IRC log for #asterisk on 20111205

02:43.32*** join/#asterisk infobot (~infobot@rikers.org)
02:43.32*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-rc2 (2011/11/15), 1.8.7.1 (2011/10/17), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org
02:43.39p3nguinOne way you could do it is to use an alternate extension to reach your phone, which sets the CALLERID(num) before it dials your phone.
02:44.01[TK]D-Fenderj-fish: ...
02:44.02[TK]D-Fender~book
02:44.03infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
02:44.06[TK]D-Fender^^^
02:44.10j-fishthanks:)
02:44.35p3nguinInstead of 600@phones, perhaps a600@phones.  Then extension a600 sets any callerid info you want before it dials you.
02:44.45p3nguinThat's just one idea.
02:45.24SeRiinfobot: YAY!!!!
02:45.43[TK]D-Fenderdijib: What do you want the CID to be?
02:46.05SeRiok p3nguin all set. reverted back and done with my cow build..... taking a brake :) going to eat....
02:46.25p3nguinWhen Originate() runs and calls his phone, it's just asterisk calling... without the real caller's info.
02:46.43p3nguinI'd imagine he just wants the real info to show up.
02:46.51dijibyes i do.
02:46.57p3nguinBut I don't know how Originate will do it on its own.
02:47.05p3nguinSo I'd use the alternate extension.
02:47.24dijibhuh alternate extension?
02:47.27[TK]D-Fenderp3nguin: If it shows up with no callerid, what do you imagine every other call he uses that peer for does?  Just takes device level CID..
02:47.43p3nguindijib: You should have been paying attention.  I'm not typing it again.
02:47.49[TK]D-Fenderp3nguin: Or he's setting it in the dialplan the same way can't can't stretch that logic to this process on his own.
02:48.11p3nguinAsterisk originates a call to his phone.  There is no callerid info at that point.
02:48.17dijibok i saw it a600
02:48.41p3nguinWhen Asterisk originates a call to my phone, it says External Call.
02:48.43dijibalso if originate fails i need to voicemail
02:48.56dijibmines all anon
02:49.23[TK]D-Fenderdijib: If originate fails... who's going to leave a VM?  It didn't connect.  There is nobody.
02:49.41p3nguinIf you use the alt extension, you can take the caller's info from the original call and set it between the Originate() and the Dial().
02:49.41dijibgotof
02:50.25p3nguinOriginate requires your channel to be answered before it will call the other number.
02:50.45p3nguinIf your channel does not answer, there is no call to the other phone, and there is no voicemail.
02:51.05p3nguinBut........
02:51.14[TK]D-FenderHis Originate is backwards.
02:51.31p3nguinThat's where I was going.
02:51.34[TK]D-FenderBut I figured I'd leave him to beat that around a while first
02:52.10p3nguinIf you reverse it, then it will call the other person first, and when that phone answers, it'll call you.
02:52.32[TK]D-Fenderdijib: You Originate to your target then dump them into your dialplan.  Instead you are Originating to * dialplan where you intend to answer and then trigger the callout.  The Originate will always be "successful" (or fail depending on how tragically you coded it).
02:53.31p3nguinOriginate(Local/${DB(callback/${CALLERID(num)})}@phones,exten,phones,600)
02:54.54[TK]D-FenderShould specify a priority in there...
02:55.04p3nguinIf none, assume 1.
02:55.10[TK]D-FenderEW
02:55.20p3nguinExtensions start at 1.
02:55.31p3nguinSeems safe/sane to me.
02:55.39[TK]D-Fender~assume
02:55.39infobotAbout assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight&quote=assume.txt&file=assume.wav  It makes an (ass) out of (u) and (me)
02:55.51p3nguinAlternatively, you may specify a priority if you want.
02:56.25SedoroxAnyone happen to know if it's possible to stop a Polycom IP650 from flashing the BLF light when a phone associated with that line is not registerd? (I'm not sure if this would fall on the Polycom side, or the Asterisk/Switchvox side)
02:56.50p3nguinHaving the originate reversed is my fault.  I have a tendency to call my side first when originating so that I hear the ringing while calling the other person.  I don't like sending a call out, get an answer, and then having ringing.
02:57.01p3nguinIf someone called my phone and it started ringing, I'd hang up.
02:57.13p3nguinSo that's why I did the originate the way I did.
02:57.36p3nguinIt's an easy change, which I already listed, to turn it around.
02:58.43[TK]D-FenderSedorox: Change the indication pattern in your provisioning
02:59.04p3nguinI'm going to go have some tomato soup and grilled (baked) cheese sandwich, so I'll be back soon.
03:00.16[TK]D-Fenderp3nguin: he's doing an automated callback to the caller.  2 ideas depending on the goal.  If he wants to just let the caller call in free as inbound, then caller first, then internal.  If he wants an agent to call him back and know that the agent is there then the other idea works.
03:00.21SedoroxI don't think I can change that manually (at least easily), although I'm running Switchvox 5.1.2, which is handing the phone provisioning
03:00.25dijibthanks p3nguin that was very helpful
03:00.30dijiband [TK]D-Fender
03:00.38dijibim going to switch the context now.
03:00.39[TK]D-FenderSedorox: That's what you've got to do.
03:05.17Sedoroxwonder if that is dug in the GUI somewhere.. I wouldn't have a problem manually editing the stuff, I just want to make sure if there were any changes done on the system, that it carried over
03:07.27*** join/#asterisk mindCrime (~chatzilla@cpe-076-182-089-009.nc.res.rr.com)
03:08.13[TK]D-FenderSedorox: I can't imagine any PBX GUI would cover anything like this.  You are cheating standard indications...
03:08.36Sedoroxgood point :)
03:09.56Sedoroxfor this setup, the majority of extensions aren't going to be a problem (hardphones in the office), however they have about 5 right now that will most likely be softphones, and not always registered
03:10.11Sedoroxso lines flashing on the sidecars I'm sure will get annoying
03:10.13Sedoroxbut we'll see
03:12.44[TK]D-FenderSedorox: I manually do my receptionist's directory and was negligent for months (as in lots).  We had a lot of turnover in that time and people changing ext #'s,  Her phone was a Christmas Tree before I got off my ass to fix it up right...
03:14.01Sedoroxlol, nice
03:14.43[TK]D-Fender3 loaded sidecars
03:15.31Sedoroxouch
03:15.37Sedoroxthis is only one luckily
03:16.42[TK]D-FenderSedorox: Do you have proper filesystem access on your box and to that folder?
03:17.59Sedoroxmm most likely
03:18.13dijibthis callback is ready for production. shall i pastebin? or you guys dont care.
03:18.16dijib?
03:18.31[TK]D-FenderSedorox: then you could simply override them in the <mac>-phone.cfg
03:18.51[TK]D-Fenderdijib: We already know how to do it... just tell us if you need help on failure :)
03:27.24dijibits working solidactually.
03:27.30dijiblike this
03:27.33dijibnow to populate.
03:27.44dijibwhich will save me like x7 the cost.
03:27.50dijibof the toll free for known callers
03:29.08dijibok ive got one for you, how do you dial an outside did and then send an extension argument for that did's pbx?
03:31.01[TK]D-Fenderdijib: "core show application dial" or on answer do it in your dialplan you toss them into
03:31.11[TK]D-Fender"core show application senddtmf"
03:43.18dijibthat sounds more like it
03:48.13SeRibbl finishing kids bday stuff (Angry Birds Theme)
03:48.24*** join/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com)
03:48.48FlyingbullGood evening everyone:)
03:50.21*** part/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com)
03:50.42*** join/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com)
03:51.23*** join/#asterisk Kobaz (~kobaz@its.kobaz.net)
03:51.38FlyingbullHi there, I was wondering if someone can help me figure out an error I'm getting with dahdi -- during the install it gives me an odd error, and I can't seem to figure out how to get around it.
03:52.12ChannelZlike what error
03:53.42FlyingbullOne moment, my screen crashed over there.  I'll give you the exact error:
03:54.54FlyingbullYou do not appear to have the sources for the 2.6.18-028stab091.2 kernel installed.
03:56.47FlyingbullI thought about going and getting the sources Centos.org, but I wanted to verify that it wasn't something else entirely.  I did read something about grabbing the source code that there was a patch for this problem -- but I can't seem to find a place to find that source code.
03:57.17[TK]D-FenderFlyingbull: Seems to say pretty clearly that you are missing your kernel source & headers
03:57.39[TK]D-FenderFlyingbull: yum install kernel-headers
03:57.45FlyingbullNope, I thought that was the problem and I went and checked, I had the kernel source and headers.
03:58.16[TK]D-FenderFlyingbull: re-run ./configure
03:58.45[TK]D-FenderFlyingbull: IIRC that is also sometimes given if you are missing ncurses-devel, etc
03:58.52[TK]D-Fendercheck the full dependency list for *
04:01.08FlyingbullOk, I had done that, but I'll go through it again, to make sure I got everything.  brb
04:07.27*** join/#asterisk LiuYan (~liu.yan@211.154.128.171)
04:14.54FlyingbullOk, what I have to wonder, does FreePBX even need the dahdi part for a pure voip network?
04:15.10ChannelZonly for MeetMe
04:15.49ChannelZthough I beg you not to go down the FreePBX road
04:17.20*** join/#asterisk gajini (~root@61.12.17.170)
04:17.33FlyingbullWhat is that? I'm setting up a "simple" IVR with it, so I was thinking it would work better.  I don't think I can do Queues with Asterisk -- with annoucnements every 30 seconds saying where you are in the queue..
04:17.41Flyingbullwhat s/b Why.
04:18.07FlyingbullNot by tomorrow morning anyway ;)
04:20.41ChannelZThat's how it starts, then you want to do something cool and you're stuck trying to figure out why FreePBX keeps erasing your configs
04:23.14FlyingbullWell that is danger when using frameworks.  You become an expert of the framework not the underlining technology.  While I've taken the time to read about Asterisk, time is of the essence at the moment.
04:23.40Flyingbullmake menuselect
04:23.54Flyingbullwront window -- damn multiple monitors screw me up once in a while.
04:24.22ChannelZMaybe you should just download AsteriskNOW
04:25.17FlyingbullUsing a remote server, or I would have.
04:27.35[TK]D-FenderChannelZThat's how it starts, then you want to do something cool and you're stuck trying to figure out why FreePBX keeps erasing your configs <-- jumping the gun on this...
04:28.11[TK]D-FenderFlyingbull: I don't think I can do Queues with Asterisk -- with annoucnements every 30 seconds saying where you are in the queue.. <- sure you can
04:28.44p3nguinMaybe he can't.
04:29.06p3nguinAsterisk can with ease, but that can't speak to his capacity.
04:29.27FlyingbullI can't at the moment, I've only been playing with Asterisk for a few weeks overall really.
04:29.29[TK]D-Fendercan != going to on his own
04:29.54p3nguinThe sample queues.conf has pretty good commenting.
04:30.01[TK]D-Fender!= guaranteed immediate
04:33.24FlyingbullOk, well it seems to me that would require a good knowlege of the dial plan as well, because I have the tree to consider:  if it is during a certain time, then announce that the queue is closed.
04:34.13[TK]D-FenderFlyingbull: All easy to do.
04:34.27[TK]D-FenderYourself or using FreePBX
04:36.06FlyingbullInteresting.  Well I don't like Framework from the perspective that they do their own stuff.  I guess I forego some sleep and look at that, the first question I need to figure out, do you have a sample of where I can look at an IVR for Asterisk?  It seems like google insists on sending me to the paid for advertising sites lately.
04:36.41[TK]D-FenderFlyingbull: it's in the book
04:36.44[TK]D-Fender~book
04:36.44infobotAsterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook
04:37.10[TK]D-FenderFlyingbull: Dump call in context.  Call WaitExten.  Have options they can dial in the context./
04:37.53*** join/#asterisk mintos (mvaliyav@nat/redhat/x-bccupyjiphguomaq)
04:38.27FlyingbullWell the plan is that they have the option to join the queue, leave a message, check store hours or hangup.  I mean it is pretty simple.  There will be 15 operators in the queue.
04:39.44FlyingbullThe next headache is setting up a predictive dialer.
04:43.20[TK]D-FenderFlyingbull: Decidedly less volunteers for that one...
04:43.56ChannelZhave fun with the treefrog
04:44.24[TK]D-FenderFlyingbull: as for queue options... exiting to leave a message is certainly doable.  checking store hours you'd have to rejoin the queue and lose your place, so better to ask before going to queue.  Also.. if you don't let them in outside of hours... you shouldn't need to mention the hours.
04:46.08*** join/#asterisk radic (~radic@dslb-178-002-225-173.pools.arcor-ip.net)
04:48.10p3nguinDuring business hours, I don't offer playback of hours.  When the call comes in, you have a chance to enter a person's extension or pressing a single digit for the directory.  If you choose to not ignore those, you head off to the queue, where phones start ringing.
04:48.39p3nguinif you choose to ignore, rather
04:49.16p3nguinOutside of hours, you there is a menu choice for playback of hours.
04:49.21Flyingbull[TK]D-Fender  Thanks for the book, already figured out part of the problem with Dahdl.  Well the idea was that the store hours they can hit 1 to speak with an agent, 2 for store hours and directions.   The deal is, it is for a pharmacy, and sometimes people just want to know they can come in to pick up their perscriptions.  otherwise they need to talk to someone to renew it.
04:49.57[TK]D-FenderFlyingbull: then make the hours an IVR option before hitting the queue (which is a separate options
04:50.22Flyingbull1 >> to speak with an agent is the queue.
04:54.24*** join/#asterisk troyt (~troyt@c-24-10-222-127.hsd1.ut.comcast.net)
04:55.48[TK]D-FenderFlyingbull: So "Hours" is not a queue exit option, only "Leave a VM"
04:56.44*** join/#asterisk dijib (~root@bas10-kitchener06-1279681924.dsl.bell.ca)
05:09.18*** join/#asterisk razu_ (~razu@195.222.7.35)
05:19.29connex[TK]D-Fender, sorry, had to get some sleep in. The tech support is helpful as my Spanish.
05:20.48*** join/#asterisk irroot (~gregory@197.172.59.133)
05:26.03jercosnomnomnomasterisk
05:33.53*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-147.cablep.bezeqint.net)
05:46.26SeRip3nguin: cower -d b43-firmware
05:46.27SeRicd b43-firmware
05:46.27SeRimakepkg
05:46.27SeRisu - root
05:46.31SeRipacman -U /home/builds/b43-downloads/b43-firmware-5.10.56.27.3-2-i686.pkg.tar.xz
05:46.53SeRiexit
05:47.02SeRip3nguin: and now we are up and running :)
05:48.15p3nguinEasy?
05:49.38SeRiyes sr!
05:50.14p3nguinIf you'll configure sudo for pacman, you'll be able to skip three of those steps.
05:50.38p3nguinYou'll run makepkg -i and it will do the rest for you.
05:51.18SeRinice
05:51.34FlyingbullCorrrect -- if someone hits 3 for store hours, they get returned to the main root, afterwards.
05:51.46p3nguinor you can use packer and skip like four or five steps.
05:51.48FlyingbullAnywy, thanks for your help, I'm going to see if I can implement it.
05:51.53Flyingbulllater.
05:51.59*** part/#asterisk Flyingbull (~Flyingbul@cpe-065-190-158-078.nc.res.rr.com)
05:53.56p3nguinI think I've decided that I probably shouldn't skip a second night of sleep in a row, so I'm heading off in a few minutes.
05:54.47SeRip3nguin: head to bead bro. have a good night sleep
05:55.13*** join/#asterisk kaushal (~kaushal@49.248.16.122)
06:01.25SeRip3nguin: I am out for teh night as well. I am connected to the wirless with my laptop... Now all of my devices are officially arch :)
06:01.40SeRiSleep and have a g/n
06:01.56SeRig/n #asterisk !
06:09.46*** join/#asterisk LiuYan (~liu.yan@211.154.128.171)
06:12.53*** join/#asterisk coppice (~chatzilla@m121-202-59-61.smartone.com)
06:18.12dijibg/n SeRi
06:23.17*** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247)
06:33.53*** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt)
06:41.54*** join/#asterisk irroot (~gregory@197.168.83.75)
06:52.15*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
06:54.38*** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net)
06:55.47*** join/#asterisk BuenGenio (~Gene@203.145.92.172)
06:57.02*** join/#asterisk vader-- (vader@c-68-83-57-218.hsd1.nj.comcast.net)
06:57.47dymmornings
07:15.34*** join/#asterisk Ad-Hoc (~nimbus@194.30.241.105)
07:27.16*** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4)
07:28.54*** join/#asterisk irroot (~gregory@196-210-140-50.dynamic.isadsl.co.za)
07:30.43*** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk)
07:35.03*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
07:35.07gavimobileis it a bad idea to build a newer version of say libri when an older version is already installed?
07:47.40*** join/#asterisk Takapa (vegard@svanberg.no)
07:52.21*** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk)
07:53.36*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
07:59.31*** join/#asterisk bjornts (~BTS@it010226.klientdrift.uib.no)
07:59.49*** join/#asterisk freeman_u (~freeman@gw.quart-soft.net)
08:00.41*** join/#asterisk stix (~stix@193.89.191.209)
08:01.34*** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
08:02.57*** join/#asterisk ChannelZ (channelz@burner.com)
08:04.35*** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190)
08:14.19*** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247)
08:15.45*** join/#asterisk waschtl (~waschtl@HSI-KBW-078-043-090-014.hsi4.kabel-badenwuerttemberg.de)
08:16.49*** join/#asterisk Faustov (~fst@gentoo/user/faustov)
08:22.13olliigavimobile: sounds like an update
08:22.34olliiif you install a newer version of libpri you might need to recompile * and dahdi
08:36.59*** join/#asterisk bulkorok (~bulkorok@85.183.36.36)
08:41.09*** join/#asterisk hehol (~hehol@2001:1438:1009:200:1566:5343:ebbc:4a96)
08:42.15*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
08:42.20gavimobileollii: thanks
08:42.25gavimobilethat's what I did!
08:42.28*** join/#asterisk Nasga (~Nasga@110.113.117.78.rev.sfr.net)
08:46.22dymAny way I can prevent this?  ERROR[23893] res_jabber.c: PubSub Server error, 503
08:46.37dymoccours quite frequently (1.8.7.1)
08:47.04*** join/#asterisk irroot (~gregory@196-210-140-50.dynamic.isadsl.co.za)
08:47.22*** join/#asterisk binbash_ (~peter@server.digitog.nl)
08:58.18*** join/#asterisk freeman_u (~freeman@gw.quart-soft.net)
08:59.58*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
09:04.15*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
09:10.28*** join/#asterisk kriegerod (~krieger@chaf.tenet.odessa.ua)
09:10.46*** join/#asterisk ferdna (~yup@cpe-67-10-220-35.elp.res.rr.com)
09:14.15paquestion: is it possible/easy to configure asterisk such that it creates a sort of "com" device, in order to let legacy fax software work with the pbx as it was an old landline?
09:14.21paor an old modem?
09:16.26bulkoroktry iaxmodem
09:16.39paah thanks!
09:16.48olliihttp://www.voip-info.org/wiki/view/Asterisk+IAXmodem
09:16.56bulkorok:)
09:18.01olliicould someone tell me about queue holdtime ? after reading * src code it seems like an avg value...is that value persistent or is it reseted after a restart of * ?
09:18.06olliiasterisk 1.4 and 1.8
09:22.12*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
09:23.47*** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com)
09:27.55*** join/#asterisk jetlag (jetlag@pool-71-168-248-89.cmdnnj.east.verizon.net)
09:32.42*** join/#asterisk ashd (~ashleyd@94-194-208-216.zone8.bethere.co.uk)
09:35.31*** join/#asterisk turtlefence (~turtlefen@CPE-144-132-156-210.eypg1.ken.bigpond.net.au)
09:38.39*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
09:40.20*** join/#asterisk verywiseman (~verywisem@unaffiliated/verywiseman)
09:40.54*** join/#asterisk LiuYan (~liu.yan@211.154.128.171)
09:42.01verywisemanin "call files" that is using for automatic dialout , can i put many extensions with suitable priority for each ?
09:42.44*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
09:43.36*** join/#asterisk LiuYan (~liu.yan@211.154.128.171)
09:51.30*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
09:54.35*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
09:58.58*** join/#asterisk ashd (~ashleyd@94-194-208-216.zone8.bethere.co.uk)
09:59.58*** join/#asterisk jkroon (~jkroon@196.25.195.42)
10:00.40jkroonhi guys, not sure if i'm just missing something or what is going on, but is it possible to get dynamic (realtime) sip peers from automatically re-apearing after an asterisk restart (probably a crash)?
10:02.45bulkorokwhat do you mean? Do the clients should register after crash?
10:03.09jkrooni have a peer, it registers so now I can send calls to it.
10:03.26jkroonasterisk -rx "core restart now" ... now the peer is no longer registered to asterisk
10:03.34jkroonand as a result I can't send calls to it.
10:04.24jkroonheck, sip reload is sufficient.  under normal operation this won't happen, however, crashes do (unfortunately) happen and I'd like to provision for them.
10:04.34bulkorokthe client can not know this. so you have to tell the client to register all 2 minutes or so.
10:04.39jkroonhttps://issues.asterisk.org/jira/browse/ASTERISK-6591 (they guy's tone is very harsh, but same problem)
10:05.36jkroonbulkorok, i don't want the client to have to reregister, i'd like to simply have asterisk store it's realtime peers as part of some database (either the rt database it got loaded from, or the astdb) and automatically reload them as part of any reload (permitting they haven't expired yet)
10:06.40*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
10:07.45*** join/#asterisk turtlefence (~turtlefen@CPE-144-132-156-210.eypg1.ken.bigpond.net.au)
10:07.48*** join/#asterisk freeman_u (~freeman@gw.quart-soft.net)
10:08.13bulkorokjkroon: I know that in extconfig you can configure the table "sipregs" where the reg-informations will be stored. maybe this can help you
10:08.31jkroonsipregs?  i'll def take a look at that thanks!
10:08.39jkroonsounds like what i need.
10:08.47*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
10:13.04*** part/#asterisk gajini (~root@61.12.17.170)
10:14.54*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
10:20.13*** join/#asterisk tomer67 (~tomer@95.143.243.2)
10:20.30tomer67hi all
10:21.07tomer67I want to create some rule or macro, which will be perform conference handling
10:21.37tomer67I want to have only one number, 5000
10:22.22tomer67when I call to it, I should type PIN and after that accoridng with my PIN should be transfer to properly conf room
10:22.28tomer67is it possible?
10:22.49tomer67one number multiple conf room ?
10:25.53dymtomer67: yes, of course. you just start the "meetme" application and then define conference rooms in the config
10:26.21dymcheck out meetme.conf in /etc/asterisk
10:26.38tomer67dym: but it isn't only one to one?
10:26.49tomer67one exten one conf room
10:26.51dymtomer67: depends on your dialplan
10:27.03dymyou can explicitly set Meetme(1000) for one conference room
10:27.19dymor you can just MeetMe to have the caller insert a number
10:27.45dymWell MeetMe() (:
10:28.03dymBut the rooms have to be defined on meetme.conf
10:28.23tomer67brb
10:29.54dymtomer67: http://www.voip-info.org/wiki/view/Asterisk+config+meetme.conf
10:34.35*** join/#asterisk TimeRider (steve@host81-136-216-215.in-addr.btopenworld.com)
10:37.26*** join/#asterisk TimeRider (steve@host81-136-216-215.in-addr.btopenworld.com)
10:38.51*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
10:41.22*** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net)
10:45.05*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
10:45.34*** join/#asterisk BuenGenio (~Gene@059148208218.ctinets.com)
10:48.06*** join/#asterisk turtlefence (~turtlefen@c58-111-144-182.thorn2.nsw.optusnet.com.au)
10:53.53*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
11:02.31*** join/#asterisk tzafrir_laptop (~tzafrir@local.xorcom.com)
11:14.08*** join/#asterisk Lovelu (BanglaCafe@202.126.124.58)
11:14.08Lovelu[ GrEEtiNgS EvERyOnE ]
11:15.13Loveluhi
11:15.32*** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no)
11:16.00*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
11:18.45*** join/#asterisk fromol (~n1x@mail.orient-logic.com)
11:19.02fromolhi guys , anyone is experienced in dlink fxo fxs configuration?
11:19.03dymLovelu: lovely. Hi.
11:19.13fromoli have one simple problem but can't understand
11:19.32dym~ask
11:19.32infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
11:19.59*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
11:20.14fromolokeya ,
11:20.23fromoli have problem about outgoing calls on fxo
11:20.42fromolwhen im ringing to number call is going different number and added additional number
11:20.54fromolbut where im cant see
11:23.43Lovelucan any one help me out transcoding to g729 to g23
11:25.45*** join/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
11:26.16*** join/#asterisk zerohalo (~zerohalo@74-94-141-204-ma.hfc.comcastbusiness.net)
11:26.43*** part/#asterisk gavimobile (~user@bzq-218-196-30.red.bezeqint.net)
11:27.21Loveluhmmm
11:27.22defsworkI have a very strange problem - got a site on 1.6.2.20 with 20+ extens and every now and then all the extens show no service and asterisk thinks they are "NOT IN USE"  - the networking is fine as * shows pings times still and I can ping etc..
11:27.47defsworkI stop/restart * every night just to see if it was a resource issue but it has had no effect
11:28.52defsworkI'm wondering if its a handset issue - they are all aastras - the same model
11:29.29*** join/#asterisk dom| (~domi@mail.tas.de)
11:29.58Loveluhmmmm
11:30.38tomer67dym: thank you
11:30.44*** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net)
11:31.56tomer67but in my configuration I have configured me room from 9100-9200
11:32.46tomer67and when I call on external number, for example 123800
11:33.07tomer67after type of PIN it should be transfered to properly meeting room
11:33.17tomer67so call to 123800
11:33.31tomer67pin 5555 ->room 9100
11:33.41tomer67pin 6666 -> romm 9150
11:33.43tomer67etc.
11:33.59tomer67I don't know if it is clear for you
11:39.30*** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net)
11:42.00*** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za)
11:42.44*** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za)
11:48.38*** join/#asterisk Cain (~Geek@unaffiliated/cain)
11:50.55*** join/#asterisk TimeRider (steve@host81-136-216-215.in-addr.btopenworld.com)
11:54.01*** join/#asterisk oej (~olle@static-213-115-251-100.sme.bredbandsbolaget.se)
11:57.07*** join/#asterisk AmirBehzad (~behzad@86.57.4.5)
12:04.02*** join/#asterisk puzzled (~patrick@2001:980:5e31:1:6ef0:49ff:fe50:659d)
12:12.33*** part/#asterisk AmirBehzad (~behzad@86.57.4.5)
13:10.21*** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162)
13:19.24*** join/#asterisk lcat (~lcat@187.45.254.174)
13:20.06*** join/#asterisk BuenGenio (~Gene@cm61-15-218-59.hkcable.com.hk)
13:24.56*** join/#asterisk hehol (~hehol@2001:1438:1009:200:1566:5343:ebbc:4a96)
13:26.20*** join/#asterisk james_zhu (~Administr@113.91.166.60)
13:26.56*** join/#asterisk bintut (~bintut@111.65.29.43)
13:27.06james_zhuhello
13:27.24james_zhui have a problem with digium 2 port E1
13:27.25james_zhu<PROTECTED>
13:27.34james_zhuanyone what is the problem?
13:30.54*** join/#asterisk FainaUkraina (~Gene@cm61-15-218-59.hkcable.com.hk)
13:32.09*** join/#asterisk rgsteele (~rgsteele@apo.aweber.com)
13:35.24*** join/#asterisk akrohn (~akrohn@38.101.60.42)
13:37.51james_zhu328]: l4isup.c:1559 t22_timeout: T22 timeout (No "circuit group reset acknowledge" from peer) CIC=1.
13:37.51james_zhu[Dec  5 21:37:05] NOTICE[23328]: l4isup.c:1559 t22_timeout: T22 timeout (No "circuit group reset acknowledge" from peer) CIC=17.
13:39.32james_zhu328]: l4isup.c:1559 t22_timeout: T22 timeout (No "circuit group reset acknowledge" from peer) CIC=1.
13:39.33james_zhu===anyone knows that? give me help
13:41.47gordonjcpis there a GUI for asterisk that will show pretty diagrams of call states as you dial up extensions and stuff?
13:42.00gordonjcpit doesn't have to be particularly practical or useful
13:42.04WIMPyjames_zhu: Maybe you should tell us what kind of line you've got.
13:42.08gordonjcpit just has to impress a PHB
13:43.04WIMPygordonjcp: Like FOP or astman?
13:43.24gordonjcpWIMPy: possibly, let me google
13:44.15oejAsterisk is 12 years today! Happy Birthday!
13:44.49gordonjcpWIMPy: that sounds like it'll do
13:45.19gordonjcpWIMPy: I need to do a swift bit of baffling with bull organic fertiliser product
13:45.20WIMPyPretty old and still so young.
13:49.58*** join/#asterisk mjordan (~mjordan@nat/digium/x-fpkvlndilsxkledc)
13:54.48*** join/#asterisk irroot (~gregory@196-210-140-50.dynamic.isadsl.co.za)
13:56.36*** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net)
13:59.24jkroonhi guys, I've asked this before but seemingly can't find it now.  COUNTRYCODE=27, so on an outbound call to 2712... I want to replace the leading 27 with a single 0, however, this needs to work generically, so the COUNTRYCODE needs to remain in a variable.
14:00.30jkroonin dialplan (if COUNTRYCODE was static) I could do exten => +27!,1,Goto(0${EXTEN:2},1) but alas, as soon as the 27 is stored in a variable it's a tad trickier.
14:00.45*** part/#asterisk james_zhu (~Administr@113.91.166.60)
14:00.46WIMPyNo problem as long as it is a global variable defined in your dialplan.
14:01.21WIMPyYou can do exten => +${COUNTRYCODE}.
14:01.34jkroonit is, so I can actually do exten => _${COUNTRYCODE}!,1,... awesome, let me test!
14:01.57WIMPyMissed the _, sorry.
14:02.16jkroons/+/_ :p
14:02.34*** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
14:02.49WIMPyYou had +27 in your example, so I intended to write _+...
14:03.23jkroonhehe, and I intended to write _27 to begin with, not +27 (those are stripped off long before)
14:03.56WIMPyAnd how do you know if it's an international number then?
14:04.54*** join/#asterisk as001 (~uros@82.117.198.142)
14:05.16*** join/#asterisk serafie (~erin@nat/digium/x-ylqjachnxsmvczle)
14:05.38as001Hello is it possible to configure to playback sound message to client which is in Queue during his conversation to Agent ?
14:07.22jkroonWIMPy, i always deal with the calls internally in full international format
14:07.29[TK]D-Fenderas001, Not though any normal process.  You'd have to brigde in a local channel to use chanspy, etc to do this
14:07.37jkroonso if I get 0XYZ I replace the leading 0 with COUNTRYCODE
14:07.51jkroonif I get 00Z or +Z I just strip off the leading 00 or +
14:07.57jkroonso far it works quite well.
14:08.10jkroonand I don't have to deal with NANP
14:09.56as001Can I redirect or transfer that call to some new extension where I will have just Playback and then to return that call to same agent who transfered call ?
14:10.33[TK]D-Fenderas001, Depends how you dial the agent
14:10.56as001agent is sitting in queue after AgentLogin
14:12.00[TK]D-Fenderas001,  Membermacro
14:12.34as001ok thanks
14:12.52*** part/#asterisk as001 (~uros@82.117.198.142)
14:20.31*** join/#asterisk [TK]D-Fender (~aoulton@216.191.106.162)
14:23.42*** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net)
14:26.34*** join/#asterisk retentiveboy (~pdugas@74-95-28-33-Atlanta.hfc.comcastbusiness.net)
14:32.22*** join/#asterisk mzahariev (adminimini@piem-nafta-v.unixsol.org)
14:35.40*** join/#asterisk cbwest (~cbwest@nat/cisco/x-ukknxitskcjokjzn)
14:35.44*** part/#asterisk cbwest (~cbwest@nat/cisco/x-ukknxitskcjokjzn)
14:38.00*** join/#asterisk jastrup (jastrup@login.konstant.no)
14:38.41jastrupDo any of you have any recommendations regarding a switchboard application for asterisk?
14:39.42*** join/#asterisk l2trace99 (~jr@74.118.40.1)
14:40.59*** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-42-45.inter.net.il)
14:42.01*** join/#asterisk ollii (~risker@2001:470:1f15:1384:62eb:69ff:fe31:b4)
14:42.40[TK]D-Fenderjastrup, FOP / FOP2 , astassistant
14:45.24davlefouhi,
14:45.52davlefouis it possible to show an history of the communication?
14:46.55WIMPydavlefou: Ask the NSA.
14:49.19dymor KGB
14:49.45WIMPyDo they still exist?
14:49.57[TK]D-Fenderdavlefou, what kind of history?  For what communication?
14:50.25davlefou[TK]D-Fender: yes!
14:50.27chuckfWIMPy: by that name, I don't think so
14:50.34[TK]D-Fenderdavlefou, EXACTLY
14:50.53WIMPyBut for telecommunication, Mossad is definitely 1st place.
14:50.55dymWIMPy: course they do
14:51.02dymyou
14:51.10dymyou're so uninformed!
14:51.15*** join/#asterisk ZogG_n900 (~michael@213.8.57.217)
14:51.17ZogG_n900hello
14:51.31dymHi ZogG_n900
14:51.38dymwhats cracking?
14:52.00dym(except from your telephony)
14:52.01ZogG_n900i have a lot of warrning kinda "NOTICE[4363]: chan_sip.c:9510 check_auth: Correct auth, but based on stale nonce received from ...."
14:52.23dymi suggest removing asterisk
14:52.30dymwould certainly get rid of that problem
14:52.31ZogG_n900dym, i hope for better future let's say like that
14:52.41ZogG_n900dym, yeah i know =)
14:53.02WIMPydym: Are you a medical doctor?
14:53.12ZogG_n900i googled the problem, it said that it's releated to snom certain setting while i have audio codec
14:53.23davlefou[TK]D-Fender: i want to looks about historique about the techincal message.
14:53.23ZogG_n900WIMPy hope not
14:53.23dymWIMPy: Philosophy
14:53.30ZogG_n900no head no pain =)
14:53.48WIMPyYes :-(
14:53.56dymdavlefou: /var/log/asterisk ?
14:53.58[TK]D-Fenderdavlefou, That does not describe anything specific.
14:54.05WIMPydavlefou: You need to be more specific if you hope for any sensible answer.
14:54.32[TK]D-FenderHOW CAN I EVERYTHING!?!? :S
14:54.43WIMPyJust do it!
14:55.05davlefouok, i install and i want to know if my call had works well with asterisk! I start to looks log!
14:58.11dymWIMPy: works for nike!
14:58.19dymKNEWIT
15:05.10*** join/#asterisk tompaw (~tompaw@slave30.tesserakt.eu)
15:05.31*** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart)
15:08.35*** join/#asterisk jboy1010 (~jhash1010@bas1-montreal19-1177815033.dsl.bell.ca)
15:09.49*** join/#asterisk n3hxs (~ed@63.68.135.4)
15:09.58jboy1010urgent: Need some help fixing what i beleive is a firewall problem with my asterisk server. Can't hear anything on any side of calls after a call is connected, willing to pay through paypal for help
15:13.00jboy1010anyone here?
15:14.48jkroonjboy1010, tcpdump is your friend.
15:19.46jboy1010i'm really new at this, i'll try looking into tcpdump, thanks
15:22.04WIMPyHas anyone ever seen the screening indicator "user provided, veryfied and faild" being used anywhere? If so, in what situation?
15:22.17Qwell~nat
15:22.17infobotnat is probably Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
15:22.19Qwelljboy1010: ^^^
15:22.41QwellFeel free to donate to the FSF once you've fixed your problem with that info.
15:24.49jboy1010well its weird because everything was working great until i tried to install openfire onto my asterisk box
15:25.00jboy1010(elastix pbx)
15:25.22jboy1010so i had to enable port 9090 and ever since then calls can connect but there is no voice transfer happening
15:25.31jboy1010Nat is enabled in config files
15:25.43Qwell"enable" how?
15:26.11jboy1010nat=yes in trunk settings
15:26.18Qwellno, the port 9090
15:26.26jboy1010oh
15:26.40jboy1010i ssh'ed into my box
15:27.00jboy1010and used "system-config-securitylevel-tui"
15:27.20Qwelland in the process probably blocked your NAT ports.
15:27.53jboy1010i did not change any setting that where already there, all i did was add a new setting for 9090 port forward
15:28.07Qwell9090 falls in the default NAT range
15:29.41*** join/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net)
15:29.46jboy1010i wasn't able to access the openfire installation dialogue before what i did, so after forwarding port 9090 openfire was ok but my calls arent working
15:30.06jboy1010this is in the elastix gui
15:30.09Qwellyes, because adding 9090 killed your NAT ports
15:30.51jboy1010well i tried removing the same rule i added but it didn't help..
15:31.21Qwellno, it wouldn't.  Since it overlaps, I'm betting it removed any other conflicting (read: NAT ports) rule.
15:32.04jboy1010so i have to reset my firewall rules?
15:32.08Qwellmaybe
15:32.09leifmadsen...and this is why we test on development servers and write deployment procedures
15:32.42*** join/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net)
15:33.23jboy1010anyone willing to help? its pretty urgent, i can pay through paypal
15:33.40Qwellpastebin your iptables rules
15:34.49jboy1010sure
15:35.22*** part/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net)
15:35.22[TK]D-FenderQwell, what range is 9090 interfere with?  Stock sample RTP is 10k-20K, and its nowhere near SIP...
15:35.24*** join/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net)
15:36.15Qwell[TK]D-Fender: my coffee is still too hot.
15:36.19Qwellmy point stands though
15:36.52*** join/#asterisk joshaidan (~brianj@S010600095bfe15df.tb.shawcable.net)
15:36.58tuxxieif a call is transfered does the  ${UNIQUEID} get reassigned?
15:37.10jastrup[TK]D-Fender: FOP2 is not an alternative, astassistan might be, but is there any others? Maybe enterprise ones?
15:37.53[TK]D-Fenderjastrup, HUD / HUDlite
15:39.49jastrup[TK]D-Fender: thanks
15:42.09*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)
15:42.30jboy1010Qwell, http://pastebin.com/YRRpRhsg
15:42.57Qwellit's not a firewall issue then.  check your router.
15:43.24QwellYour default policy is accept.
15:43.35Qwellalso, you have your NAT ports using tcp, which is wrong
15:45.53[TK]D-Fenders/NAT/RTP
15:46.09QwellCOFFEE IS TOO DAMN HOT
15:46.19Qwellmore caffeine needed, stat.
15:46.33jboy1010i don't get it.... these firewall settings worked perfect before. I didnt change a thing, except for the openfire isntall.
15:46.45Qwellshrugs
15:47.10jboy1010100$ paypal?
15:47.17WIMPyDon't use shiney tools that do what they want without you knowing.
15:47.23Qwelljboy1010: See above, re: FSF
15:47.41Qwellor I can PM you a link for DWB
15:48.11jboy1010sure
15:48.26[TK]D-FenderQwell, http://www.quickmeme.com/meme/35f1p8/
15:48.33*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
15:48.39Qwell[TK]D-Fender: heh, nice
15:51.49*** join/#asterisk Defraz (~Defraz@corp.fuzecore.com)
15:53.26*** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247)
15:55.51*** join/#asterisk wcselby (~wcselby@pdpc/supporter/professional/wcselby)
15:55.54wcselbyo/
15:57.38*** join/#asterisk akrohn (~akrohn@38.101.60.42)
15:58.06akrohndoes anyone know how compatible 1.8 and freebsd are these days?
15:59.43pabelangerakrohn: should work, we use it for testing our remote bamboo build agents
16:03.14carrarnot the BAMBOO BUILDS!!!
16:05.27wcselbyanyone know the default max size of a CNAM record?
16:05.55*** join/#asterisk ik_5 (~ik@109.226.17.43)
16:06.00ik_5hello
16:06.11akrohnexcellent. thanks pabelanger
16:06.52ik_5i have an incoming call from a SIP trunk that arrive with the proper caller id, but Asterisk sends invite back with From: "Anonymous" <sip:Anonymous@anonymous.invalid> . how can i figure out why ?
16:07.18Qwellwcselby: 255
16:07.26pabelangercarrar: vs?
16:07.32QwellI'm wrong.  255 * 63?
16:08.08wcselbyi was trying to figure out max char that is retrieved from a CNAM dip, but thanks :)
16:08.19Qwellthat's a completely different question
16:09.08carrarVS ALIEN!
16:09.11QwellThe theoretical limit for CNAM is bigger than valid response limits.
16:09.15r0m|uwaz up wcselby. God is cold!
16:09.15[TK]D-Fenderik_5, sHOW US
16:09.40[TK]D-Fenderdern capz...
16:09.41wcselbyQwell obviously
16:09.42Qwellwcselby: It would be invalid to return something that isn't a valid record (A, MX, etc)
16:09.44wcselbyr0m|u sup
16:09.52wcselbyQwell wait what?
16:10.01wcselbyCNAM, not CNAME
16:10.03QwellCNAM is a pointer.  It has to return another record.
16:10.07r0m|uwcselby: nothing much man just cold... :(
16:10.13wcselbycallerID name
16:10.18wcselbyafk, stupid meeting
16:10.20Qwellumm
16:10.25Qwelldon't use the word record then. :P
16:10.51ik_5[TK]D-Fender, show what exactly ?
16:11.08defsworkwilly ?
16:11.54*** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl)
16:14.30[TK]D-Fenderik_5, These calls so we can compare
16:14.35[TK]D-Fenderik_5, * CLI w/ SIP DEBUG
16:14.37[TK]D-Fender~pb
16:14.38infobot[~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude.
16:14.39[TK]D-Fender^^^
16:15.57*** join/#asterisk oej (~olle@87.96.134.129)
16:17.12*** join/#asterisk ChannelZ (channelz@burner.com)
16:18.31*** join/#asterisk jetlag (jetlag@pool-71-168-248-89.cmdnnj.east.verizon.net)
16:19.17[TK]D-Fenderbai bai
16:20.16*** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage)
16:20.16*** mode/#asterisk [+o leifmadsen] by ChanServ
16:23.30*** part/#asterisk mzahariev (adminimini@piem-nafta-v.unixsol.org)
16:24.45*** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com)
16:26.12*** join/#asterisk mandla (~quassel@168.167.180.161)
16:28.58*** join/#asterisk navaismo (~navaismo@189.230.118.2)
16:40.00*** join/#asterisk vinhdizzo (~vinh@dhcp-v016-024.mobile.uci.edu)
16:44.21*** join/#asterisk irroot (~gregory@41.51.173.137)
16:44.38*** join/#asterisk brdude (~brdude@12.155.183.30)
16:47.02*** join/#asterisk IsUp (~nocturne@unaffiliated/isup)
16:47.13IsUphello
16:51.15*** join/#asterisk oej (~olle@87.96.134.129)
16:53.49r0m|uquiet morning it seems.
16:55.24leifmadsenapparently all the asterisk systems are running fine
16:55.54r0m|uhehehe lol :)
16:55.58r0m|uindeed
16:58.26WIMPyOr everyone os busy fixing them.
16:59.02WIMPyis
17:12.22*** join/#asterisk eZz (~ez@178.137.178.86)
17:12.42eZzhi
17:12.44*** part/#asterisk stevedude77 (~stevedude@50-76-3-4-static.hfc.comcastbusiness.net)
17:19.25eZzguys I have a problem with originate from AMI to SIP channels.
17:19.31eZzWhen I'm sending a request to Local channel, with some context and exten and Dial to SIP - it's fine (I see ... Executing [s@dialout:1] ...)
17:19.39eZzBut when I'm using SIP channel, I'm getting ... Executing [@dialout:1] ...
17:19.48eZzI don't understand, is it a bug or I
17:19.58eZzI'm wrong
17:20.19WIMPyShow us what you send.
17:20.40eZzdo you mean Channel,Context,Exten in AMI request ?
17:21.07eZzone second please, will publish on pastebin
17:21.12WIMPyThe whole request
17:21.25eZzsure, one moment pls
17:23.35*** join/#asterisk darkskiez_ (~dz@cpc4-broo7-2-0-cust167.14-2.cable.virginmedia.com)
17:32.27eZzWIMPy: http://pastebin.com/FkCg7kkV
17:32.32eZzhere is a link
17:35.43WIMPydoesn't spot anything obvious.
17:36.54ZogG_n900<PROTECTED>
17:37.06ZogG_n900what is this error about i get a lot of output
17:37.14ZogG_n900any ideas tips?
17:37.18ZogG_n900audio-codes phones
17:37.21oejYou're device is trying to authenticate based on an old challenge-response
17:37.29WIMPyBut it looks like it execues , even if it doesn;t say so.
17:37.34oejIt needs to read the current auth message and respond
17:38.07ZogG_n900as i get every second from all exts
17:38.28*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
17:38.34eZzWIMPy: even I tried to use _. to catch everything and DumpChan and I did not found an extension too
17:38.43ZogG_n900may it ruin the quality as it uses traffic and actually it's not good?
17:38.55QwelleZz: _. is 1 or more characters.
17:39.10ZogG_n900oej should i update fw or is there way to fix it?
17:39.17eZzyes I know, it should be s
17:39.31WIMPyeZz: But you second log also shows s being executed ifven if it says [@.. in the log.
17:40.44QwelleZz: Are you sure that the entire originate action is being received by Asterisk?
17:40.46eZzyes but it is a strange...
17:40.56eZzQwell: yes
17:41.18eZzbtw that was working on 1.6.some
17:41.45eZznow I'm using 1.8.7.1
17:43.24QwellAre you willing/able to test earlier versions of 1.8, to see if you can figure out where it started happening?
17:43.25oejI don't know if there's an update of your phones. It's the phone firmware that retries an old authentication nonce
17:43.43Qwellit might also be useful to try the latest 1.8.8.0 RC
17:44.40eZzI will be able to test it but some later. Now I have to use Local trick to localize a problem for now
17:45.11eZzalso I will test IAX the same way
17:45.22WIMPyeZz: Why do you think it does't work? From your logs both attemts look successfull.
17:46.33eZzWIMPy: I don't like to see that it's working 'somehow'. It should work as designed but not somehow
17:47.33eZzthere are 2 cases: 1) I'm wrong or doing some incorrect ways, 2) it's a bug
17:47.41WIMPyI don;t see any issue except for an inaccuracy in the console output.
17:47.43eZznothing else
17:48.19eZzok, I will do another test, one sec pls
17:48.49*** join/#asterisk dijib (~root@bas10-kitchener06-1279681924.dsl.bell.ca)
17:48.55*** join/#asterisk anthm (~anthm@freeswitch/developer/anthm)
17:49.01*** join/#asterisk honree (~honree@net2.icemans.co.uk)
17:49.02honreehi
17:49.20honreeis it possible to change the callerid for console orginated calls?
17:49.39honreeit seems i can change the name but not number (which shows as unavailable)
17:50.03honreei do this :
17:50.05honreeexten => 229,1,Set(CALLERID(name)="TEST NAME")
17:50.05honreeexten => 229,2,Set(CALLERID(num)="1234")
17:50.22honreetest name appears on the phone's display below 'UNAVAILABLE'
17:50.49honreeif i call from a sip fone it displays the callers's number (222 for example)
17:50.50[TK]D-Fenderhonree, You are never supposed to put quotes for those
17:50.55eZzWIMPy: ok, I did another test. I changed 'Extension: s' to 'Extension: 9529'. It should hit '_X.' isn't it ?
17:51.02eZzbut it's not
17:51.05wcselbyi hate stupid meetings
17:51.09honreeok let me try
17:51.11[TK]D-FendereZz, No
17:51.17wcselbythis one is still going on
17:51.29WIMPyhonree: What TK said, and you might have to set CALLERID(pres)=allowed.
17:51.36[TK]D-FendereZz, Well not in that context.  show a complete sample please
17:52.06eZzExecuting [9529@dialout:1] NoOp("Local/9529@from-dialers-8afe;1", "Incoming call to 9529") in new stack
17:52.06honreeunquoted 1234 works :D
17:52.12eZzwhy it's hits on Local ?
17:52.18honreeis possibel to send alpha as well as num in that field?
17:52.18eZz[TK]D-Fender: it's to you
17:52.36eZzin this case:
17:52.36eZzexten => _X.,1,NoOp(Incoming call to ${EXTEN})
17:52.41WIMPyhonree: yes
17:52.42[TK]D-FendereZz, You are not showing us the whole picture and I don't trust the little dialplan segment youa re showing us
17:53.01honreedoesnt work :(
17:53.13[TK]D-Fenderhonree, Fix it right and show us
17:53.19honreeunquoted numeric works
17:53.28eZzwhat whole picture ?
17:53.31[TK]D-Fenderhonree, Good, that is what you should be doing
17:53.38eZzok it seems need to search myself
17:53.42honreewhat?
17:53.52[TK]D-FendereZz,It's clearly executing things you haven't shown us, in odd contexts.  Dump larger portions
17:54.11[TK]D-Fendernvm
17:54.13[TK]D-Fenderscratch that
17:54.22[TK]D-FendereZz, Ok, what wrong with it now?
17:54.41WIMPyI fail to see anything going wrong so far.
17:54.55honreewimpy : was your earlier 'yes' directed at my clid issue?
17:55.03WIMPyhonree: yes
17:55.17honreeive tried text in the num field and it doesnt seem to work
17:55.38[TK]D-Fenderhonree, And no reason to imagine why it should.  It says number, so give it a number.
17:55.48WIMPyhonree: Must be your phone then.
17:55.57honreei wasnt imagining - i was responding to what wimpy said
17:56.19honreetkd-fender can you see what wimpy is typing?
17:56.53honreewimpy it's the gigaset :}
17:57.04WIMPyWell, as "number" in SIP usually means account name, non-numeric characters aren't uncommon there.
17:57.17WIMPyThen the gigaset doesn't like it.
17:58.24honreeokey
18:12.45*** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za)
18:14.41*** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com)
18:18.41eZz[TK]D-Fender: ok, I published another one strange log. See it: http://pastebin.com/gTBEAEtz
18:18.51eZzmy brain is going crazy
18:19.10eZzI'm sure it's my hands but I haven't a clue where is an error
18:20.48eZzthis is a full dump, not a chunks
18:21.01*** join/#asterisk gpearson (~gspearson@fw1.niesc.k12.in.us)
18:22.37*** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-42-45.inter.net.il)
18:23.59eZzok nevermind, will figure out myself...
18:24.21*** join/#asterisk autofsck (~que@unaffiliated/autofsckk)
18:24.25[TK]D-FendereZz, I see a lot of missing parms in your AMI calls, no context, no priority, etc... Yuo are filtering things and I'd jsut as soon see your actual code that calls it
18:25.24eZz[TK]D-Fender: I did not listed them. The full params was listed here: //pastebin.com/FkCg7kkV
18:25.41eZzoops, http is missing, http://pastebin.com/FkCg7kkV
18:26.13eZz[TK]D-Fender: on that link you can see all the params that was not listed
18:28.12[TK]D-FendereZz, Show me your actual code that is generating the last call
18:28.18*** join/#asterisk cbwest (~cbwest@nat/cisco/x-vmryrbkmhaycokpz)
18:30.13eZz[TK]D-Fender: why you need this code ? the code is written in python, is working on a production the last 2 years on 1.6 and so on... Ok, I will paste a code
18:31.02[TK]D-FendereZz, Also when in doubt do a tcpdump on manager.  We need to see this raw.  Hand-made reconstructions often fail to represent what's really happening.
18:31.14[TK]D-FenderI can accpt just a TCP dump rather than raw code
18:31.20[TK]D-Fenderat least that's what * really sees
18:33.07eZz[TK]D-Fender: http://pastebin.com/dQTyS5dm
18:33.48eZznote: varList is empty so nothing is adding there
18:34.54[TK]D-FendereZz, Ok, new attempt with TCPDUMP would do it...
18:36.20eZzok I think it's just a waste a time... Will figure out myself but some later... [TK]D-Fender, WIMPy, anyways thanks guys for help
18:36.32[TK]D-Fenderok...
18:37.07eZzthe problem is somewhere on the surface but my tired brain can't find it... Just need to rest and find it
18:38.52[TK]D-FenderRest is always a good idea...
18:39.05eZzyeah
18:39.16[TK]D-FenderIf you're too bombed out you'll burn yourself even further while missing tings
18:42.25r0m|uI learn that the hard way :/
18:44.10*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
19:05.11*** join/#asterisk jkroon (~jkroon@dsl-241-250-126.telkomadsl.co.za)
19:07.41gpearsonOn a brand new * install 1.8.7.1, is it common to have to press # to dial an extension? If I dont then it takes 10 seconds to dial an extension, if I do then the extension rings instant. Where should I look as I am learning on the Fly.
19:08.22*** join/#asterisk devmikey (~irc@unaffiliated/devmikey)
19:09.24navaismogpearson: timeout in the phones dialplan
19:10.18*** join/#asterisk cbwest (~cbwest@nat/cisco/x-fxzwnhvrjemdovvo)
19:14.11Qwellp3nguin: You're slacking.
19:14.45*** join/#asterisk pietro (~pietro@88-149-227-4.dynamic.ngi.it)
19:20.18*** join/#asterisk KavanS (~KavanS@LINBIT/KavanS)
19:22.18*** join/#asterisk ASUChander (~asuchande@fl-71-52-2-90.dhcp.embarqhsd.net)
19:22.52*** join/#asterisk pdtpatrick (~pdtpdt@12.249.4.226)
19:23.04vader--whats the average time you guys have seen from verizon for porting a number?
19:23.04*** join/#asterisk scubes13 (~scubes13@cpe-024-168-196-000.sc.res.rr.com)
19:23.18pdtpatrickQuestion .. lets say ur PRI goes down or ur service provider d/c your account. Can Voicemail still work?
19:23.31ASUChanderHello all.  I'm trying to port a number away from a VoIP provider (FlowRoute), but the CLEC is telling me that the number (which is an NC 919 number) is outside of the "NC voice footprint" so they have to charge an additional $25/month for service for the ported number  - are they just trying to bill me extra?
19:23.37pdtpatrickfor instance can another server connect to the voicemail app using IAX or SIP ?
19:24.00ASUChanderFlowRoute has told me they support porting the number "away" from flowroute..
19:30.41gordonjcpwhat's an "ur PRI"?
19:31.08QwellI imagine it's like a sub-par US PRI.
19:31.15Qwellus-- pri
19:32.15gordonjcpis it like a primordial PRI?
19:32.38gordonjcpurstoff, primordial matter, that which classical matter is made of
19:32.42vader--ASUChander hehe I am actually moving a customer over to flowroute
19:32.55ASUChanderflowroute is great
19:33.05vader--how come you are moving away?
19:33.08ASUChanderBeen with them for awhile, but I have a customer that wants to move away
19:33.11ASUChanderIt's not me...
19:33.25ASUChanderBut they're screaming at me because of this $25/month fee that TWC wants them to pay
19:34.05vader--im actually tying to port two of my buddy's numbers over from verizon... trying to figure out how long that is going to take
19:35.16pdtpatrickgordonjcp, i was talking about my landline provider
19:36.10leifmadsenQwell: I asked him to turn off that auto-join-announcement when cbwest joins
19:36.23Qwellleifmadsen: boo :p
19:36.27leifmadsencbwest: are you a bot, or do you actually talk?
19:36.59gordonjcppdtpatrick: oh, okay, I just wasn't sure if an "ur PRI" was some groovy new presentation I hadn't played with yet
19:37.03QwellI've never seen it talk.  It disconnects a lot, which makes sense being on Ciscos network.
19:37.16leifmadsenQwell: heyo!
19:37.35gordonjcpQwell: his switch power supply is probably on fire
19:37.37gordonjcpagain
19:38.10*** join/#asterisk imox (~imox@91-64-181-64-dynip.superkabel.de)
19:40.12*** join/#asterisk jrose_atDigium (~jon@nat/digium/x-ivpegsbsjzlqdoys)
19:43.22*** join/#asterisk linuxplatform (~centoslin@88.87.48.115)
19:44.22*** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-42-45.inter.net.il)
20:07.29*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
20:07.48*** join/#asterisk andygraybeal (~andy@h28.215.22.98.dynamic.ip.windstream.net)
20:08.31andygraybealwhat is a good voip provider, for personal use.. mainly just to learn how to run asterisk and have a voip provider?
20:09.18[TK]D-Fenderandygraybeal, To learn, absolutely anything will do and doesn't even have to hit the PSTN
20:09.40[TK]D-Fenderandygraybeal, SIP is SIP.  The fact it might hit the PSTN doesn't change the quality or range of the test
20:09.46andygraybealok
20:09.52andygraybealthanks [TK]D-Fender
20:10.19[TK]D-Fenderandygraybeal, Ekiga.net for in/out testing.  ipkall for a free DID in a few zones in washington, etc
20:10.41andygraybealnice
20:12.19wcselbyI use flowroute, they're nice, cheap, prepaid
20:12.29wcselbyonly need like 20 to open an account with them I think
20:12.40wcselbyand the monthly cost of a did is like 1.49 I think
20:12.55wcselbyi always end up throwing 20 buck sinto the account every 2-3 months
20:13.01wcselbyand that's for my home number / business line
20:13.23wcselbysorry, all of that was directed towards andygraybeal
20:18.25*** join/#asterisk heffer (~felix@fedora/heffer)
20:21.11*** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net)
20:21.23*** join/#asterisk cbwest (~cbwest@nat/cisco/x-nztnxcpgnkxysahk)
20:21.52andygraybealwcselby: ah thanks man
20:22.29andygraybealwcselby: i grew up with some selby's here in southeastern ohio ;)
20:22.44andygraybeali'll look up flowroute
20:25.12*** join/#asterisk joshaidan (~brianj@S010600095bfe15df.tb.shawcable.net)
20:31.15*** join/#asterisk turtlefence (~turtlefen@c58-111-144-182.thorn2.nsw.optusnet.com.au)
20:43.17*** join/#asterisk PoWeRKiLL (~powerkill@IGLD-84-229-42-45.inter.net.il)
20:43.35vader--andygraybeal i like flowroute, they give you 25 cents to test your system with... Good rates and yuo can be setup with a DID in like 5 mintues
20:49.31*** join/#asterisk fprior (c8317ffd@gateway/web/freenode/ip.200.49.127.253)
20:55.56andygraybealvader--: nice i booked marked it - i will keep it in mind.  thank you guys.
20:56.18andygraybealwho spells bookmarked like that?
20:56.21andygraybealanyway thanks again.
20:58.49*** join/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld)
20:58.52DelphiWorld'Lo all
20:58.57DelphiWorlddo asterisk support the AMR codec ?
21:03.22fpriorhi all, what about MixMonitor, is the correct way to record all calls ? Is mandatory use StopMixMonitor() for each call ?
21:04.53r0m|up3nguin: ping
21:07.07DelphiWorldack r0m|u Dynamic fake firewall:P
21:07.58*** part/#asterisk DelphiWorld (~VoiceClou@openvpn/user/DelphiWorld)
21:12.22leifmadsenI wish delphiworld would actually help himself once in a while and actually look
21:16.18*** join/#asterisk s[X] (~mark@ppp118-208-82-70.lns20.bne4.internode.on.net)
21:16.20*** join/#asterisk ariel_ (u3533@pdpc/supporter/active/abatista)
21:16.42ariel_Hello everyone
21:17.40ariel_Is there a way from the dial plan to setup an extension that lets you reload asterisk or the queues?  Like doing an asterisk -rx "dialplan reload"
21:18.23p3nguinYes.  Create a new extension that runs that command via System().
21:19.25p3nguinleifmadsen: He's blind, so looking might be rather difficult.
21:19.55ariel_I was hoping not to do it via a system() call but directly from the dial plan
21:20.00leifmadsenp3nguin: perhaps "look" was the wrong word; I meant investigate
21:20.16p3nguinariel_: Do it from the dial plan.  Use System().
21:20.26leifmadsenariel_: only via AGI(), SHELL(), System() etc.
21:20.38leifmadsenthere is no DIALPLAN_RELOAD() function
21:20.53p3nguinThey are dial plan apps, so choose one and use it.
21:21.03ariel_OK, t/y I am already doing it via a system() call.
21:21.25p3nguinIf you're already doing it in the dial plan, what's the problem?
21:21.40ariel_did not want to put that much load on it.
21:22.00p3nguinqwell: leifmadsen had me turn off the announcement for cbwest.
21:22.17Qwellp3nguin: I saw that.  What a nub.
21:22.37p3nguinariel_: Are you sure you understand what you're saying?
21:22.45ariel_yes
21:22.51p3nguinI have my doubts.
21:23.15ariel_System(/usr/sbin/asterisk -rx reload)
21:23.23p3nguinYou asked how to do it via dial plan, and you've been given three apps to do it.  Yet you still keep asking.
21:23.34ariel_is what is there now, just don't need to reload everything cutting it down to just the dial plan right n ow
21:23.36p3nguinSystem(asterisk -rx "dialplan reload")
21:24.07ariel_p3nguin: yes, thank you.
21:29.16*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-218-155-146.cablep.bezeqint.net)
21:31.08*** join/#asterisk singler (~singler@84.15.129.49)
21:32.24r0m|up3nguin: you in?
21:32.33p3nguinsure
21:33.14r0m|uI have the script. I am going to pb.
21:33.51*** join/#asterisk cbwest (~cbwest@nat/cisco/x-ljtcqggohjnuepvx)
21:34.07p3nguinis leery of scripture.
21:41.21r0m|up3nguin: http://pastebin.com/kjyutmWB
21:41.59r0m|uthat will emil. But you can take out the email portion and have it echo under on your bash...
21:42.16r0m|uI have it set on my bash to just display fs space. and the rest to be emailed
21:42.54r0m|uthe email portion is set on a cron
21:43.01MaliutaI fail to see how that script is asterisk related
21:43.28Qwellthrows a paper airplane at cbwest
21:43.54r0m|uMaliuta: It was not directed to you. It was to p3nguin. We had a conversation about this script :)
21:45.30*** join/#asterisk turtlefence (~tsmart@110.76.135.10)
21:46.46p3nguinr0m|u: Oh, that script.  I couldn't figure out what script you were talking about.
21:47.20leifmadseneyes the ban button on cbwest
21:47.23r0m|up3nguin: :) Yes that script :P
21:47.34Qwellleifmadsen: according to infobot, he's talked.  once.
21:47.40leifmadsenhow many years ago?
21:47.47Qwell3 weeks ago
21:47.53Qwellwants to use Asterisk 10 packages O.o
21:48.05leifmadsenQwell: I asked for it in years
21:48.19r0m|uIs a disguised!
21:48.19leifmadsen(0.057496 years is the answer)
21:48.33Qwell...308
21:48.35*** join/#asterisk thebitguru (~Adium@50.93.209.154)
21:48.46Qwellwait, where'd your 4 come from?
21:48.46p3nguin0.057692308 years
21:48.54Qwellsilly Canadian years.
21:48.54leifmadsenThe Google is fun
21:49.02leifmadsenI typed in:  3 week / 1 year
21:49.10leifmadsen(3 weeks) / (1 year) = 0.0574960946
21:49.14Qwell3 weeks in years
21:49.15Qwellboom
21:49.18Qwelll2google, sir :p
21:49.26Qwell3 weeks = 0.0574960946 years
21:49.40p3nguinI used 3weeks/52weeks
21:49.44leifmadsenp3nguin: ah
21:49.49Qwellp3nguin: same, at first
21:49.52leifmadsenI wonder if it takes in account for a leap year
21:49.58Qwellit would
21:50.01leifmadsenQwell: lies
21:50.06Qwell1 year = 52.177457 weeks
21:50.19Qwell1 year = 365.242199 days
21:50.26thebitguruHi, I have recently started having a problem with my PIAF install where I can't hear the ring tone when calling out, and no audio with outbound calls, but inbound calls seem to work OK.  I am thinking that this is probably related to firewall, but I am not sure what it might be.  Any ideas?  I have ports 5060 TCP/UDP and 10k-20k UDP open
21:50.26Qwellit knows leap second even
21:52.42leifmadsenhuh, asterisk does not like loading modules with:  <description></description> or <description />, but using <description><para /></description> is fine
21:52.54leifmadsendtd must not be quite right
21:53.00Qwellleifmadsen: it dislikes the empty description
21:53.16QwellI remember hitting that a while back
21:53.20leifmadsenQwell: yes, I understand that :)
21:53.37Qwellthe <para/> makes it not empty.
21:53.39leifmadsenI'm trying to think where the best spot is to "fix" that
21:53.46leifmadsenQwell: I also understand that, which is why I put in <para />
21:53.51Qwelllamesauce
21:53.53leifmadsenbut I don't think that is the right fix
21:54.03QwellPutting a real description would be a good start. ;p
21:54.10leifmadsenI don't have time for that right now :)
21:54.23leifmadsenoh well, it's just a warning really
21:54.28leifmadsenwell, not even a logged notice
21:54.39leifmadsenjust console noise on start up
21:56.55Qwellmain/pbx.c, probably remove this:
21:56.59Qwell<PROTECTED>
21:57.09QwellI totally lied.
21:57.45Qwellmoves on
21:58.53*** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net)
21:59.00*** part/#asterisk turtlefence (~tsmart@110.76.135.10)
21:59.18*** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net)
21:59.28*** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net)
21:59.42dijibanybody have a working wakeup call script running?
21:59.58p3nguinfprior: I don't know if anyone answered you or not, but when the call ends, MixMonitor() also ends.  You only need to stop MixMonitor() manually if you need to stop recording while dialplan progresses.  Some people might use a dial plan that starts MixMonitor when a call comes in and record the entire call as it goes to a queue and whatnot, but if the caller enters a personal extension the record would stop.  That's just an ...
22:00.04p3nguin... example, but there are other cases where someone would need to stop the recording when doing different things.
22:04.16r0m|udijib: make one up.
22:04.29r0m|ushouldnt be that hard :)
22:05.24r0m|udijib: the link to leifs book that p3nguin gave you has examples of wake up calls
22:05.47r0m|utime to go,
22:05.55r0m|ucya in a bit!
22:06.22p3nguingives an odd look at "inbound call: +905548142836 <905548142836>"
22:07.13p3nguin+9 ?  That's a new one.
22:07.50_Corey_I think it's turkey
22:08.00p3nguin+90 Turkey
22:08.06p3nguinI think you might be right.
22:10.49honreeis it possible to make a console call to a sip extension, and hang up the call as soon as the called party picks up?
22:11.15honreei spose i could do it by playing a short announcement...
22:11.26p3nguinYou can make the call to a SIP *phone* and then make it hang up.
22:11.47honreeyea, whatd i say? ;)
22:11.57p3nguinYou said SIP extension.
22:12.10p3nguinchannel originate SIP/whoever application Hangup
22:12.12p3nguinTry that.
22:12.36honreeoo
22:12.50honreehang on then...
22:13.33*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
22:15.08*** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194)
22:15.36honreemmmm that doesnt quite do what i want.... ive got an entry in extensions.conf that is like a dummy number that allows me to chnage the clid
22:15.55*** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com)
22:15.58p3nguinHow do you want to incorporate that into this call?
22:16.04honreethe purpose of all this is to use a sip fone as a programmable alarm
22:16.21p3nguinDoes that extension end up Dial()ing the phone?
22:16.31honreeyus
22:16.38p3nguinOkay, that's easy, then...
22:16.51p3nguinchannel originate Local/123@context application Hangup
22:17.00p3nguinwhere 123 is the extension in context
22:17.09honreeok just a mo
22:17.38p3nguinAs soon as there is an answer, Hangup will run.
22:17.48honreeawesome
22:17.52honreeworks a treat :D
22:18.02honreethanks :)
22:18.30*** join/#asterisk nix8n82-phone (~AndChat@75-174-157-152.chyn.qwest.net)
22:18.36p3nguinYou could also make it play back a sound file.
22:18.54p3nguinchannel originate Local/123@context application Playback your-announcement
22:19.10p3nguinchannel originate Local/123@context application Playback silence/1&your-announcement
22:19.21honreecool
22:20.02p3nguinOnce the Playback() ends, the call ends.
22:20.25honreeright
22:20.37p3nguinIf you need more elaborate things to happen, you can create another extension to do things...
22:20.45*** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk)
22:20.51p3nguinchannel originate Local/123@context extension 5555@alarms
22:21.22p3nguinWhen the phone picks up, extension 5555 runs.
22:21.23honreeso once the called party picks up they get connected to 5555...
22:21.27honreeright
22:21.31honreevery cool
22:21.46p3nguin5555 could do Playback(), or anything, really.  Even Dial() another phone if you wanted.
22:22.18honreei have quite an old version of asterisk i probably should update...
22:22.30honree1.4.4
22:22.34leifmadsenwoh
22:22.37honreeheh
22:22.46p3nguinIt could even be used for allowing the called party to press a key to talk to a person, or press another key to do something else.
22:23.18*** part/#asterisk mjordan (~mjordan@nat/digium/x-fpkvlndilsxkledc)
22:25.11pabelangerleifmadsen: I did not see that coming
22:25.23leifmadsenpabelanger: I almost typed that too :)
22:27.12*** join/#asterisk darkskiez_ (~dz@cpc4-broo7-2-0-cust167.14-2.cable.virginmedia.com)
22:32.48Netgeekspfaw, 1.4.4 is relatively new.  I still have a 0.95 asterisk running
22:33.11Qwellpfft, versioned releases?
22:33.24Netgeeks^^
22:34.20honreeo
22:34.58Netgeeksi should have added a ;) at the end of that just to be sure no one took me serious
22:35.24honreeive had it installed for ages but only used it with a pretty basic config - couple of sip fones and a sip pstn gateway
22:36.32Netgeeksat least you didn't say 'I've got a fairly old trixbox install'.....
22:36.53honreeheh
22:37.25Netgeeksthe only response I have to that line is, 'Where do I send the flowers'?
22:37.32QwellNetgeeks: The heads on stakes outside deter those people now
22:37.42Netgeekslol
22:38.15Netgeeksi got a call recently from a guy who wanted me to fix his asterisk install, he was trying to do some fancy stuff like chanspy and such and it wasn't working right
22:38.21honreepushes his trixbox under the desk with his toe and whistles...
22:39.07NetgeeksI get in there to find out he had installed trixbox to get asterisk, then deleted all the trixbox config garbage and wrote his own dialplan....  he said that was easier than just installing a clean unix repo and then asterisk... I cried
22:43.22fpriorhi all: howto manage asterisk failover where telefony is core business, like call center, but when finance don't permit a second Asterisk server for HA nor precious motherboard for using with VMware Esxi (for example) ?
22:44.07QwellYou can't failover without a box to failover...to.  That doesn't even make sense.
22:44.48QwellIf your box dies, it's dead.  That's it.
22:45.05[TK]D-Fenderfprior: NOW is the point where telephony becomes "faith based", because you'd better PRAY it doesn't go down, but prayer is really all you've got left :P
22:45.40[TK]D-Fenders/but/because/
22:50.33*** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com)
22:50.35fprioryeah, I understand .....are you faithful or do you have implemented any solution of failover/HA ? Which is the scenario most used in * environment: HW virtualization or cluster ?
22:52.38*** join/#asterisk zerohalo (~zerohalo@74.60.136.128)
22:54.54*** join/#asterisk fisted (~fisted@unaffiliated/fisted)
22:55.31Netgeeksif you don't have any telephony hardware involved, i.e. pri cards, etc., your options are many
22:56.41Netgeekssrv records and secondary registration/proxie servers in SIP UA configs will get you quite far,  you can get even farther with a little more effort....
22:56.52*** join/#asterisk master_of_master (~master_of@p57B540F4.dip.t-dialin.net)
22:58.43NetgeeksI've not played with VMWare's cluster function in a while, and back when I was playing with it, asterisk didn't get along vmware in general too well.  I've heard that has changed, and I've seen some asterisk systems on vmware platforms running with no issue.
23:02.20fpriorNetgeeks: thanks. what about emergency: manuals don't explain how to manage this. do you give root password to IT Admin of your customer or they must waiting for you ?
23:04.23Netgeeksmy experience with giving root password access to customers is not a pleasant one.  Most treat it with the respect that they should, but some.....
23:05.32NetgeeksIf it were me, I would try and design it so that a single failure doesn't create an outage, and that gives you leeway to fix the failure and bring the until back up as a backup
23:09.42*** join/#asterisk WebSprocket (~user@dsl82-163-49-147.as15444.net)
23:10.38*** join/#asterisk jpsharp (~jsharp@74-95-145-82-Naples.hfc.comcastbusiness.net)
23:10.38WebSprocketHey guys, just needing a little advise, i have a sangoma card attached to a POTS line how in asterisk do i make our voice louder for the person we are calling,
23:10.50WebSprocketWe can hear them fine, but they cannot hear us.
23:11.11fpriorNetgeeks: yes, but depends on which is failure. Isn't the same an HW failure without remote access than a problem with Dialplan.
23:13.37jpsharpWebSprocket: Increase TX gain in your config file?
23:13.50Netgeeksfprior: well, a problem with the dialplan shouldn't be an emergency, that kind of problem should be debugged before you put the system into production.  The kind of issues you need to deal with in production will be failures that cause the underlying hardware/os environment to 'go away' and network failures.
23:14.46WebSprocket•jpsharp• Thanks was confirm it was TX not RX, what is an advised value to start off with to see if it helps.
23:16.00jpsharp6 is a good start.
23:16.17jpsharpI believe that will apply 6db of gain.
23:17.09fpriorNetgeeks: I'm so sorry, I need to go. Tomorrow we can continue the conversation, if you agree
23:17.56Netgeeksfprior: unfortunately I don't often monitor this channel fprior, I just happened to have it open today while working on something, I'll try and have it open tomorrow
23:18.08NetgeeksYou might do better by asking this question on the mail list
23:19.20*** join/#asterisk cbwest (~cbwest@nat/cisco/x-xjaapnzolluuodaz)
23:19.51fpriorNetgeeks: I will do this. thanks
23:30.00*** join/#asterisk grantm (~grant@68.142.138.4)
23:30.10*** join/#asterisk _Corey_ (~chatzilla@mail1.infradapt.com)
23:50.43*** part/#asterisk pietro (~pietro@88-149-227-4.dynamic.ngi.it)
23:58.56*** join/#asterisk mducharme-work1 (~nothing@206.188.121.4)
23:59.29*** join/#asterisk nix8n82-phone (~AndChat@75-174-157-152.chyn.qwest.net)

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