kobazbut is not running just yet
04:02.18Kobazi just wanna see if the chanspy waits forever if it can't find a match
04:02.19p3nguinAttaching to SIP/aardvark
04:02.25Kobazokay now hang up
04:02.29Kobazand see if the channel is still there
04:02.30p3nguinIt's just chillin'
04:02.47p3nguinIt's gone.
04:02.56p3nguinIt never hangs that quickly.
04:02.56Kobazokay so it's not an every-time sort of thing
04:03.11p3nguinIt's an every time thing, but some times it is a long time.
04:03.24p3nguinSometimes 15 minutes, sometimes 3 hours.
04:03.27Kobazevery time as in like every single time you run it
04:03.36Kobazi know you said it happens every day
04:03.49Kobazbut it's different if it happens every day, one out of 100 runs
04:03.52p3nguinLet's say I have nothing better to do than listen to call quality while a type documents and crap...
04:03.59Kobazor you run it every time and it blocks
04:04.14Kobazrunning one all day
04:04.15p3nguinI might load up the chanspy and listen to 100 calls.
04:04.24Kobazand then having that one lock up, is a different thing too
04:04.30p3nguinEventually, I'm not hearing anything anymore.
04:04.34Kobazyeah
04:04.45p3nguinOr another time, I might listen to 3 calls before I notice I'm not hearing anything.
04:04.57Kobazso for tomorrow, try CHanspy(channel,S) on a loop
04:04.58p3nguinBut it seems inevitable.
04:05.01Kobazand see if you still get a stuck channel
04:05.21Kobazif that works
04:05.30p3nguinNow sometimes I can't just sit and listen to calls on this system, so I might hang up before it got stuck.  Those channels always go away correctly.
04:05.37Kobazthen that means there's some code in the path of using 'S' that should be put into the regular chanspy code
04:06.06Kobazif it doesn't work (ie: still stuck), then it affects chanspy as a whole
04:06.46Kobazit helps to have a very specific set of things that cause the problem
04:06.55Kobazor, it's also very good to know, if there's multiple ways to cause the problem
04:06.57p3nguinSo you want me to do something like 1,1,ChanSpy(prefix,S) ; 1,n,Wait(1) ; 1,n,Goto(1) ?
04:07.05Kobazyeah pretty much that would do it
04:07.12Kobazit will be very little load on the system
04:07.22Kobazyou could put a wait(5) in there if you're paranoid
04:07.22p3nguinI'm going to test it right now.
04:09.24p3nguinIt fills the console with ChanSpy()s, Wait()s, and Goto()s... but I can shut verbose down while testing it tomorrow.
04:10.30p3nguinWith no active calls, it is just cycling away.  Only raised CPU usage about 0.5%.
04:12.00Kobazyeah
04:12.08Kobazit's gonna just do it's thing until it matches a channel
04:12.12*** join/#asterisk troyt (~troyt@2001:1938:240:3000::3)
04:13.40Kobazi think a good idea is for every single function everywhere that's related to operating on channels, to take a channel pointer so it can check for hangup
04:14.40p3nguinI'm not sure if I have ever lost audio on an active spy.  There is a possibility that the channel gets stuck in between the end of one call and the start of the next.
04:14.58*** join/#asterisk RogersJA (~RogersJA@S0106001d7e20d4e4.wp.shawcable.net)
04:15.04Kobazcould very well be
04:15.10Kobazthat's the same thing i saw in earlier versions
04:15.31p3nguinThis S option may prevent it from ever hanging, but it might not really prove anything.
04:15.38Kobazwhen you post up your jira ticket give me a poke and i'll put a watch on it.  I use chanspy for several things and this would affect me potentially
04:15.41Kobazwell
04:15.56Kobazif it locks up also, it proves that there's more than one way to break it
04:16.13Kobazif it doesn't lock up, then there's a decent workaround for the problem
04:16.30p3nguinShould I file it under the asterisk project since it has nothing to do spefically with asterisknow, or should I file it under asterisknow since that is where I am experiencing the problem?
04:16.38p3nguinspecifically
04:16.55Kobazdo you know if asterisknow makes any code changes?
04:16.55*** join/#asterisk coppice (~chatzilla@123203240234.ctinets.com)
04:17.15Kobazor do they bundle a vanilla asterisk with their gui
04:17.30p3nguinThat isn't going to be an acceptable workaround because it fills the CLI when verbose is up (and it is normally up if I am spying and watching calls).
04:17.48p3nguinAs far as I know, it is pure asterisk, just packaged.
04:17.50Kobazyou can write your log out to a file and grep -v your chanspy
04:18.06Kobazokay, throw it into a chanspy bug then
04:18.21p3nguinBut asterisk or asterisknow was the question.
04:18.25Kobazasterisk
04:18.37Kobazif it's certain the asteirsknow people don't monkey the source code
04:18.56Kobazthen they won't have any way to fix it since I would think many of them are the gui programmers
04:19.21p3nguinIt's just qwell and a CentOS box, as far as I know.
04:19.32p3nguinHe packages all the stuff.
04:19.40Kobazk
04:20.06Kobazwell i know qwell works on the asterisk code
04:20.44Kobazworst comes to worst the issue gets bumped over to asterisknow
04:20.49Kobazwhich seems unlikely
04:22.51Kobazman this gluten free cereal is awesome
04:27.50p3nguinWhat you had me do in gdb wasn't a backtrace... so what would I call it?  Just refer to it as a gdb trace?
04:27.57Kobazbacktrace
04:27.58p3nguingdb dump?
04:28.12Kobazbt is backtrace
04:28.16p3nguinWe didn't do bt.
04:28.20Kobazbt will do the active running thread
04:28.27Kobazthread apply all bt  will do a backtrace on all threads
04:28.30p3nguinWe attached to somethign.
04:28.40p3nguinOH!  I forget you had me run that.
04:28.44p3nguinmy fault.
04:28.46Kobazattached to asterisk and did an all threads backtrace
04:29.07Kobazdid you restart asterisk yet?
04:29.08p3nguinI just copied/pasted what you typed, so I totally forgot I started bt at all.
04:29.28Kobazif not you can do a more detailed bt
04:29.33Kobazwhich is sometimes helpful
04:29.51p3nguinI have not restarted yet.
04:29.59Kobazgood good
04:30.01Kobazattach again
04:30.06Kobazand do thread apply all bt full
04:30.23Kobazthe regular bt is easier to look at
04:30.27Kobazbut the full bt has much more data
04:31.37p3nguinOkay, it's done.
04:31.46p3nguinI'll paste this into the bug report.
04:31.51Kobazgood good
04:32.41Kobazholy crap, 3tb drives are still like $350
04:32.50p3nguinI don't need to include the output between the 'at' and the 'thread apply all bt full' do I?
04:33.02Kobaznope
04:33.09Kobazjust the function traces
04:33.17Kobazthat's all local system-specific stuff at the top
04:34.32p3nguinASTERISK-19644
04:36.22Kobazi think the severity is gonna get downgraded
04:36.41Kobazmajor bugs are like crashes and deadlocks and things that make asterisk totally dead
04:36.47Kobazor really bad behavior
04:37.10p3nguinIt was the default level.
04:37.12Kobazattach the console log too
04:37.24Kobazwhere you said "sip channel being destroyed"
04:37.29Kobazyou can paste that in too
04:37.46p3nguinI'll have to see if I can scroll back that far.
04:38.00Kobazdo you have asterisk write to a log?
04:40.13p3nguinFound the last one at 14:21 (Thursday).
04:47.12p3nguinI added that tidbit.
04:47.31Kobazk
04:47.41Kobazi think that's all i can really get
04:52.58p3nguinI thought of something else.
04:54.23p3nguinThere are usually multiple calls going on concurrently with the same prefix.  I always press * to cycle them manually.  When I lose audio, pressing * to cycle active channels no longer does anything.
04:55.01Kobazyeah because the channel is in that audiohook wait
04:55.12Kobazthere's nothing to trigger an out other than the hook
04:55.36Kobazdefinitely needs a channel hangup check in there somewhere
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04:58.01p3nguinDo I need to add that note about cycling manually, or should that be clear by looking at the backtrace?
04:58.22Kobazpeople can always ignore information they don't need
04:58.48Kobazso whatever you think is relevant, you can paste in there.
04:59.40Kobazbut it would be obvious that anything on that channel will be sent into a black hole since it's stuck waiting on the audiohook
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05:05.09p3nguinInfo added.
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05:29.42din3shmrning all
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06:11.15din3shis there any script for dahdi if E1 is on alarm?
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06:20.56LiuYandahdi show status ?
06:21.04din3shnope
06:21.11din3shi want a script to mail me back
06:21.18din3shwhen dahdi/e1 is on alarm
06:28.30LiuYanjust googled, AMI manager provide an event called Alarm and also an AlarmClear event, maybe youc an write a small script to catch such events and mail to yourself
06:28.45LiuYanlink => http://www.voip-info.org/wiki/view/asterisk+manager+events#AlarmEvent
06:35.25din3shok thanx will have a look
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06:41.30sawgoodIs it possible to have two or more options for "extennotify=" inside of /etc/asterisk?
06:41.46sawgoodI've tried two statements (one on each line), but it did not work as I thought?
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06:47.17schmidtsgood morning
06:48.43din3shmrning
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06:51.21doctorrayupgrades from 1.6.1.2 to 10.2.1; now 'core show hints' shows everyone as idle.. hints on what may have changed?  scratching my head over this one and a sip debug doesn't seem to be much help
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06:53.50schmidtsdoctorray there were a lot of changes to hints and subscribes between 1.6.1 and 10 but one config related comes to my mind do you have enabled the sip setting limitonpeer ?
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08:18.23*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.3.0 (2012/03/29), 1.8.11.0 (2012/03/29), dahdi-linux 2.6.0 (2012/01/04), dahdi-tools 2.6.0 (2012/01/04), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org
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09:08.20adnchello, I'm using asterisk management interface to make a call with a php script. I wanted to ask if there are commands that could give me the status back, for example if the call has been established something like an OK or similar. Can this be done with the management interface?
09:09.30WIMPyIt will tell you about the calls progress in any detail you want.
09:11.11adncwimpy how can I get that information?
09:11.15adncis it in the reply?
09:11.27WIMPyIn Events
09:11.35adncwimpy do you haven an example?
09:11.51WIMPyIf you have them enabled, they will just poor at you.
09:12.12adncthat is far to less information
09:12.16adncanyway thank you very much
09:12.54WIMPyLog in to AMI manually and make a call. You will see lots of events if you have them enabled for yopur account.
09:13.17adncthey can be enabled per account?
09:13.37WIMPyyes
09:13.42WIMPySee manager.conf
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09:14.27adncthis config file only has four lines
09:14.40adncnothing that could tell me anything about events
09:16.08WIMPyThen take a look at the sample config.
09:17.07adncit is ok WIMPy
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09:35.34the_weardcan aneyone here help me with a problem im having
09:35.48the_weardplz my music on hold does not work
09:35.58the_weardi need to install mpg123
09:36.15the_weardbut im running centos 5 64bit
09:40.14rjvvlietthe_weard: is the anything here that might help? http://www.powerpbx.org/content/rhel-asterisk-freepbx-install-guide-rhel-v6-asterisk-v18-freepbx-v29
09:40.45rjvvlietthe_weard: Just a thought, not using 64 bits OS.
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09:48.50WIMPyMaybe it's a better idea to convert the MOH to a more sensible format?
09:51.51the_weardyes i did but it is still not playing
09:53.15the_weardwhen i put someone on hold it is yust silent
09:54.23rjvvlietthe_weard: it the MOH working when not using MP3? you could also have a timer issue
09:55.18rjvvlietthe_weard: But i agree with WIMPy, i'm only using WAV format. better for the CPU ;-)
09:55.22the_weardok how would i go about getting a log to see what is the error?
09:55.44the_weardi use 8k wav files
09:55.56the_weardthe recomended formats
09:56.30the_weardit uploads them they apear in freepbx but no music when the go on hold
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09:58.32rjvvlietthe_weard: set de debug and verose to 5, and see the /var/log/asterisk/ dir usually its the full log file.
09:59.02rjvvlietthe_weard: what version of Asterisk?
09:59.11the_weardim running 1.6
09:59.26the_weardcan't i use asterisk -rvvvv
09:59.50the_weardto log into the cli?
09:59.55rjvvlietthe_weard: mmm, never used 1.6, skipped that version, i know 1.8 had some troubles with the timer module
10:00.57rjvvlietthe_weard: Yep that would set the verbose to 4 but not the debug. i use 'core set verbose 5
10:01.12the_weardok someone told me i need to install mpg123 to get it going but hell i cant get that right
10:01.12rjvvlietthe_weard: and 'core set debug 5'
10:01.39rjvvlietthe_weard: mpg123 should only be needed for MP3 formats, not the native (if i'am correct)
10:01.51the_weardyes
10:02.36the_weardso my command will look like (asterisk set core set debug 5)
10:02.36the_weard?
10:02.41rjvvlietthe_weard: maybe this has usefull info : http://www.voip-info.org/wiki/view/Asterisk+config+musiconhold.conf
10:03.00the_weardlet me check qwikly
10:03.21rjvvlietthe_weard: on the CLI type : core set verbose 5
10:03.33rjvvlietthe_weard: and : core set debug 5
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10:04.08rjvvlietthe_weard: when you connect to the CLI you see a message telling the debug and verbose levels.
10:05.49the_weardok i tryed to install it from source but it kepton asking for a install path and i don't know where it wants to be installed
10:09.24rjvvlietthe_weard: asterisk seems to look for /usr/bin/mpg123 so select /usr/bin/ as install folder.
10:09.53rjvvlietthe_weard: i think its best to read the link i send you of VoIP info.
10:10.31the_weardcool thax let me begin
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11:35.55sekilhi
11:36.12sekilcan I buy hw canceller addon for TE110P ..pci express PRI card?
11:36.34sekilI have the card ..but no EC chip
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12:00.31din3shi suppose u can
12:00.47din3shi buy them with the canceller card
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12:20.48WIMPyHas anyone else had problems with SRTP after a hold?
12:21.13sekilturns out I have TE121
12:21.21sekilso I just have to buy EC card
12:22.28sekilanyone knows if I can use it with * 1.2?
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12:58.55adncis there a propper documentation for events in AMI?
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14:12.19MariusKarthausHi, have a number of phones registered as friend in the sip.conf. I can call them using SIP/phone1. How I want to make a rule in the extensions that will allow a user to dial '2' and the dial should go to SIP/phone2.  I'm trying to get this to work using exten =>Z,1,dial(SIP/phone${EXTEN},30,t) , but it seems that the call still is made to SIP/2 instead of SIP/phone2
14:12.24MariusKarthausHow Can i do this?
14:13.38WIMPyDid you relod the config?
14:15.19MariusKarthausyes
14:16.13MariusKarthausI can see in the debug output that it is trying to call 2@MYIP, instead of phone2@MYIP
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14:16.45MariusKarthausso somewhow the concatination of the string to the $(EXTEN) variable is not working
14:17.12WIMPyWhat does your peer look like?
14:18.10MariusKarthausO never mind. I was missing the _
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14:18.44MariusKarthauscorrect: exten =>_Z,1,dial(SIP/phone${EXTEN},30,t), bad: exten =>Z,1,dial(SIP/phone${EXTEN},30,t)
14:18.46WIMPyOh, right, so how did you get a match then?
14:19.34MariusKarthausGetting the match turned out to be the problem. Stupid how sometimes you can stare at something like this without spotting the issue :P
14:20.41MariusKarthausWIMPy: anyway thank you for the time :P
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14:36.23jiremekhello all
14:36.35jiremekI have a problem with gotoIf, maybe du to mistake
14:37.06jiremekexten => 2,n,GotoIf(${comptemembre}==0?s,exeptclosed:)
14:37.32jiremekbut if ${[comptemembre}=1, gotoif exec s,exeptclosed ?
14:37.33jiremek!:!!
14:37.41jiremekIt's du to me ??
14:37.51jiremekanybody can help me ?
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14:39.43Guggejiremek: GotoIf($["${comptemembre}" = "0"]?label1:label2)
14:40.37jiremekgo, if I enter only GotoIf(comptemembre=0?label1:label2) works also ?
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14:41.37Guggejiremek: try it and see
14:41.59jiremekThe last question is 'cause this sentence worked fine, but not today...
14:42.21Guggei dont understand what you are asking
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14:43.30jiremekIn the past, the sentence exten => 1,n,GotoIf(comptemembre=0?s,exeptclosed:), worked fine.
14:43.45din3shis there a script to notify if E1 is on alarm?
14:44.43Guggejiremek: ive never seen that work, but what do i know.
14:45.04jiremekI try now your corrected sentence.
14:45.29jiremekyour sentence worked fine
14:45.34jiremekthanks for that
14:45.51jiremekIt's my falt so....
14:45.57jiremekThanks Gugge
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15:01.31p3nguinIt shouldn't work.  No expression has been formed in which to perform a test.
15:07.15schmidtshappy eastern folks!
15:11.31p3nguinNow why would eastern folks be so happy today?
15:11.40p3nguinponders
15:12.04WIMPywonders if happyness is allowed today in Austria.
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15:28.20din3shhappy western then
15:30.40p3nguinHappy western folks?
15:30.47p3nguinWhy are western folks so happy, too?
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15:34.50adnchow do I enable an event from the AMI and which event gives information on pickup and hangup?
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15:37.05gustoi have a problem again, when i call from my ata-adapter (analogue phone) myself or over other voip provider it throws uri 192.168.1.1 (local asterisk), and call goes on
15:37.29gustobut with one it throwse uri <myexternalip> and authentification does not happen
15:38.07fnwhatIs it possible to register asterisk as an MTP with CUCM?
15:38.31gustois there some parameter that i can use  to tell him to use always the local IP addresses for local phones?
15:43.46p3nguingusto: This isn't an asterisk problem.
15:44.12p3nguinThis is a networking problem.  You have misconfigured a network setting somewhere.
15:44.26p3nguinCould be an IP address, a host name, a DNS setting.
15:44.40p3nguinBut it is not an asterisk problem.
15:45.40p3nguinThe first thing I would check is the Proxy address in the ATA that is showing the issue.
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15:46.11p3nguinI would check for a host name which is resolvable to a public IP address instead of a local one.
15:47.42p3nguinIf that is the case, you have a few options:  change to a local IP address for asterisk; change the host name to a local host name which resolves to a local IP address; change the DNS for the host name to resolve to a local IP address when queried from a local IP address.
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15:54.35KavanSis there an FXO to ethernet type device I can use w/asterisk?
15:54.44KavanSi.e. connect a POTS line to ethernet, so I can virtualize asterisk
15:58.45gustop3nguin: yes, and i noticed that that may not even be problem on my side, because another voip provider works
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15:59.09p3nguinkavans: SPA-3102
15:59.10gustop3nguin: i get     -- SIP/personalvoip-00000033 is circuit-busy
15:59.11gusto<PROTECTED>
15:59.22KavanSp3nguin, reliable?
15:59.23p3nguingusto: Show me the peer and the dial plan.
15:59.28p3nguinkavans: fairly, yes
15:59.44gustop3nguin: i did not change it, and yesterday it worked
15:59.47p3nguinI've been using one for a while.
16:00.10gustop3nguin: '_49.' =>         1. Dial(SIP/personalvoip/${EXTEN})            [pbx_config]
16:00.49p3nguinAnd check the host= line for the peronalvoip peer entry in sip.conf.
16:01.06p3nguins/peronalvoip/personalvoip/
16:01.17gustop3nguin: http://pastebin.com/pRfQtsMz
16:02.00p3nguinAre you required to authenticate with a password?
16:02.34p3nguincore set verbose 3 ; sip set debug on
16:02.34gustop3nguin: ??? i have auth= set
16:02.40p3nguinMake a call.
16:02.45p3nguinPastebin the output.
16:04.14gustop3nguin: http://pastebin.com/WChAwkwk
16:04.26KavanSp3nguin, is there a more "enterprise" version of something like the SPA-3102?
16:04.41Qwellyou want it to run java or something?
16:04.46p3nguinkavans: Sure.  Look for SIP gateways.
16:05.24Qwellp3nguin: hey, I get to link this now.  http://www.digium.com/en/products/hardware/gateway
16:05.46gustop3nguin: http://pastebin.com/fM9LaPw1
16:05.49KavanShrm...
16:05.55KavanSI need analog sip gateways though
16:06.30p3nguingusto: You failed to authenticate on INVITE.  You need to put secret or remotesecret in your peer.
16:06.40gustop3nguin: really?
16:06.43p3nguinReally.
16:06.50gustop3nguin: but yesterday it worked without, why?
16:07.23gustoor maybe he did not match the auth= statements?
16:07.24p3nguinI'm not sure why it DID work, but I'm telling you now why it does NOT work.
16:07.35gustomaybe they are using other realm?
16:07.36gustook
16:07.40p3nguinMaybe you matched on realm once yesterday.
16:07.41gustoi ll set it and try again
16:07.50gustook
16:07.55p3nguinI would add remotesecret=whatever to the peer.
16:09.34gustop3nguin: well, that is weird
16:09.37gustop3nguin: now it works again
16:09.49p3nguinwith remotesecret?
16:10.14gustoyes
16:10.17p3nguinThat isn't weird, then.  That's how it is supposed to work.
16:10.22gustothat was the first time i used remotesecret
16:10.29gustothe most peers work without
16:10.31p3nguinNow you know to use it.
16:10.48p3nguinSometimes you have to authenticate to make calls.
16:11.00*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
16:11.03gustonot sometimes, everytime
16:11.18gustobut to me he takes auth= statements normally
16:11.20p3nguinSometimes you aren't matching realms like you think you are.  I, personally, never use realm authentication.  It has never been necessary for me.
16:11.34gustoreally?
16:11.36gustocool!
16:11.41p3nguinI use secret or remotesecret only.
16:11.45gustoah
16:12.03p3nguinin every peer.
16:12.15gustowell, maybe the voip providers are idiots
16:12.20p3nguinUsually.
16:12.24gustosome even have realm - asterisk set
16:12.41gustoi do not know how should i then distinct myself from them, how do i set my realm?
16:12.42p3nguinDon't rely on them to know how to configure things.
16:12.51gustowell
16:13.00gustoi know, thats the biggest problem with voip
16:13.13p3nguinThey usually build systems like you have built yours -- throw a bunch of settings together and see if it makes calls.
16:13.24gustono
16:13.28gustoi never did like that
16:13.40p3nguinIf it does not make calls, they change some things around and try again.
16:13.56p3nguinEventually, they have it so messed up that it's virtually impossible to configure anything else to work with it.
16:13.59gustowell, i did never do that
16:14.26p3nguinThen they switch everything to insecure settings so people can configure their devices to work with the service.
16:14.41gustoand that's even worse
16:14.50gustop3nguin: what insecure settings do you mean?
16:14.57p3nguinThen calls go through, so they feel like they did a great job and leave it as is for the rest of forever.
16:15.13gustop3nguin: i am not like that
16:15.56gustop3nguin: i never leave an unfinished job, i just put a telephone in the bedroom and one in my room, so that two people can make calls and so on
16:16.23*** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net)
16:16.57gustowell, i ll try endesha be-converged with that remotesecret, maybe it gets that one also to work because that one was saying the same today
16:17.39*** join/#asterisk rhg412 (b221a923@gateway/web/freenode/ip.178.33.169.35)
16:17.47*** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net)
16:18.12*** join/#asterisk oej (~olle@h87-96-134-129.dynamic.se.alltele.net)
16:18.22*** part/#asterisk the_weard (~arthur@41-132-18-108.dsl.mweb.co.za)
16:20.07gustowell, i get that voice error message there, as usual, so i am blocked to make calls, what is actually a fancy thing :-D
16:20.31gustobut maybe i only need to use another prefix
16:20.31p3nguinsip debug
16:20.40p3nguincore set verbose 3 ; sip set debug on
16:20.45p3nguinMake a call.  Pastebin the output.
16:20.56gustop3nguin: there will be nothing, because that woman's voice is on the provider's side
16:20.59p3nguinThis is the standard form for troubleshooting call failure.
16:21.05gustoyes yes
16:21.17gustoi have following idea
16:21.28gustolet's do exten => _49.,1,Dial(SIP/be-converged/${EXTEN})
16:21.33gustolet's do exten => _49.,1,Dial(SIP/be-converged/+${EXTEN})
16:21.39gustowould that work?
16:21.48p3nguinCan you ask the provider which format they require the number to be in?
16:21.55p3nguinTry the second format.
16:21.58p3nguinwith the +
16:22.16gustoyes
16:22.20gustothat was my idea
16:22.33p3nguinIf you dial 4912345, it would send +4912345 to the provider.
16:22.39gustow8
16:23.23gustowell, he took  it over
16:23.24gusto<PROTECTED>
16:23.35gustobut i get the same voice error message
16:23.52gustobut somewhere i read that that message comes only when people are blocked so maybe that's the case
16:24.49gustop3nguin: and how do you register to your peers when you do not use auth=-statements?
16:25.08p3nguinI use register statements, just like everyone else.
16:25.22gustop3nguin: well, and you put the password inside?
16:25.40p3nguinYes, because the password is part of a register statement.
16:25.47gustoyes
16:25.57gustobut to me some providers do not work that way
16:26.08gustothey need auth= and realm match set and then they work
16:26.49p3nguinYou can include the domain in the register statment.
16:26.58p3nguinregister => [peer?][transport://]user[@domain][:secret[:authuser]]@host[:port][/extension][~expiry]
16:27.43p3nguinregister => user@domain.com:mypassword@sip.provider.com
16:27.55p3nguinsimple
16:28.05gustoyes, but what i am telling you is that some providers do not authentificate that way
16:28.10p3nguinYes they do.
16:28.13gustobe-converged for example does not work that way
16:28.24p3nguinIt will if you configure it properly.
16:29.14gustowell, i think there is that problem that the user for authentification and the user for login are different
16:29.32gustoand the authuser has an \@ in the name
16:30.40p3nguinThat's fine.
16:30.42p3nguinSee the full syntax above.
16:31.12gustowell, that would not mean user@domain:password:username@host, how i expected it but
16:31.12p3nguinregister => user@domain.com:mypassword:differentauthuser@sip.provider.com
16:31.15gustono
16:31.47gustothat would be telnr:password:user@domain@host
16:32.31p3nguinI think you have that wrong.
16:32.40p3nguinIs your username telnr?
16:32.44gustoyes
16:32.56gustoit authorizes over telephone numberg
16:32.57p3nguinWhat is the authuser?
16:33.14gustoauthuser is that user with \@ in the name
16:33.58gustoand that is why it works only with realm matching because there i can put the username@domainusername in w/o problems
16:34.05WIMPyAre you sure that's the username and noct a username and a proxy?
16:34.10p3nguinInstead of giving me little pieces of information, give me all of the information at one time.
16:34.26p3nguinIt's a user name and a proxy, I'm sure.
16:34.42gustono
16:34.51gustousername = username@domain
16:34.52p3nguinLet me give you an example of how I register like that.
16:35.48gustoAuthorization: Digest username="username@domain",realm="ims.telefonica.de"
16:36.02gustoAuthorization: Digest username="username",realm="ims.telefonica.de" <- does not work
16:36.57*** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf)
16:37.50p3nguinregister => rob@proxy.provider.com:s3cr3tp4ssw0rd:rob@proxy1.provider.com/3149691077
16:38.13gustothat does not work
16:38.22p3nguinYou're doing it wrong.
16:38.25gustoi tried that i am not silly
16:38.40p3nguinThat is a matter of opinion.
16:38.42gustoit gets forbidden, because he needs TELNR for login
16:38.55p3nguinThen send the correct login name.
16:39.03gusto:-D
16:39.36gustohowever, to most providers that register statements work, you are right
16:39.45p3nguinWhat is the domain they want you to use?
16:39.54gustobut to some your only option is realm matching for registry
16:40.08gustothe domain is the same as the domainname inside the username
16:40.38gustoand to me it works over auth= so there is no problem to be solved here
16:40.40gusto:-)
16:40.43p3nguinWHAT IS THE DOMAIN?
16:40.52gustoendesha.be-converged.com
16:41.31p3nguinThey want you to use endesha.be-converged.com as the proxy, also?
16:41.37gustono
16:41.45p3nguinWhat is the proxy host?
16:41.48gustoproxy is sipproxy.endesha.be-converged.com
16:41.55gusto^^host
16:41.56p3nguinregister => telnr@endesha.be-converged.com:password:telnr@sipproxy.endesha.be-converged.com
16:42.15gustoi tired that too, doesnt work
16:42.20p3nguinTry again.
16:42.47gustono
16:42.53p3nguinOkay, then I'm off.
16:42.57gustoyou did not put the username in
16:43.08p3nguinYes I did.
16:43.10gustono
16:43.17p3nguinI put everything in except YOUR REAL PASSWORD.
16:43.31gustoyes yes, and my real telephone number
16:43.31p3nguinCopy and paste it.  Change "password" to your actual password.
16:43.41p3nguinDon't change anything else.
16:43.44p3nguinCopy.  Paste.  Change password.
16:43.52p3nguinSave.  sip reload
16:43.55gustorelax, there is nothing going on around that, the server is not configured properly, we know already
16:44.19p3nguinOkay, good luck then.  I have better things to do than fuck around with someone not willing to try.
16:44.29*** join/#asterisk AlfE_ (~quassel@83-215-36-12.bruck.dyn.salzburg-online.at)
16:44.37gustop3nguin: i said to you, i ve tried that versions already a lot of times
16:44.51p3nguinAnd I said try it again copy and paste.
16:44.53gustop3nguin: i tried every combination you can think of
16:45.00chuckfwonders if someone will help but I'm not willing to do as you suggest because you're doing it wrong
16:45.29p3nguinchuckf: What?
16:45.39chuckfp3nguin: just messing wit hyou
16:45.46chuckf:)
16:46.26gustop3nguin: the bottom line is that it works and incoming calls work, so there is nothing to be fixed here, rela
16:46.29gustorelax
16:46.33gustoand calm down
16:46.41p3nguinLet me put it this way:  one of us managed a LOT of asterisk boxes which are all online and making phone calls.
16:46.56gustoall other voip providers work with register => statements with passwords in, i tried that already
16:47.26gustop3nguin: so then let us talk about that
16:47.54*** join/#asterisk din3sh (din3sh@41.136.241.243)
16:47.55gustop3nguin: how do you make phone calls from one asterisk box to another? you go like [peer] secret / remotesecret and so on?
16:48.38gustobecause that would be cool to set up asterisk to everyone and bypass the PSTN completely
16:48.56p3nguinI got some to work by trial and error with configuration problems of that same kind.  Something normal did not work, so I ask questions.  Someone suggests something to me, I try it for them even if I think I already tried it.  What do I have to lose?  It already doesn't work, right?  I may as well try to satisfy their suggestion so we can move to another suggestion.
16:49.11gustothere are not many people i make calls with, so for me it would be sufficient
16:49.17p3nguinAsterisk to asterisk:  http://pastebin.com/Ag7tknm2
16:49.40[TK]D-FenderAsterisk to asterisk is virtually no different that to an ITSP.
16:49.41[TK]D-FenderSIP is SIP
16:49.47gustop3nguin: it works, calm down
16:49.51[TK]D-FenderDon't know why you think it'd be some sort of "magic"
16:50.14gusto[TK]D-Fender: i did think the same way
16:50.23[TK]D-FenderYou just might aviod a couple of stupid settings that aren't essential to judge on.
16:52.41p3nguin<gusto> p3nguin: it works, calm down    <--what works?
16:52.59gustop3nguin: why is that one allowing only ulaw codec in that settings? when i do asterisk to asterisk i can use the best codec around, dont i? btw. what do you see as the best HQ codec? speex?
16:53.13gustop3nguin: registry to endesha
16:53.26p3nguin*sigh*
16:53.28gustop3nguin: sipproxy.endesha.be-converged.com:5060  N      499113104246      3465 Registered           Fri, 06 Apr 2012 16:18:29
16:53.41p3nguinAll that arguing and you were wrong the whole time?
16:53.51gustop3nguin: it always worked, but only with auth=-statements
16:54.02p3nguinSo you tried my register statement?
16:54.08gustop3nguin: and not with register => w/ password included
16:54.41p3nguinIsn't it neat how things work when you do it right?
16:54.59p3nguinAs for the codec...
16:55.15p3nguinYou don't necessarily need to use the best quality of codec all the time.
16:55.22gustop3nguin: in this case the right way was "auth=username@domain:password@ims.telefonica.de-relam
16:55.26gustorealm
16:55.50gustop3nguin: well, do you see ulaw as a codec that is OK?
16:55.51p3nguinI don't use realm auth.  It isn't necessary for me in any case.
16:56.01p3nguinDo you have any phones that use a codec better than ulaw?
16:56.03gustoyou said that already
16:56.36gustoi am new to codecs like this, but i ve read that ulaw is g711 and my linksys pap2t can do some more
16:56.52p3nguinThe quality of the call will always be only as good as the worst codec being used on any call leg.
16:57.27p3nguinIf you have one phone using only gsm to asterisk; asterisk to asterisk using g722; and then a ulaw between that asterisk and the other phone...
16:57.39gustoof course
16:57.42p3nguinThe quality of the whole call will be equal to or less than gsm.
16:58.06p3nguinSo using g722 would not be necessary.
16:58.22gustobut i mean, when i call another asterisk that was set up by me, i can set it up to allow only one codec, for examle g722 and that for the whole line end-to-end
16:58.49p3nguinIf you have a bunch of g722 phones on two asterisks, use g722 between the two asterisks also.
16:58.56gustoyes
16:59.07p3nguinThen you can have g722 end to end.
16:59.10gustoyes
16:59.21gustoand that is better than g711, ritght?
16:59.23gustoright?
16:59.32p3nguinYes.  G.722 is HD audio.
16:59.35*** join/#asterisk eicto (~morgoth@eicto.broker.freenet6.net)
16:59.47p3nguinwideband
17:01.38*** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee)
17:03.07gustop3nguin: my linksys pap2t does not support g722, that was only an example, but it does g723 and g726
17:03.22gustoand g729
17:04.41WIMPyI'd be pretty surprised if any ATA offered G.722.
17:04.52gustoyes
17:04.54gustomine does not
17:05.04Qwellthat seems dumb
17:05.04p3nguinIt would be silly to have it.
17:05.18gustoand that 729 also only 729a which is reduced quality but less complexity
17:05.25p3nguinUnless someone made an HD analog phone recently.
17:05.36gustoright
17:05.41*** join/#asterisk thecardsmith (~doug@pdpc/supporter/student/thecardsmith)
17:05.45p3nguinJust use ulaw or alaw on the ATA.
17:05.47gustoi use siemens gigasets
17:06.00gustothe recent one is siemens gigaset a400, has quite good sound
17:06.03Qwellit doesn't matter what phone you use
17:06.08Qwellit's always going to be ulaw/alaw
17:06.20QwellAlways.  100% of the time.  Never anything else, ever.
17:06.23QwellNot. Even. Once.
17:07.07gusto???
17:07.24WIMPyActually DECT only uses G.726.
17:07.50QwellWIMPy: The cable from the wall to the basestation is where the issue is.
17:08.20QwellThose phones may use higher quality between themselves, if, and only if, the call never touches analog.
17:08.34WIMPyThat should be an ethernet cable anyway.
17:08.49QwellThen it's not an analog set, so how is that relevant? O.o
17:09.05p3nguin*shrug*
17:09.14p3nguinI got lost early on.
17:09.16gustowell, mine has also g726 support
17:09.17WIMPyJust saying that it's not even G.711 quality.
17:09.18*** join/#asterisk dijib (~root@bas10-kitchener06-1176138838.dsl.bell.ca)
17:09.21dijibhey all
17:09.42QwellWIMPy: I was under the assumption that G.726 was > G.711.  You're right then, that it could be that quality the whole way.
17:09.55p3nguinThe moral of the story is to use G.711 for that type of phone.
17:11.07WIMPyIt's 32kbps, but it shouldn't be much lower than G.711.
17:13.12gustoso from the available ones g711 is the best one
17:13.29WIMPyyes
17:13.59*** join/#asterisk bluregard (~matt@c-71-201-99-216.hsd1.il.comcast.net)
17:14.12bluregardhello all
17:15.14dijibcan someone show me the two rules needed in iptables to accept & forward 5060 to a machine behind my nat.
17:15.22gustoand the only one with at least 64kbps
17:15.26dijibive been trying to figure this out for a week and i still have nothing
17:16.15WIMPyThat's only one rule.
17:16.35dijibdont you need two, one for the port to be opened and one for the forward?
17:16.36bluregardis it possible for Asterisk to detect a transfer or "3-way" initiated by the called party?
17:16.46gustodijib: i did not have to do anything on my firewall, to me it just works
17:16.57[TK]D-Fenderdijib: open is when you accept it yourself.  Forward is forwad. they are separate
17:17.04gustodijib: ah, you are making it forward ... ah sorry
17:17.11[TK]D-Fenderbluregard: no
17:17.14WIMPydijib: No. You forward in nat/PREROUTING. It will never hit the filter table that way.
17:17.17gustodijib: i have asterisk on my router .. so that would ... of course
17:17.19[TK]D-Fenderbluregard: * has no idea it's a 3-way call
17:17.21dijibyour asterisk box sounds like its running on the firewall gusto
17:17.34[TK]D-Fenderbluregard: it's just 2 calls to the same peer... wht it's doing with them is besides the point
17:17.47gustodijib: yes it is
17:17.56bluregard[TK]D-Fender, I didn't think so.  Is that even possible with any phone system?
17:17.59dijibive got prerouting going to MYSHAPER-IN
17:18.07[TK]D-Fenderdijib: "iptable port forwarding how-to.
17:18.13[TK]D-Fenderdijib: it is ONE line of config
17:18.28dijibwill it mess it up if i have the open rule and the forward?
17:18.31WIMPybluregard: Sure
17:18.42[TK]D-Fenderdijib: There is something seriously wrong if it takes more than 5 minutes of googling.
17:19.09[TK]D-Fenderdijib: It won't be open AND forward.
17:19.22[TK]D-Fenderdijib: Forward takes precedence
17:19.26WIMPyInbound traffic shaping doesn;t really work anyway.
17:19.40bluregardWIMPy, how so?  Once the call is bridged what the called party does with the call is independent of the system used to make the call, isn't it?
17:20.12WIMPybluregard: In the PSTN you get notifications about such things.
17:20.45*** join/#asterisk adeel (~adeel@72.53.78.136)
17:20.54bluregardWIMPy, but said notification can't be passed to Asterisk?
17:21.26p3nguinYou have to preroute in the PREROUTING chain of the nat table, and you have to forward in the FORWARD chain of the filter table.
17:21.38WIMPyI don;t think it reads these messages. I wouldn't even know what it could do with that information.
17:22.23bluregardWIMPy, the goal would be to disconnect the call upon the called party initiating a 3-way and play a message as to why the call was disco'd.
17:22.46p3nguinIf you have a policy of ACCEPT in the FORWARD chain, you shouldn't need to add rules there... unless you already have rules their to counteract the default policy.
17:23.46WIMPybluregard: I don't see an easy way to do that with Asterisk.
17:24.11dijib-A FORWARD -d 10.0.0.15/32 -p tcp -m tcp --dport 81 -j ACCEPT how do i redirect this from 81 to 80?
17:25.22p3nguinHowever, if you are running the services on the same system as you are executing the iptables commands, you don't forward anything.  Forwarding is for services running on a different system than the system where you are executing the iptables commands.
17:25.42dijibive got multiple IP's on this box
17:25.46p3nguinSo?
17:25.50p3nguinHow is that relevant?
17:25.50dijibthats the issue... the thing you told me not to do p3nguin
17:25.55dijibi dnt know
17:26.09p3nguinYou need to learn how to do it correctly.
17:26.17dijibno shit
17:26.29dijiband ive read some of the man page
17:26.33dijibbut im still clueless
17:27.25p3nguinHaving a bunch of different private IP addresses on the single system for this purpose is silly and unnecessary.
17:27.44bluregardWIMPy, But the PSTN does provide such notifications...
17:27.52p3nguinFor your purposes, you should be using name-based virtual hosts.
17:27.58WIMPybluregard: Yes
17:28.14WIMPybluregard: But you obviousely need a digital connection to receive them.
17:28.39WIMPyi.e ISDN or SS7
17:28.42bluregardWIMPy, PRI
17:28.45bluregardok
17:31.40dijibhttp://pastebin.com/ygZKsLcS
17:33.29p3nguinLine 67 looks wrong to me.
17:33.33WIMPyDoes iptables-save actually work by now?
17:33.35p3nguinAre you running services ON THIS BOX?
17:33.42p3nguinIt has always worked.
17:34.02WIMPyNo it hasn't.
17:34.07p3nguinI started using iptables back in like 2001, and iptables-save worked back then.
17:34.29dijibok its gone
17:34.30p3nguiniptables-save > iptables.rules
17:34.33WIMPyThen you never tried the right (wrong) options.
17:34.45p3nguinsaves to iptables.rules file.
17:34.49dijibservice iptables save works
17:34.50p3nguinworks correctly.
17:35.24WIMPyIt has been know for long times to fail on certain options.
17:35.47p3nguinThere aren't that many options.
17:35.50p3nguinonly three
17:36.43WIMPyI see tons of options.
17:37.41p3nguinhttp://pastebin.com/0v64TVxw
17:38.21WIMPyTo iptable, not to iptables-save.
17:38.32p3nguinhuh?
17:38.53p3nguinYou're saying that iptables-save won't dump the rules if your rules are goofy?
17:38.55WIMPyIt has been know to not correctly save all rules.
17:39.00WIMPyknown
17:39.14WIMPyyes
17:39.33p3nguinI've never encountered that.  Maybe I use normal and conventional rules.
17:39.43WIMPyWell, maybe it has been fixed. i don;t know.
17:40.22p3nguinI would be interested to know what kind of goofy rules would fail it.
17:40.41p3nguindijib: <p3nguin> Are you running services ON THIS BOX?
17:41.19WIMPyI don't know. It's been some time since I read about that.
17:42.29gustowhat do you ppl think about that MD5 authentification when he authenticates to a VoIP provider through UDP using that MD5 hashes, ha?
17:44.04dijibive got apache running on 10.0.0.1, 10.0.0.10-10.0.0.15
17:44.26dijiball on port 80, but i would like dyndns:80-85 to redirect to those addresses
17:44.40dijib.1 & .10 being the same apache instance
17:44.46p3nguinIt's wrong.
17:44.55p3nguinIf you have services ON THIS BOX, you do not use FORWARD.
17:44.55dijibwhat wrong?
17:44.59p3nguinDo you understand?
17:45.33dijibok then i just need to redirect :85 -> 10.0.0.15:80?
17:45.34p3nguinYou don't forward things that are TO THE BOX you are running the iptables commands on.
17:45.46p3nguinNo, you need to stop doing it wrong and use name-based virtual hosts.
17:45.54p3nguinThen you run all the sites on the same port.
17:46.01gusto:-D
17:46.02dijibok well i really would rather not as i like IP's
17:46.02gustohehe
17:46.18p3nguinI refuse to help you do something stupid and wrong.
17:46.26dijibyes all are running on :80 just different IP's
17:46.27p3nguinIf you can talk someone else into it, good for you.
17:46.37dijiblol
17:47.10p3nguinWhen you're ready to do it right, I'll direct you.
17:47.17dijibso i should do name based? like localhost:80-85?
17:47.26p3nguinno
17:47.37dijibok/?
17:47.53p3nguinName-based virtual hosts display the web site based on the name used in the URL.
17:48.16dijibthen do i need to put those names in HOSTS?
17:48.20p3nguinSo both site1.com and site2.com point to the same IP address on the server, but they display different sites.
17:48.28WIMPydijib: Look for NameVirtualHost in Apache.
17:48.30p3nguinIt has nothing to do with "HOSTS."
17:48.41p3nguinI don't even know what "HOSTS" is.
17:49.13dijibhttp://pastebin.com/DwaiG98V
17:49.13p3nguinThe google term that I have been throwing out there all this time is "name-based virtual host".
17:49.36p3nguinThat's a virtual host, but it isn't name based.
17:49.38dijibhosts is like the hostfile in windows where you can say acmecorp.com 10.0.0.50
17:50.04p3nguinYour windows hosts file isn't going to control the web server on Linux.
17:50.08dijibshouldnt i just do that with all apache to the same IP?
17:50.10p3nguinDon't be ridiculous.
17:50.24dijibi know its not p3nguin but linux has the same kindof hostfile
17:50.35p3nguinCorrectly configure name-based virtual hosts.  Stop guessing and read the manual.
17:51.33p3nguinStart by removing the extraneous IP addresses from the system.
17:51.41p3nguinYou need one.
17:51.57p3nguinForward port 80 through your gateway to that IP address on your server.
17:52.12p3nguinNow that networking part is complete.  Stop fucking with it.
17:52.43p3nguinThen configure name-based virtual hosts for all your different web sites.
17:53.56gustothis is not a supprt channel for apache/networking/linux and other stuff, this should be only about asterisk here
17:54.14p3nguinSays you.
17:54.18gustoyes
17:54.30dijibok multiple IPs are dead
17:54.32Qwellsays me, too
17:55.33*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
17:57.57dijibthis:? -A FORWARD -d 10.0.0.10/32 -p tcp -m tcp --dport 80:85 -j ACCEPT
17:58.11dijibonly rule i have regarding 80-85
17:58.25p3nguinWhy would you need 80-85?  http = 80
17:59.11dijibi want to run multiple httpd servers
17:59.17p3nguinWhy would you want to do that?
17:59.32p3nguinIt's silly and unnecessary.
17:59.44dijibso i can make different sites....
18:00.03p3nguinname-based virtual hosts   <-----------
18:00.10p3nguinWhat have I been saying all along?
18:00.27*** join/#asterisk bent_screwdriver (~bent_scre@74.255.249.66)
18:00.29dijibthey will be comming from my dyndns.... how would that work?
18:01.00dijibi need to seperate them by port as i see it
18:01.05p3nguinYou told me that you would have multiple names for the multiple sites.  Was that not accurate?
18:01.14dijibno.
18:01.18dijibonly one domain name
18:01.52WIMPyYou can still have cnames to a dyndns domain if your provider doesn't support subdomains.
18:02.04WIMPyBut that's really off topic here.
18:02.34p3nguinI would either use subdomains or directory names.  I would not run multiple httpds on multiple ports to support such a thing.
18:02.48[TK]D-FenderNo the topic is "the multitude of things dijib doesn't get about networking and the internet" ;)
18:03.03p3nguinsite1.dyn.org or www.dyn.org/site1
18:03.13p3nguinsite2.dyn.org or www.dyn.org/site2
18:03.33[TK]D-Fenderdijib: the REQUESTED URL is part of the HTTP protocol... the face of what IP & port it lands on is IRRELEVANT
18:03.39[TK]D-Fenderfact*
18:04.04dijibdyn.org:81
18:04.08p3nguinsilly
18:04.12[TK]D-Fenderdijib: Go learn HTTP and whatever daemon you're going to use
18:04.12dijibwhy?
18:04.17p3nguin(1303.02) <p3nguin> site1.dyn.org or www.dyn.org/site1
18:04.32dijibwhy not ports?
18:04.39dijibthats the way ive always done it
18:04.41[TK]D-Fenderp3nguin: Betting his provider blocks inbound web too....
18:04.48Qwelldijib: Because it's wrong.
18:04.49dijibno.jpg
18:04.51[TK]D-Fenderdijib: Is that the case?
18:05.00*** join/#asterisk d00gster (~dt@173.195.1.95)
18:05.30p3nguinIt's kind of like trying to run asterisk chan_sip on a bunch of different ports.  It's goofy and wrong.
18:05.51dijibyeah but its the way it has to be
18:05.57p3nguinNo it isn't.
18:05.58QwellNo, it doesn't.
18:06.14p3nguinI've given you two perfectly acceptable methods.
18:06.24p3nguinYou don't want subdomains, fine, use directories.
18:06.47p3nguinIf you're going to use directories, you don't even need to learn name-based virtual hosting.  You just saved yourself 6 months.
18:06.48dijibwhy is :81 not acceptable.
18:06.49dijibeff
18:07.02dijibits like :8080 used everywhere
18:07.08Qwell8080 is also wrong.
18:07.16dijibyeh and its used all over the place
18:07.21QwellSo what?
18:07.37[TK]D-FenderYou run Web on 80 like the rest of the universe that is what is "right".  Everything else is a hack
18:07.55p3nguinYou think if people everywhere do soemthing wrong, that makes it right?
18:08.00dijibso are you going to go and make sure every service running on an alternate port is fixed with either subdomain or directory... i dont think so tim
18:08.07dijibyes it does
18:08.20p3nguinI don't run services on non-standard ports.
18:08.43p3nguinAnd I don't run multiple instances of the same service to achieve listening on multiple ports.
18:09.12dijibim not really running instances im more running virtualhosts on the same instance using multiple ports
18:09.14p3nguinPeople everywhere steal and kill.  Doesn't make it anymore right than before they did it.
18:09.37dijibso who has a stick of ddr2 for a sangoma-a200
18:09.37dijib?
18:09.42dijibor an ATA
18:09.56p3nguinI'm sure there are people somewhere willing to help you configure things in a goofy manner.  I'm not one of those people.
18:10.35dijibok then i guess ive got to do subdomains
18:10.43dijibwhich is gay but its the proper way
18:10.52WIMPyGood choice
18:10.55p3nguinThen you're back to learning name-based virtual hosts again.
18:10.59p3nguinBut it's the right way.
18:11.07p3nguinAnd it's easy!
18:12.37[TK]D-FenderNo, what is "gay" is some chump hosting we services off non-standard ports and running DynDN instead of proper domains as a hosting service like some cheap-ass kiddie in their parent's basement with delusions of hitting the big-time.  Also from the same person who can't fingure his way through 2 (tops) lines of IPTABLES
18:12.43[TK]D-Fenderweb*
18:13.23p3nguinIf you have a couple IP-based virtual hosts already configured, it will be very easy to convert that to name-based.
18:13.44p3nguinPastebin a couple virtual hosts and I will show you how to turn it into name-based.
18:14.10p3nguinAnd do it before someone has an asterisk question and we get kicked out.
18:14.27bent_screwdriverwill a macro jump to exention "h" on hangup? I can't seem to make it work while using a macro.....
18:14.42p3nguinh should never run inside the macro.
18:14.42[TK]D-Fenderbent_screwdriver: yes
18:15.11[TK]D-Fenderp3nguin: Everything depends on what is coded where in the dialplan....
18:15.18p3nguinprobably
18:15.42p3nguinI guess we need to see the dial plan to know for sure.
18:17.00dijibhttp://pastebin.com/Y0xMhBug
18:17.06[TK]D-FenderAnd it depends on the definition of "hangup"
18:17.10dijibwhats my dialplan got to do with anything?
18:17.55[TK]D-Fenderfacepalms
18:19.34dijibnvmd
18:21.39bent_screwdriver[TK]D-Fender: I had h in macro, and not original context......doh! works now. thx
18:21.44*** join/#asterisk reber (~reber@tsm83-4-78-232-65-13.fbx.proxad.net)
18:25.59p3nguindijib: http://pastebin.com/index/Y0xMhBug
18:27.32p3nguinThis is for name-based virtual hosts with a single IP address on a single standard port.
18:28.17p3nguinin the most bacic form, of course.
18:29.19p3nguinI didn't define individual log files, or include configuration of directory containers.
18:33.56dijibp3nguin: that didnt open... it was blank
18:34.00dijibthis is what i have http://pastebin.com/1MNimrG5
18:34.37dijibyo back in TEN, laundry folding
18:34.50p3nguinhttp://pastebin.com/VcUEbAgm
18:34.59p3nguinI guess it didn't submit when I hit the button.
18:36.28p3nguinYou've repeated the servername in the serveralias, which is unnecessary and redundant, and you only need a serveralias if you intend to have things like someothersite.dijib.sytes.net direct to dijib.sytes.net.
18:41.59*** join/#asterisk SaRSAeOL (~sarsaeol@66-113-78-49.rev.ibsinc.com)
18:42.09dijibba ck
18:43.50SaRSAeOLanyone here been able to test the new digium phones yet? i have some and have observed some odd issues
18:44.18SaRSAeOLlike asterisk 1.4 not setting up an ftp stream to them
18:44.21dijibok those are in p3nguin now im back to the firewall
18:44.23SaRSAeOLrtp*
18:44.28QwellSaRSAeOL: We encourage issues to be reported to support, so that they can be fixed.
18:45.01SaRSAeOLyeah they just said 1.4 is end of life and they won't support it
18:45.15p3nguinTime to upgrade.
18:45.21Qwellpretty much
18:45.46SaRSAeOLyeah my thoughts too… it just a fairly complicated installation so i was combing for hacks
18:45.47p3nguindijib: What is the question?
18:45.48SaRSAeOLoh well
18:46.10dijibno clue
18:46.14QwellSaRSAeOL: well, you could attach your phones to a 1.8 box (and you really aught to use DPMA), and have that connect to the 1.4 box that already works
18:47.29p3nguin(1251.56) <p3nguin> Forward port 80 through your gateway to that IP address on your server.
18:47.42p3nguindijib: ^
18:48.02WIMPyWhat's DPMA?
18:48.10SaRSAeOLQwell: yup i plan on doing the DPMA for the presence stuff, that might be a viable solution in the interim… thanks
18:48.24SaRSAeOLWIMPy: digium phone module for asterisk
18:48.24QwellSaRSAeOL: it's for much much more than just presence
18:48.37Qwellyou get all the fancy app magic with DPMA
18:48.48Qwellwell, most
18:49.16SaRSAeOLyeah its nice, I'm not 100% sure the client will care about all that… but the call queues are nice too
18:49.40SaRSAeOLbut yeah it made these quite attractive
18:51.39*** join/#asterisk paolosupino (~yaaic@82.84.86.249)
18:56.25dijibok p3nguin the httpd server is on the gateway, shouldnt this just be an accept rule?
18:58.00p3nguinYes.  You do not FORWARD for services on the same host that you execute the iptables rules.
18:58.28p3nguinYou need a basic ACCEPT in the INPUT chain for port 80 if you have a policy or other rule that blocks the port.
19:00.37dijiband if that doesnt work?
19:00.46p3nguinYou've done something incorrectly.
19:01.12p3nguinYou have your server directly on the internet?
19:01.56dijib-A INPUT -p tcp -m tcp --dport 80 -m state --state NEW,ESTABLISHED -j ACCEPT
19:01.59dijibyes
19:02.03dijibthis is the rule i have
19:02.24dijibthats not working with any of the dyndns's also where in cent do i configure ddns?
19:03.16WIMPyWhy do you restrict that to NEW,ESTABLISHED? That doesn't make sense.
19:03.21p3nguinThe correct rule would have been iptables -A INPUT -p tcp -m tcp --dport 80 -j ACCEPT
19:03.23dijibno clue.
19:03.29p3nguinHe's just doin that because he read it somewhere.
19:04.21p3nguinwimpy: He doesn't read man pages or any type of technical docs.  He only looks at examples where other people do things and post what they did, even if they did it wrong.
19:05.14dijibstill not working with that rule
19:05.27p3nguinwimpy: He doesn't ever bother to take the time to understand what anything means, just copies/pastes things arbitrarily.
19:06.05p3nguindijib: You've probably got it so messed up that the correct rules don't do anything.  Pastebin the entire output of "iptables -L INPUT -nv".
19:06.13dijibno. i understand its the -A INPUT chain, -p protocal tcp not sure what -m is and --dport is destination port -j accept is the accept rule
19:06.51*** join/#asterisk pouyou (5243dc7b@gateway/web/freenode/ip.82.67.220.123)
19:07.00pouyouhi
19:07.06WIMPyThere could also still be some forwarding rule.
19:07.16p3nguinOne step at a time.
19:07.23dijibhttp://pastebin.com/E87ZG1pY
19:07.39dijibthere are no forwarding rules
19:08.15pouyoui have been trying to achieve a few things recently but cannot get things to work so far. I compiled app_rtsp and app_transcoder / installed them on asterisk 1.8.11 freshly updated/installed also
19:08.45pouyouevery time i try to get something from an rtsp server, i get this  kind of messages
19:08.47p3nguinLine 18 isn't a good rule.  I'm not sure about 17.
19:09.14p3nguindijib: iptables -L FORWARD -nv shows no rules?
19:09.38dijibnone
19:09.53pouyouchannel.c:1094 ast_best_codec: Don't know any of 0x0 (nothing) formats channel.c:4913 ast_write: Codec mismatch on channel Local/start@RTSP-e8ee;2 setting write format to unknown from ulaw native formats 0x4 (ulaw)
19:09.53p3nguinLet's look at the nat table next.
19:10.04p3nguiniptables -t nat -L PREROUTING -nv
19:10.16pouyouanyone got any idea as to how i could fix it ?
19:10.57p3nguinpouyou: It seems like the peers don't have codecs allowed.
19:11.00p3nguinat least one peer
19:11.08p3nguin0x0 = no codecs allowed
19:11.11pouyouusing linphone i only get a white background
19:11.24pouyouwell they are supposed to have everything allowed *
19:11.29pouyoulet me check
19:11.55p3nguindisallow=all  \  allow=ulaw
19:12.02p3nguinin that order
19:12.28dijibnothing showing in *nat
19:12.39dijibbut the masq rule should be in there no?
19:12.44dijibthats the only 1 i should have
19:13.01p3nguinIs that box doing NAT for other computers in the LAN?  It acts as a gateway?
19:14.44p3nguinIs it supposed to do NAT for other computers in the LAN?  Do you want it to?
19:15.00p3nguinsupposed to do it vs. doing it
19:15.35*** join/#asterisk evil_gordita (~evilgordi@ip70-188-50-186.rn.hr.cox.net)
19:15.38dijibit does do nat
19:15.44dijibim using it right now
19:16.04dijibPackets after routing (POSTROUTING)
19:16.04dijibSelect all. | Invert selection.
19:16.04dijib<PROTECTED>
19:16.06dijibMasquerade If output interface is ppp0
19:16.06p3nguinThen you already have the masquerade rule.
19:16.21dijibyeh but it wasnt listed in that last command
19:16.50p3nguinThen that's part of your problem.  You're using some type of GUI for it plus the command line.  Don't mix interfaces.
19:17.09dijiboi.
19:17.20p3nguinI can't help you use a GUI.
19:17.26dijibwebmin is the gui, and im just reading whats set on it
19:17.47dijibim setting them in /etc/sysconfig/iptables
19:17.58p3nguinIf you want to flush all the chains and configure it via CLI, I can do it.  Otherwise, I can't.
19:19.24pouyoui had this and it did not  work : disallow = all  \ allow = ulaw,alaw,gsm,ilbc,speex,speex16,g726,adpcm,lpc10,g729,g723,h261,h263,h263p,h264,t140,red,mpeg4
19:19.31p3nguinright
19:19.35p3nguin(1411.55) <p3nguin> disallow=all  \  allow=ulaw
19:20.06p3nguinYou don't need to list EVERY codec available.  List the one you need to use.
19:20.57p3nguinIf you aren't sure what codec the other side has available, the sip debug will reveal that.
19:21.01p3nguinsip set debug on
19:21.02*** join/#asterisk orioni (~chatzilla@46.183.121.133)
19:21.14p3nguinMake the call, check the codec negotiation.
19:21.37orioniwhat is the equivalent of 'g' but so the execution contiunues in both cases
19:21.38orioni?
19:22.07orionii mean how to continue with dialplan execution when the caller or called party hangs up  ?
19:25.39pouyouhmm ok, nothing better with only ulaw allowed, i ll check the codec. As for the codecs availables, we are still doing some tests as to which ones we will stick with  that's why I added some of them one after the other
19:26.27p3nguinIf you're just going to add all of them to an allow list, don't disallow all first.  Don't disallow or allow.
19:26.56pouyouas i said, we are supposed to strip the list down in a bit of time
19:26.59p3nguinThe sip debug will show you what the othe side supports.
19:31.11pouyoui seem to be able to see what my client supports but not what the rtsp is feeding me. Do you know what I should be looking for ?
19:32.10*** join/#asterisk TheCompWiz (~TheCompWi@wsip-68-109-200-102.mc.at.cox.net)
19:32.29TheCompWizanyone know if it's possible to do video prompts in files?  similar to the voice-prompts?
19:34.10Qwellin files, as opposed to what?
19:34.28TheCompWizstreaming between two video-capable endpoints?
19:34.49Qwellsure, it works fine if you have them in the right format
19:35.00TheCompWizand what format would that be?
19:35.05Qwellsomething Asterisk supports
19:35.15Qwelland your phone
19:35.56TheCompWiz... that's for being so descript.    I mean... typical "video" data is encoded in some sort of wrapper format (like an "avi" or "mov" ...) what does asterisk expect?
19:36.14Qwellmodule show like format_*
19:36.47Qwellerr, just format_
19:37.17TheCompWizyeah... it shows the formats asterisk supports.... but I don't know what sort of data should be written to the files.
19:37.31QwellDid you read their descriptions?
19:38.05TheCompWiz"Raw H.264 data"... but other sources say that asterisk also embeds some timing data in there... etc...
19:38.37TheCompWiz"Currently, Asterisk dumps the content of RTP packets including some timing information in .h263, ph263p, .h264 files."   (from voip-info)
19:41.19Qwellwhen you figure it out, fix the documentation
19:54.53*** join/#asterisk gusto (~gusto@nrbg-4dbe3949.pool.mediaWays.net)
19:57.15*** join/#asterisk timahvo1 (~rogue@197.178.221.179)
20:16.27*** join/#asterisk exothermc (~exothermc@m6.office2-ww.wideideas.net)
20:21.24*** join/#asterisk ideaman (~ihaveapla@173-10-29-218-BusName-utah.ut.hfc.comcastbusiness.net)
20:22.33*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
20:22.33exothermc[TK]D-Fender: You around?  I'm working on an active hack of my box, and I'm not quite understanding how they are registering.  I keep on changing my password to a 15 char random password, but they still get in.
20:24.25gustoexothermc: who?
20:25.03exothermcgusto:   You know they forgot to leave a greeting card.  I was really disappointed then didn't provide full contact information myself.
20:25.26[TK]D-Fenderexothermc: Show us
20:25.52exothermcwhere is a good way to share a file I wonder.  small little pcap I have.
20:27.35gustoexothermc: run at least a tcpdump so that you get at least the IP addresses
20:27.50exothermcgusto: Ya just a host out in Palistine.
20:28.08exothermcgusto: Not like chasing that will produce more than some box that was compromised.
20:28.18gustoexothermc: are they hacking your asterisk box or your hotel room?
20:28.42exothermcgusto: Just the asterisk box, running some international traffic.
20:29.43exothermcIn a header for an invite then "nonce" is the password for the registration right?
20:32.56exothermcwhat I don't get is it is like they can register with whatever they want.
20:33.24exothermcThey consistently use the same extension which is valid, but then after a few 401s they are in and making calls.
20:34.26gustononce is a word that is combined with the password and the hash of that combined with the hash realm/ip/domain and username is the answer
20:35.18exothermcgusto:   Ok so when I'm looking at a capture where do I see the password the host is sending?  Where in the header is it referenced?
20:35.39gustoexothermc: no one is sending passwords in clear/plain text
20:36.15gustomaybe you have not configured it right and their clients see that authentification is irrelevant so they do without
20:38.45exothermcgusto:   Ok I was just trying to see if I could figure out where they were getting the auth data from.
20:38.51exothermcI'm guessing tftp now.
20:38.53gustowas recently to mcdonald's and thaught about using SHA512 for hash for authentification instead of MD5
20:39.16gustoexothermc: ???
20:39.44exothermcgusto:   They guess the mac address of the hardware and download the phones config file.
20:42.29gustoexothermc: well, macaddresses are not easy to be guessed, because it is 48 bit, so that is some more than 32 bit, ... and macaddresses are irrelevant, because macaddresses you only use for local ethernet ... everything what is routed is IP only
20:42.42*** join/#asterisk wudles (~wudles@gateway.secureinstrument.com)
20:43.02WIMPyOr for auto provisioning.
20:43.38exothermcgusto:   Most default device profiles are a combination of some standard string and the hardware address of the device.  If you target specific hardware you can reduce the bits significantly.
20:44.39*** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com)
20:47.50exothermcgusto: Just looked back over a pcap of tftp and it doesn't look like that is the attack vector.
20:49.56*** join/#asterisk RypPn (~RypPn@unaffiliated/ryppn)
20:54.12*** join/#asterisk Johnny- (~John@70-90-6-154-BusName-pa.panjde.hfc.comcastbusiness.net)
20:54.42Johnny-hello
20:59.20Johnny-im lookin around trying to find the right hardware to buy to hook up 3-4 POTS lines
20:59.36QwellJohnny-: Google Digium TDM410
20:59.48Johnny-that's what I'm probably going to go with, Qwell
20:59.53Johnny-but I thought I'd ask about this one http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=170763740484&ssPageName=ADME:X:RTQ:US:1123#ht_2147wt_1163
21:00.00Johnny-I like the option to expand if I need to
21:00.12QwellIf you want to throw your money away, sure
21:00.19Qwellbut, Digium also sells 8 and 24-port cards
21:00.25QwellThat ebay crap is just that.  Crap.
21:00.26Johnny-could you elaborate on throwing my money away
21:00.34QwellIt's a piece of shit clone.
21:00.38Johnny-thought so
21:00.47Qwellhttp://store.digium.com/products.php?category_id=1
21:02.03Nuggetwhatever happened to the good* old days of buying a clone X100P "faxmodem" off ebay?  :)
21:02.13Nugget* (for sufficiently sucky values of "good")
21:02.18[TK]D-FenderThey were never good...
21:02.25[TK]D-FenderKinda like ... telnet ...
21:02.26Nuggettelnet is eeeeeeevil!
21:02.29Nuggethee
21:02.31[TK]D-Fenderpushes some more buttons
21:02.38WIMPyAnalog is evil
21:03.07gustowhy?
21:03.20NuggetBBuutt  II  hhaavvee  aa  XX1100PP  aanndd  II  ddoonn''tt  hhaavvee  aannyy  jjiitteerr
21:04.12Johnny-if I may ask, what makes the clones shitty?
21:04.14WIMPyWhy? Is almost no signalling and a lot of guessing good enough?
21:04.52QwellJohnny-: the hardware is crap.  they require special drivers.  the "company" behind it will disappear in 3 weeks.  there is usually no support.  shall I continue?
21:05.19Johnny-I guess I'll just take your word for it
21:05.57Nuggeteven the real X100P was pretty awful, but the clones were even worse.  I'm so glad that era is behind us
21:07.40Nuggetunrealated, I ordered a D70 this week.  I'm looking forward to fooling around with it.  I doubt I'll swap out the cisco desk phones, but our customer service folks may get digium phones instead.  The ciscos aren't really handling high call volume like I'd prefer
21:08.36Johnny-good gried there are a lot of knock-off cards on ebay :S
21:08.40WIMPyYou want to use a camera as phone?
21:08.51Nuggetheh
21:09.14WIMPyscnr, but that name is reallt well known to everyone.
21:09.26WIMPys/llt/lly/
21:09.26Nuggetyeah, I heard qwell was working on chan_nikon.  shares plenty of code with the skinny stuff
21:12.09WIMPySomeone once did a camera with phone instead of the usual opposite. Can't remember who it was.
21:12.38*** join/#asterisk NightMonkey (~NightrMon@pdpc/supporter/professional/nightmonkey)
21:16.23Johnny-what about openvox?
21:16.48Qwellsame deal
21:16.54Johnny-dang
21:17.27Johnny-seems a least a tiny bit more legitimate at least
21:18.04WIMPyLeave the anlog shit behind. It's too expensive anyway.
21:18.50Johnny-hrm
21:19.17Qwellare you suggesting he get a PRI for 3 lines? O.o
21:19.59WIMPyUsually that makes 2 BRIs.
21:20.15QwellBRI is incredibly rare in the US.
21:20.38jpsharpAnd they're expensive.
21:20.55WIMPyYes, I guess it's hard to transport digital information on steam engines :-)
21:21.04jpsharpLast BRI I had was $190/mo
21:21.28Johnny-i'm open to suggestions on how to make this work. we're moving offices and leaving the old archaic phone system behind. we have 3 lines with verizon and 1 dedicated fax line, the fax doesn't need to be hooked up to the pbx
21:21.43WIMPyAt that price I'd take a PRI :-)
21:22.22jpsharpFractional T1.
21:22.33NuggetI'd find the budget for a PRI.  The benefits are innumerable
21:23.36jpsharp4 channels of voice, combine the other 20 channels into internet for the office.
21:23.58Nuggetyou'll be able to easily grow, you'll be able to do DIDs, set custom outbound callerid, and never have any analog headaches
21:24.41jpsharpOr some telcos offer fractional PRI.
21:26.11*** join/#asterisk oneadvent (~oneadvent@ip68-105-63-31.pn.at.cox.net)
21:28.09Johnny-well, this is a really small business, that might be a bit of overkill
21:35.49NuggetI moved us to a PRI before we had any employees.  It's that much better  :)
21:36.45*** join/#asterisk rdegges (u4891@gateway/web/irccloud.com/x-zohwwuornkowizku)
21:37.13rdeggesHeya guys, I've got an AMI question for any of you. I'm listening to hangup events in Asterisk, and for some reason--I always get two duplicate hangup events whenever a user hangs up a call.
21:37.21rdeggesBoth of the responses are absolutely identical.
21:37.26rdeggesIs there any reason why I'd be seeing this?
21:37.48rdeggesI don't get two duplicate events when I see a 'newchannel' event come in, only the 'hangup'.
21:38.37ideamanOn a Bash script in Ubuntu, I'm trying to make it completely automated, and for the Asterisk portion, I have apt-get install -y, but still get the ITU select option that requires interaction. Is this a question for here, or best for the Ubuntu room?
21:40.31WIMPyrdegges: I only get one, but there are additional hanguprequest/softhanguprequest events.
21:44.34*** join/#asterisk TimeRider (~steve@188-220-34-144.dsl.cnl.uk.net)
22:00.52gustoppl are talking about analog and so on ... what is digital? VoIP TelePhone?
22:01.00dijibfinally got it p3nguin it was the listen *:80 that was messing up i had it set to 10.0.0.1:80
22:01.14WIMPyVOIP or even better ISDN or SS7.
22:01.30gustowhat is SS7?
22:01.51gustoand is ISDN developed still or is it a technology that is dying out slowly?
22:02.00WIMPyThat's the stuff for the big boys.
22:02.14WIMPyThe whole PSTN is dying.
22:03.26gustoi still do not see a lot of VoIP providers, so what is dying? i do not know
22:03.41gustothe only thing where i see some growth would be mobile phones
22:05.02Johnny-from what I have read and people have told me, VoIP can be kinda unreliable
22:05.30Johnny-but I am the furthest person in here from being an expert
22:05.38WIMPyThe whole telephony thing is dying.
22:06.02WIMPyAnd yes, VOIP is bad quality son several ways.
22:06.18p3nguingusto: Digital is something entirely different.  It is more similar to an analog system than to a VoIP system, but it is not compatible with either.
22:06.21WIMPys/son/in/
22:06.44Johnny-what a handy bot
22:06.48WIMPyVOIP isn't digital?
22:07.14p3nguinIn the sense of a digital phone, no it is not.
22:07.36WIMPyThat's an interesting definition.
22:07.39fileVoIP itself isn't unreliable
22:07.45fileit's the network that it is run over that is unreliable
22:07.53WIMPyI very much doubt it can be analog.
22:07.56p3nguinWhile IP might be considered digital, digital phones are something entirely different than IP phones.
22:08.42*** part/#asterisk mjordan (~mjordan@nat/digium/x-powqyimhaskrryip)
22:08.53file(unless the network is tightly controlled in which case VoIP works perfectly fine)
22:09.12p3nguinWhen you talk about a digital phone, you're talking about something like, but not limited to, a Nortel, Lucent, or Panasonic phone that was used on a special digital phone system.
22:09.24WIMPyThe biggest issue is usually the delay.
22:09.43jpsharpjitter & packet loss, not so much the delay itself.
22:09.45p3nguinWhen you talk about VoIP phones, you are talking about phone used on IP networks.
22:09.53WIMPyAre you sure these phones really are digital?
22:10.19WIMPyI've seen system phones that were actually analog phones with some digital addons.
22:10.36WIMPyNo, the delay, not the packet losses.
22:11.14jpsharpI've run voip over 550ms VSAT links without a problem.  But it was a consistent 550ms.
22:11.19WIMPySometimes the delays are so high that you tart talking in to each other.
22:11.41*** join/#asterisk dijib (~root@bas10-kitchener06-1176002447.dsl.bell.ca)
22:11.46WIMPyAnd sometimes they delay is even high enough to make echo cancellers fail.
22:12.02jpsharpYes.  Anything over about 800ms gets ugly and the brain can't grok.
22:12.36WIMPyIt becomes irritating at about 100ms.
22:13.20p3nguinWhile digital telephony likely encompasses a lot of the stuff we do, a digital phone system is not the same as VoIP.
22:13.53WIMPyVOIP is not a phone system at all.
22:14.12WIMPyJust a class of technology.
22:14.13p3nguinI have no idea how to even respond to such a statement.
22:14.14jpsharpI dunno.  My system used to pass about 40k minutes from VSAT customers an nobody ever complained.
22:14.40*** join/#asterisk metabsd (454672ae@gateway/web/freenode/ip.69.70.114.174)
22:14.48p3nguinExcept maybe:  OF COURSE IT ISN'T.
22:14.59metabsdhi all. i have paging problem with aastra phone + asterisk 1.8.10 anyone can help me ?
22:15.12p3nguin1.8.10.0?
22:15.16p3nguin1.8.10.1?
22:15.24WIMPymetabsd: Only if you tell us about it.
22:15.34metabsdThe problem is when i make page it's working but when i hangup the source phone all destination phone stay in line ...
22:15.36p3nguin1.8.11.0 is out.
22:16.02metabsdp3nguin: yes i now im compiling right now to test this version
22:16.18fileI don't believe page was touched at all between those.
22:16.21*** join/#asterisk eicto (~morgoth@144-71.dsl.aichyna.com)
22:16.30file(or the underlying stuff used)
22:19.39metabsdk now im running 1.8.11.0 and i have the same problem
22:21.49metabsdgoogle is my frien : http://www.freepbx.org/forum/freepbx/beta-program-issues/paging-groutps-dont-hang-up
22:32.35metabsdThe solution is :: i need dahdi for meet-me application in asterisk because confbridge dont support hangup my phone right now ...
22:33.46WIMPydidn't even know Page() coule use anything else.
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