| 00:09.26 | *** join/#asterisk heffer (~felix@fedora/heffer) |
| 00:09.31 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 00:10.20 | sawgood | exten => _NXXNXXXXXX,2,Dial(SIP/ITSP/${EXTEN}) |
| 00:10.41 | sawgood | with the above statement, how to do have a "1" added to the front of the 10-digits before going off to the ITSP? |
| 00:12.27 | *** join/#asterisk dewman (~dewman@68-188-190-218.dhcp.bycy.mi.charter.com) |
| 00:17.20 | sawgood | exten => _NXXNXXXXXX,2,Dial(SIP/ITSP/+1${EXTEN}) |
| 00:17.25 | sawgood | does that look correct? |
| 00:19.42 | *** join/#asterisk m4xx (~m4xx@c-76-19-95-158.hsd1.ct.comcast.net) |
| 00:19.49 | dewman | hi there. Today i downloaded the asterisknow.iso. Did the install and then (i am guessing this is correct) ran yum update, however it comes back and complains about missing a kernel dependency. |
| 00:20.13 | m4xx | for some reason random incoming calls get rejected |
| 00:20.25 | m4xx | saying the extension cant be found |
| 00:20.38 | m4xx | yet my extension targets _X. |
| 00:20.43 | sawgood | m4xx: are you using Asterisk? |
| 00:20.45 | m4xx | yes |
| 00:20.57 | artista_frustrad | dwayne, thanks.. I had already done that and did not work.. after you mentioned I tried again and it worked |
| 00:21.00 | m4xx | 1.6 |
| 00:21.04 | sawgood | can you pastebin your extensions.conf? |
| 00:21.09 | m4xx | sure |
| 00:22.05 | Wiretap | sawgood, 1+ |
| 00:22.33 | sawgood | Wiretap: ty! |
| 00:22.36 | m4xx | sawgood http://paste2.org/p/1361898 |
| 00:24.45 | sawgood | I guess 1+ is different than +1 |
| 00:25.14 | Wiretap | sawgood, +1 = international dialprefix of 1, 1+ = prefix 1 |
| 00:25.59 | sawgood | So 1408+ is correct and +1408 is incorrect? |
| 00:26.48 | sawgood | assuming I wanted 1 + 408 added to the front of _NXXXXXX |
| 00:27.57 | *** join/#asterisk coppice (~chatzilla@62.166.232.220.dyn.pacific.net.hk) |
| 00:29.17 | m4xx | anything jump out at ya? |
| 00:32.02 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.81) |
| 00:33.13 | pabelanger | sawgood: you shouldn't need a + |
| 00:33.31 | pabelanger | actually, your ITSP will tell you what to send |
| 00:36.19 | sawgood | I need to add the 1 and or the 1+area code ... |
| 00:36.31 | sawgood | I fixed it by adding +1 |
| 00:36.34 | sawgood | it is working now |
| 00:36.55 | pabelanger | sawgood: Odd that your ITSP requires the + |
| 00:37.04 | sawgood | They do not require the + |
| 00:37.12 | sawgood | they want 11 digits on all calls |
| 00:37.16 | sawgood | I was sending only 10 |
| 00:47.11 | *** join/#asterisk pushpop (~pushpop@pool-173-77-230-33.nycmny.fios.verizon.net) |
| 01:07.24 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 01:09.08 | *** join/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 01:15.57 | *** join/#asterisk coppice (~chatzilla@m121-202-80-240.smartone-vodafone.com) |
| 01:17.09 | *** join/#asterisk magicblaze007 (~y@67.237.112.224) |
| 01:41.36 | *** join/#asterisk teathsch (~chatzilla@108-73-146-32.lightspeed.irvnca.sbcglobal.net) |
| 01:45.39 | jaytee | (crickets chirping) |
| 01:47.45 | coppice | we tried crickets chirping as an idle sound for a conference, but they chirped at about 4.2kHz, and nothing came out in the conference unless you connected with a wideband codec :-\ |
| 02:02.48 | *** join/#asterisk Kumbang (~kumbang@180.245.137.5) |
| 02:13.24 | *** join/#asterisk DNK0 (~DNK@189-19-113-208.dsl.telesp.net.br) |
| 02:14.12 | DNK0 | please what is new in asterisk realtime architecture 1.8 ? |
| 02:20.07 | *** join/#asterisk Bryanstein (~Bryanstei@shellium/admin/bryanstein) |
| 02:44.46 | Juggie | is it me or is downloads.asterisk.org slow? |
| 02:52.40 | kuku | Any hints on how to solve this issue are much appreciated: http://pastebin.com/EntSitg9 |
| 02:53.35 | Juggie | is this asterisknow? |
| 02:54.54 | kuku | I have no clue - just got into this system not too long ago - it doesnt seem like it. |
| 02:56.16 | Juggie | cat /etc/redhat-release |
| 02:56.26 | kuku | 5.6 |
| 02:57.03 | kuku | It seems I'm running 2.6.18-238 but these kmod-dahdi packages want 2.6.18_194 |
| 02:57.49 | kuku | i just did a big yum update, I'll reboot |
| 03:01.07 | Juggie | ya mismatched dahdi and kernel it seems |
| 03:01.24 | kuku | I changed the kernel back to 2.6.18_194 - lets see what hhappens |
| 03:04.54 | kuku | How do I check if dahdi is ok on this vm ? |
| 03:20.41 | *** join/#asterisk ajkaanbal (~ajkaanbal@189.181.84.227) |
| 03:34.08 | kuku | <PROTECTED> |
| 03:38.36 | Juggie | no |
| 03:46.11 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 03:56.30 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-ccarnnhdguvxxujb) |
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| 04:02.46 | kuku | For each secund, the timer should give 100 interrupts, correct? |
| 04:03.53 | kuku | dahdi show status shows no devices, is that bad ? |
| 04:04.59 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 04:08.32 | psilikon | coppice, audacity, drop it down a kilohert or two |
| 04:09.35 | psilikon | kuku, what type of tdm device? |
| 04:09.48 | DNK0 | please what is new in asterisk realtime architecture 1.8 ? |
| 04:09.54 | coppice | psilikon: it didn't sound realistic when we tried manipulating it |
| 04:10.09 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 04:10.11 | kuku | psilikon: no tdm device. |
| 04:10.33 | psilikon | coppice, right on. Makes sense. |
| 04:11.36 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
| 04:12.25 | coppice | psilikon: interestingly, the recording we used was about 30 minutes long, and the crickets kept dead on the same pitch for the whole duration |
| 04:15.47 | psilikon | coppice, that is an interesting foray into entomology |
| 04:16.27 | psilikon | coppice, was the population of crickets primarily male? |
| 04:18.05 | coppice | I wouldn't even know how to sex a cricket. the population of people who play cricket is certainly mostly male |
| 04:20.32 | psilikon | coppice, sexing a cricket is easy. They are so small that they can offer little resistance. |
| 04:20.57 | *** join/#asterisk Ean (~Ean@unaffiliated/ean) |
| 04:27.50 | *** join/#asterisk waterfoul (~chatzilla@67.129.121.92) |
| 04:34.50 | waterfoul | i'm getting a ton of errors about missing xmldocs how do i fix this? |
| 04:41.22 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 04:46.01 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
| 04:47.35 | pabelanger | waterfoul: install libxml2-dev and recompile asterisk |
| 04:52.10 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 05:09.51 | *** join/#asterisk tahiralmas (~tahir@182.177.182.77) |
| 05:13.08 | tahiralmas | hi all |
| 05:13.40 | tahiralmas | I am looking the way to detect 183 & 200 sip message and process these message in mysql database |
| 05:13.47 | tahiralmas | can any body help ? |
| 05:28.23 | *** join/#asterisk reber (~reber@cl-157.dub-01.ie.sixxs.net) |
| 05:29.24 | *** part/#asterisk tahiralmas (~tahir@182.177.182.77) |
| 05:31.54 | *** join/#asterisk chazzam (~chazz@173-24-239-247.client.mchsi.com) |
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| 06:23.52 | *** join/#asterisk SirSquidness (~sirsquidn@zomg.dongues.com) |
| 06:24.01 | *** join/#asterisk EugeneKay (EugeneKay@jedediahsmith.kashpureff.com) |
| 06:24.15 | EugeneKay | Anybody happen to be up at this ungodly hour? |
| 06:28.24 | coppice | ungodly hour? some fo us just finished lunch |
| 06:29.14 | EugeneKay | Anytime before noon is bad for me. |
| 06:29.59 | EugeneKay | To get to the point, I'm investigating a SIP-based phone system for my small business. I've found a lot of interesting stuff on commercial packages, but I am a sysadmin by trade, and rolling my own appeals to me far more. |
| 06:30.33 | EugeneKay | I just wanted a bit of confirmation that Asterisk is the "right tree" to bark up - everythign I've read thus far says it is |
| 06:32.18 | Sertys | well |
| 06:32.22 | Sertys | of course it is |
| 06:32.27 | EugeneKay | Heh |
| 06:32.35 | EugeneKay | I'd like to run it as a service on one of my colo'ed servers, accepting dial-ins to a number(via SIP), and then presenting the classic "Press 1 for ____, 2 for ____, 0 for an operator" menu. |
| 06:32.46 | *** join/#asterisk mac-mini (~mac-mini@unaffiliated/macmini/x-648924) |
| 06:32.49 | Sertys | na |
| 06:33.03 | Sertys | it supports only "press 1 for ____" |
| 06:33.08 | Sertys | it does not go up to 2 |
| 06:33.24 | EugeneKay | Depending upon how the user progresses through the menu, it ends up ringing through to my cell / home(whichever I set to "active"), or what-have-you |
| 06:33.53 | coppice | its no good unless it goes to 11 |
| 06:34.46 | EugeneKay | Is this something that is feasible(without TOO much cursing) with Asterisk? |
| 06:35.03 | Sertys | EugeneKay: yes, it is |
| 06:35.14 | Sertys | it's a pretty easy setup |
| 06:35.28 | Sertys | once you get familiar with how asterisk works, u'll be able to set it up in no time |
| 06:35.41 | Sertys | and there's FreePBX and asteriskNOW |
| 06:35.47 | Sertys | which will make it even easier |
| 06:38.58 | EugeneKay | How is stuff like XMPP / web integration? Stuff like changing the destination(from a soft SIP client on my desktop to my cell #) of an Extension |
| 06:39.24 | *** part/#asterisk Cesare (~Adium@creati59.lnk.telstra.net) |
| 06:40.24 | EugeneKay | On a scale from 1 to pounding nails through my extremities |
| 06:44.50 | *** join/#asterisk imcdona (imcdona@2001:470:e8f1:1:2506:95b3:40c6:372f) |
| 06:45.37 | *** part/#asterisk SirSquidness (~sirsquidn@zomg.dongues.com) |
| 06:51.06 | *** join/#asterisk mpe (~mpe@212.45.120.202) |
| 06:52.04 | *** part/#asterisk mpe (~mpe@212.45.120.202) |
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| 07:00.36 | *** join/#asterisk gajini (~gajini@117.230.186.160) |
| 07:00.44 | gajini | Hi |
| 07:01.48 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 07:02.27 | gajini | I am installing asterisk 1.6 and i want to install G.729 codec . How to do that? |
| 07:04.19 | kaldemar | gajini: http://downloads.digium.com/pub/telephony/codec_g729/README |
| 07:05.41 | *** join/#asterisk weta (~aross@CPE485b390978ce-CM00222ddf42dd.cpe.net.cable.rogers.com) |
| 07:06.26 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:18f0:3231:ec0e:7e9) |
| 07:06.29 | gajini | kaldemar: Is there any open source G729 codecs is there for commercial use? |
| 07:08.06 | kaldemar | that is the legal option. |
| 07:11.27 | gajini | <PROTECTED> |
| 07:31.07 | *** join/#asterisk creativx (~creadurex@226.62-97-205.bkkb.no) |
| 07:38.37 | *** join/#asterisk jkroon (~jkroon@dsl-241-233-50.telkomadsl.co.za) |
| 07:44.54 | *** join/#asterisk Tim_Toady (~moi@79.103.8.172.dsl.dyn.forthnet.gr) |
| 08:07.45 | *** join/#asterisk schmidts (~schmidts@213.235.212.193) |
| 08:07.47 | schmidts | good morning |
| 08:08.02 | *** join/#asterisk davlefou (~david@41.225.9.81) |
| 08:11.44 | _justdave | I have a BRI line on an Asterisk 1.8 box... Asterisk shows it Up/Active and placing calls and receiving calls works fine. But dahdi_scan from the command line shows both spans in RED alarm. Since it actually works, that seems bogus... |
| 08:12.21 | _justdave | would I possibly have something messed up in dahdi/system.conf or something that's making it mis-detect the alarm state? |
| 08:13.04 | _justdave | pri intense debug is showing identity check pings going back and forth periodically, looks fairly normal to me. |
| 08:13.58 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 08:14.35 | _justdave | system.conf: http://pastebin.mozilla.org/1203939 |
| 08:34.39 | *** join/#asterisk stoffell (~stoffell@66.44-200-80.adsl-dyn.isp.belgacom.be) |
| 08:35.09 | *** join/#asterisk devil_evoxxx (~d3v1l@157.27.183.122) |
| 08:35.23 | devil_evoxxx | hi all |
| 08:38.14 | devil_evoxxx | i have bought 3 digium B410P cards ( 4 Bri ), i want to put 3 of this cards in the same pc. It is possible? |
| 08:42.02 | *** join/#asterisk Denial (Denial@drgi.co.uk) |
| 08:43.41 | jkroon | yes. |
| 08:44.07 | jkroon | i've put three of those along with a quad PRI in the same box once. worked quite well. |
| 08:45.17 | devil_evoxxx | Thankyou :) Are g1, g2, gx.. after DIAL function for selecting the first, second or third card? |
| 08:48.30 | *** join/#asterisk Dovid (Dovid@office.mypbxmanager.net) |
| 08:50.24 | devil_evoxxx | or is the group? |
| 08:52.26 | kaldemar | a group as defined in chan_dahdi.conf |
| 08:55.03 | devil_evoxxx | ok, but when i define a group can i select which card to use? Sorry, but i don't find any example similar to my config, if you have some example i appreciate |
| 08:55.42 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 09:01.25 | kaldemar | devil_evoxxx: you select channels in the group, and channel numbers are assigned for cards and spans in /etc/dahdi/system.conf |
| 09:02.32 | *** join/#asterisk aberrios (~aberrios@195.171.4.82) |
| 09:03.56 | *** join/#asterisk zamba (marius@flage.org) |
| 09:04.24 | zamba | i'm trying to set up a conference room, but when entering the room number followed by the # key, i get the following error in asterisk console: |
| 09:04.24 | zamba | 2011-04-14 11:02:41] WARNING[1144]: app_meetme.c:1097 build_conf: Unable to open pseudo device |
| 09:04.35 | zamba | and then the nice lady informs me that there's no such room |
| 09:05.55 | *** join/#asterisk carloimperia (~carloimpe@109.112.58.150) |
| 09:07.40 | *** join/#asterisk florz (nobody@2001:1a50:503c::1) |
| 09:08.05 | zamba | i'm running ubuntu, and i'd rather not compile up a bunch of stuff manually |
| 09:09.11 | kaldemar | isntall dahdi |
| 09:09.31 | zamba | already have it installed |
| 09:09.41 | *** join/#asterisk tahiralmas (~tahir@182.177.182.77) |
| 09:09.51 | jkroon | zamba, is it loaded? |
| 09:09.52 | teathsch | it's kinda not frustrating coding on zolpidem.. actually makes it easier cuz i'm so chilled out |
| 09:09.59 | zamba | jkroon: nope |
| 09:10.06 | jkroon | then get it loaded. |
| 09:10.10 | zamba | jkroon: as a kernel module or as a module in asterisk? |
| 09:10.33 | jkroon | modprobe dahdi_dummy; then load chan_dahdi |
| 09:10.59 | zamba | FATAL: Module dahdi_dummy not found. |
| 09:11.16 | kaldemar | modprobe dahdi |
| 09:11.24 | zamba | same |
| 09:11.25 | zamba | not found |
| 09:11.38 | zamba | # dpkg -l | grep dahdi | wc -l |
| 09:11.38 | zamba | 3 |
| 09:11.50 | tahiralmas | hi all |
| 09:11.53 | zamba | dahdi, dahdi-linux and dahdi-dkms |
| 09:12.43 | zamba | trying a aptitude reinstall dahdi here |
| 09:12.46 | zamba | let's see what that does |
| 09:13.59 | tahiralmas | will anybody help me how to process sip message , like to find time delay of between 183 and 200 sip messages ? |
| 09:17.48 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 09:18.22 | *** join/#asterisk SiNGLer (~singler@81-7-123-162.static.zebra.lt) |
| 09:29.22 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 09:29.30 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 09:45.22 | Gugge | Can i change the timeout asterisk uses when dialing a device that is down (internet down or device turned off) ? |
| 09:46.12 | Gugge | When i do Dial(SIP/device,120) it takes about 30 seconds before giving up |
| 09:49.23 | *** join/#asterisk sekil (~sekil@80.93.247.26) |
| 09:52.38 | kaldemar | Gugge: what version are you using? |
| 09:52.39 | *** join/#asterisk coppice (~chatzilla@m180-219-195-152.smartone-vodafone.com) |
| 09:53.14 | Gugge | kaldemar: 1.6.2.11 |
| 09:54.07 | kaldemar | Gugge: see "SIP Timers" in http://svn.digium.com/svn/asterisk/tags/1.6.2.11/configs/sip.conf.sample |
| 09:54.56 | Gugge | timerb seems like the thing im looking for :) |
| 09:55.25 | *** join/#asterisk DarkRift (~dark@modemcable233.53-81-70.mc.videotron.ca) |
| 09:55.32 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
| 09:55.40 | *** join/#asterisk dandre (~daniel@ble59-2-81-56-122-47.fbx.proxad.net) |
| 10:02.19 | *** part/#asterisk sekil (~sekil@80.93.247.26) |
| 10:03.36 | Gugge | but timerb does not work in 1.6.2.11 :P guess ill have to upgrade :) |
| 10:03.42 | *** join/#asterisk DarkRift (~dark@modemcable233.53-81-70.mc.videotron.ca) |
| 10:03.45 | Gugge | https://issues.asterisk.org/view.php?id=16643 |
| 10:04.21 | zamba | how do i get icecast streams as music on hold? |
| 10:04.25 | zamba | it's an mp3 stream |
| 10:07.01 | *** join/#asterisk wonderworld (~ww@port-92-201-61-7.dynamic.qsc.de) |
| 10:48.36 | *** join/#asterisk nickfennell (~nick@cov1.appliansys.com) |
| 10:48.38 | nickfennell | hihihihih |
| 10:48.45 | nickfennell | Suggestions for a decent distro of Asterisk |
| 10:49.03 | nickfennell | need something with a GUI, reporting and management trimmings etc |
| 10:49.11 | nickfennell | Asterisk@Home still good? |
| 10:52.44 | Tim_Toady | nickfennell its caller trixbox now, but i would suggest asterisknow or elastix |
| 10:52.52 | Tim_Toady | called* |
| 10:54.17 | nickfennell | elastix is something I've seen recently |
| 10:54.22 | nickfennell | is that a full distro install? |
| 10:54.30 | JerJer | tar zxf asterisk-current.tgz && ./configure && make menuelect :) |
| 10:54.37 | nickfennell | lol |
| 10:55.12 | JerJer | +s |
| 10:55.23 | Tim_Toady | nickfennell its based on centos, so yes its a 'full distro' |
| 10:55.47 | nickfennell | Lovely |
| 10:56.06 | nickfennell | I'm feeling exceptionally lazy today so having to install an OS then Asterisk just doesn't seem like fun |
| 10:57.14 | nickfennell | Elastix will do |
| 10:57.28 | nickfennell | I have a AsteriskNow in this office somewhere |
| 11:05.06 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 11:09.52 | nickfennell | Is RAID good for asterisk ? |
| 11:10.02 | nickfennell | Was going to RAID1 two SATA disks |
| 11:17.53 | *** join/#asterisk radic (~radic@dslb-178-002-227-210.pools.arcor-ip.net) |
| 11:34.52 | c0rnoTa | Hello all |
| 11:35.24 | c0rnoTa | can anyone tell me, how DTMF detection works on PRI channels? |
| 11:36.05 | c0rnoTa | My trouble is that my customer press rapidly "7196", but system only receives "79" |
| 11:38.47 | c0rnoTa | i have already looked in log file. I see messages from chan_dahdi, that it capturing DTMF digit, but where second digit and 4th ? |
| 11:39.16 | c0rnoTa | relaxdtmf does not helped |
| 11:40.12 | c0rnoTa | i'm using asterisk 1.4.40, chan_dahdi 2.4.0 and libpri 1.4.12 beta3 on x68 gentoo. all compiled from sources. |
| 11:45.01 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 11:47.54 | *** join/#asterisk jacc0 (~jacco@D522448D.static.ziggozakelijk.nl) |
| 11:48.50 | jacc0 | any clue when 1.8.4 is going to be released? |
| 11:49.48 | jacc0 | I thikn it fixes some of the crashes,memoryleaks and deadlocks I'm experiencing in 1.8.3.2 |
| 11:50.39 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 11:50.57 | jacc0 | I'm experiencing great instability with 1.8.3.2 |
| 11:51.30 | mocker | jacc0: s/great/horrible? :) |
| 11:51.40 | jacc0 | horrible |
| 11:51.52 | jacc0 | losing customers over it |
| 11:52.18 | jacc0 | by the dozen |
| 11:52.45 | jacc0 | astcanary, safe_asterisk and monit have problems keeping things running :p |
| 11:53.13 | jacc0 | segfaults flooding the disk |
| 11:53.27 | mocker | jacc0: Is your error on the bugtracker? |
| 11:53.45 | jacc0 | right now I'm implementing fail-over asterisk servers at 4 sites |
| 11:54.19 | jacc0 | it's all very unclear where the crashes, segfaults, deadlocks and memoryleaks are coming from |
| 11:54.33 | jacc0 | becasue they dont happen in my dev. enviremnt |
| 11:55.13 | jacc0 | and I don't have asterisk compiled without optimization in the productioni envirements that are having problems |
| 11:55.34 | jacc0 | so it is almost imposible to trace |
| 11:56.17 | *** join/#asterisk freeedrich| (~eeePC@hansaserver.de) |
| 11:56.22 | jacc0 | I've moved /tmp to different partitions to stop core dumps from taking down the whole system |
| 11:56.53 | jacc0 | afther that the system still crashed; mysql needed some free space in /tmp |
| 11:57.10 | jacc0 | I'm sleeping bad lately |
| 11:57.13 | jacc0 | :S |
| 11:57.27 | jacc0 | and it's the main reson I'm hanging around here |
| 11:57.31 | jacc0 | for the last weeks |
| 11:58.24 | jacc0 | I guess some inter op. problems with some PBX's used in the production env. |
| 11:58.46 | jacc0 | and memory leaks, dealocks and coredumps that only accure over time |
| 11:58.54 | jacc0 | and/or with heavy load |
| 11:59.31 | jacc0 | I'm building a pure Asterisk alarm central |
| 11:59.50 | jacc0 | I'm using almost every function in asterisk |
| 11:59.56 | jacc0 | besides voicemail |
| 12:00.02 | *** join/#asterisk TobSnyder (~schneider@dslb-088-073-180-175.pools.arcor-ip.net) |
| 12:00.50 | jacc0 | and I have been ideling in the sangoma support channel every working day |
| 12:01.05 | jacc0 | for soma sangom problems |
| 12:01.08 | jacc0 | :S |
| 12:01.26 | jacc0 | all of this makes my job en hell |
| 12:01.29 | jacc0 | :S |
| 12:01.41 | mocker | If you are losing customers sounds like maybe you should try 1.6 on a box and see how that runs |
| 12:02.02 | jacc0 | 1.6 is missing app_originate |
| 12:02.08 | jacc0 | and some more functions I'm using |
| 12:02.16 | *** join/#asterisk dbrotman (~dbrotman@static-209-195-165-34.consolidated.net) |
| 12:02.47 | jacc0 | I could go back to echoing text to callfiles but that is also not a sollid solution |
| 12:02.54 | kaldemar | 1.6.2 has app_originate |
| 12:02.59 | dbrotman | What's the point of FXO module accepting RJ11 rather than RJ14? since each module can only support one line and the system provides all the power? |
| 12:03.04 | dbrotman | am I missing something? |
| 12:03.55 | jacc0 | :) I see |
| 12:03.59 | jacc0 | Originate a call. New in Asterisk 1.6.2 |
| 12:04.24 | jacc0 | could be an option |
| 12:04.44 | *** join/#asterisk luckman212_ (~irc@pool-173-77-253-141.nycmny.fios.verizon.net) |
| 12:04.57 | jacc0 | 2 steps forward one step back still makes me move forward |
| 12:06.14 | jacc0 | but right no I'm working on the fail-over option and I'm compiling fail-over server without otimization so that I can do some backtracing |
| 12:06.16 | kaldemar | there is no such branch as 1.6, but 1.6.0, 1.6.1 and 1.6.2 which are all different. |
| 12:07.02 | jacc0 | thank for one more option to consider kaldemar |
| 12:08.43 | jacc0 | *thanks |
| 12:16.57 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 12:18.06 | leifmadsen | kaldemar: +1 |
| 12:18.18 | leifmadsen | they are all very different |
| 12:19.06 | leifmadsen | you shouldn't be deploying a new asterisk system on any 1.6.x branch earlier than 1.6.2 (they have no support) |
| 12:19.28 | leifmadsen | 1.4 and 1.6.2 bug support stops on April 21 this year (then one more year of security support) |
| 12:20.52 | *** join/#asterisk ccesario (~ccesario@189-29-59-116-ac.cpe.vivax.com.br) |
| 12:21.43 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 12:26.09 | *** join/#asterisk jonnysupersonic (~IceChat77@dsl-241-205-216.telkomadsl.co.za) |
| 12:26.28 | jonnysupersonic | Hi All. Please can i ask a quick question about SIP Options in Asterisk 1.4? |
| 12:27.57 | *** join/#asterisk wonderworld (~ww@92.201.61.7) |
| 12:28.42 | Chainsaw | jonnysupersonic: Bit old, but... sure. |
| 12:28.50 | jacc0 | thanks for the X-tra info leifmadsen |
| 12:28.58 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 12:31.14 | jonnysupersonic | i have set a SIP peer in sip.conf. I set qualify=yes, and the host to a static IP address. The peer is an ACME SBC. When the ACME sends SIP OPTIONS, Asterisk is responding 404 not found, instead of OK |
| 12:31.51 | jonnysupersonic | so options are going in both directions. but some asterisk servers are rejecting 404 to options. cant figure this out |
| 12:32.14 | *** part/#asterisk dewman (~dewman@68-188-190-218.dhcp.bycy.mi.charter.com) |
| 12:32.45 | leifmadsen | jonnysupersonic: yes that is normal -- you need to enable the peer name in the default context |
| 12:32.52 | *** join/#asterisk cerberus_za (~coert@41-134-110-162.dsl.mweb.co.za) |
| 12:32.53 | leifmadsen | exten => 0004f2040001,1,NoOp() |
| 12:32.55 | leifmadsen | for example |
| 12:33.17 | leifmadsen | look at the SIP trace -- you'll see where it is trying to find in the dialplan the peer name |
| 12:33.26 | leifmadsen | it'll only respond OK if it finds it |
| 12:36.02 | jonnysupersonic | so is this because i set context=nowhere |
| 12:36.10 | jonnysupersonic | because i want to block incoming calls |
| 12:36.44 | leifmadsen | jonnysupersonic: no, my response still applies: <leifmadsen> jonnysupersonic: yes that is normal -- you need to enable the peer name in the default context |
| 12:37.02 | leifmadsen | the default context is whatever you set the default context to be |
| 12:37.10 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 12:37.17 | jonnysupersonic | ok i unerstand. how do you enable it? |
| 12:37.25 | *** join/#asterisk gruvfunk (~gruvfunk@user-160uac8.cable.mindspring.com) |
| 12:39.02 | gruvfunk | Hello, I'm starting a new Asterisk project, this time in Amazon Web Services - does anyone have any advice on which AMI and AKI to use? (I prefer Ubuntu and vanilla Asterisk 1.8) |
| 12:40.09 | Chainsaw | gruvfunk: We wrote our own in PHP. But perhaps others have tried some boxed solutions. |
| 12:40.17 | jonnysupersonic | @leifmadsen: must i create a noop with the same exension name as the peer name within the default context? |
| 12:41.14 | gruvfunk | @Chainsaw: Gentoo? |
| 12:41.36 | Chainsaw | gruvfunk: No, my employer. |
| 12:41.50 | Chainsaw | gruvfunk: I'm not aware of Gentoo using a telephone system. (I am the Gentoo maintainer for Asterisk though, yes) |
| 12:46.45 | leifmadsen | jonnysupersonic: please read what I said |
| 12:49.39 | dbrotman | \clear |
| 12:49.43 | dbrotman | sorry |
| 12:51.11 | *** join/#asterisk chuckf (~chuckf@ubuntu/member/chuckf) |
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| 12:58.45 | jonnysupersonic | yes i read it. but i have about 40 servers in our network. i have never added a peer in the default context. |
| 12:58.54 | jonnysupersonic | like your syntax of a NoOp |
| 13:00.55 | leifmadsen | that's what is required for asterisk to respond with 200 OK instead of 404 Not Found. Sounds like you'll need to create an #exec script for each server. |
| 13:01.23 | jacc0 | sometimes i get this error when bridging from dialplan: http://pastebin.com/de8ZPRPq |
| 13:01.31 | jacc0 | why? |
| 13:01.39 | jonnysupersonic | thanks for your help so far. i am guessing that the asterisk peer name and the ACME peer name should be the same. they are not right now |
| 13:02.29 | *** join/#asterisk ks3 (~ksandy@74.203.195.1) |
| 13:03.52 | Dovid | anyone here using voipmonitor ? |
| 13:05.06 | *** join/#asterisk fofware (~Fabian@host120.186-109-187.telecom.net.ar) |
| 13:05.57 | fofware | hello all |
| 13:06.42 | Dovid | hi |
| 13:09.25 | *** join/#asterisk kamikazemicrowav (~kamikazem@seraph.contegix.com) |
| 13:09.35 | *** join/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 13:10.16 | fofware | I have a TDM410p with 2 fxo and 2 fxs ports, It was working untile I try webgui and something was wrong because after that never more work my TDM410p, I did reintall all, Linux, asteterisk and dahdi and my issues presist... Asterisk 1.8.3.2 and last dahdi from asterisk page. |
| 13:11.19 | fofware | each time I try to call some number dahdi hangup |
| 13:11.21 | fofware | 17:17 UTC |
| 13:11.22 | fofware | <PROTECTED> |
| 13:11.22 | fofware | <PROTECTED> |
| 13:11.22 | fofware | <PROTECTED> |
| 13:11.46 | fofware | any one can help me with that? |
| 13:12.12 | aberrios | Anyone recommend a UK datacentre for virtual dedicated servers? |
| 13:15.11 | Chainsaw | aberrios: Positive Park. http://www.positive-internet.com/ |
| 13:16.52 | aberrios | Chainsaw: Aww look how happy Mr Fry looks |
| 13:17.40 | *** join/#asterisk tasca (~tasca@189.73.88.227) |
| 13:18.34 | *** join/#asterisk Karen_m (~karen@66.222.153.231) |
| 13:18.41 | Karen_m | does asterisk take much load ? |
| 13:18.57 | Karen_m | I'm thinking of installing asterisk on my server but it hosts websites and what not lol |
| 13:19.17 | aberrios | Karen_m: now that's a broad Question. |
| 13:19.37 | *** part/#asterisk titter (~Justin@c-98-208-153-148.hsd1.fl.comcast.net) |
| 13:19.52 | leifmadsen | Karen_m: it depends on what kind of server it is, and what asterisk is doing, and how many simultaneous channels it is handling |
| 13:19.59 | leifmadsen | per channel, asterisk doesn't use that much CPU |
| 13:20.13 | leifmadsen | assuming your server is relatively new, and isn't maxed out already |
| 13:20.18 | *** join/#asterisk serafie (~erin@nat/digium/x-ktjhwljfepdnjyyj) |
| 13:20.27 | leifmadsen | but if it overwhelms your server, it's your own fault :) |
| 13:20.35 | leifmadsen | you should use a development server to test first |
| 13:21.02 | jacc0 | __ast_pthread_mutex_unlock: features.c line 5863 (bridge_exec): mutex 'current_dest_chan' freed more times than we've locked! |
| 13:21.02 | jacc0 | [Apr 14 14:59:40] ERROR[12837]: lock.c:416 __ast_pthread_mutex_unlock: features.c line 5863 (bridge_exec): Error releasing mutex: Operation not permitted |
| 13:21.28 | jacc0 | what does that meam? a bug in asterisk? or am I doing somethin wrong? |
| 13:22.31 | *** join/#asterisk Defraz (~Defraz@184-155-137-27.cpe.cableone.net) |
| 13:23.08 | Karen_m | leifmadsen, the server has 8GB of ram, 1Gbps connection and load is always like 0.3~ I guess I will try it :) |
| 13:23.16 | leifmadsen | have fun |
| 13:24.47 | jonnysupersonic | @leifmadsen - the default context is default. Inside [default] i added an extension with the name of the SIP peer in sip.conf. I still get 404 not found |
| 13:30.42 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
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| 13:35.57 | mocker | Karen_m: If it's a web server on the Internet you will want to watch your security as well. |
| 13:36.04 | mocker | toll fraud is a bitch. |
| 13:38.57 | *** join/#asterisk Tim_Toady (~moi@79.103.8.172) |
| 13:43.06 | *** join/#asterisk anonymouz666 (~anonymouz@187-28-37-118.poolip.RJO.embratel.net.br) |
| 13:44.30 | jonnysupersonic | @leifmadsen are you avail to help me? |
| 13:44.41 | Karen_m | what do you mean toll fraud ? |
| 13:45.06 | JonathanRose | People use your Asterisk server to make phone calls which may add to your phone bill |
| 13:45.38 | *** join/#asterisk Freeaqingme| (~dolf@dsl-083-247-011-232.solcon.nl) |
| 13:46.17 | JonathanRose | http://www.asteriskguide.com/mediawiki/index.php/Ten_security_tips_to_avoid_toll_fraud |
| 13:46.19 | *** part/#asterisk tasca (~tasca@189.73.88.227) |
| 13:47.15 | *** join/#asterisk nite613 (~chris@CPE001839c16d35-CM00237453c586.cpe.net.cable.rogers.com) |
| 13:49.45 | nite613 | Hi there. * 1.4.24, having problems with transmit_silence and transmit_silence_during_record, namely that rtp packets are not transmitted during "silence". We are recording audio via an AGI and our provider is hanging up after 300s of "no RTP media" |
| 13:53.37 | *** join/#asterisk bchia (~chatzilla@nat/digium/x-dllugbfnoycwszuc) |
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| 13:59.11 | jonnysupersonic | Hi All. i have a problem with asterisk replying 404 to OPTIONS messages. anyone know how to fix this? |
| 14:03.12 | anonymouz666 | jonnysupersonic: you need a valid extension |
| 14:03.28 | anonymouz666 | s,1,NoOp on default context, for example. |
| 14:03.47 | jonnysupersonic | thanks for the reply. so in sip.conf that peer points to context default |
| 14:03.55 | jonnysupersonic | so in default i need a s,1,NoOp? |
| 14:03.59 | jonnysupersonic | thats all it is? |
| 14:04.14 | anonymouz666 | try it |
| 14:04.45 | *** join/#asterisk joshaidan (~brianj@S0106001c1023e838.tb.shawcable.net) |
| 14:05.46 | jonnysupersonic | ok thanks. trying it now |
| 14:06.13 | jonnysupersonic | i thought the extension in that context must be exten => peername,1,NoOp |
| 14:09.53 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
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| 14:27.12 | nite613 | Hi there. * 1.4.24, having problems with transmit_silence and transmit_silence_during_record, namely that rtp packets are not transmitted during "silence". We are recording audio via an AGI and our provider is hanging up after 300s of "no RTP media". What other option do I need to twiddle in order for transmit_silence to work? |
| 14:28.04 | jonnysupersonic | anonymouz666 thanks for your assistance. what a simple fix |
| 14:28.11 | *** join/#asterisk Aut0ExeC (~Jack@24.244.156.75) |
| 14:28.55 | kuku | lk |
| 14:29.28 | Aut0ExeC | hi guys |
| 14:29.33 | anonymouz666 | autoexec.bat |
| 14:29.51 | Aut0ExeC | :) |
| 14:29.53 | jacc0 | lol |
| 14:30.04 | anonymouz666 | config.sys |
| 14:30.25 | jacc0 | win.ini |
| 14:30.39 | Aut0ExeC | ok ok windows lovers... enough |
| 14:30.45 | Aut0ExeC | :) |
| 14:30.48 | joshaidan | Was it f8 or f5 you pushed to get into mode where you could choose what to execute? :) |
| 14:30.50 | anonymouz666 | I remember that I needed to comment some lines on config.sys in MS-DOS 5.0 to run some games |
| 14:31.18 | Aut0ExeC | anonymouz666: msdos was the bomb back then man |
| 14:31.26 | Aut0ExeC | 6.22 |
| 14:31.43 | Aut0ExeC | loved doom and duke nukem most |
| 14:31.56 | anonymouz666 | we all did |
| 14:31.59 | Aut0ExeC | yeah |
| 14:32.08 | Aut0ExeC | i miss those days...:( |
| 14:33.04 | *** join/#asterisk jkroon (~jkroon@dsl-241-233-50.telkomadsl.co.za) |
| 14:33.26 | jacc0 | i relay love CAT on my CGA monitor |
| 14:33.31 | jacc0 | :p |
| 14:34.45 | jacc0 | I guess the degree of b#llsh!t tells me I need to go home |
| 14:34.50 | jacc0 | :p |
| 14:34.57 | jacc0 | I'm calling it a day |
| 14:35.10 | jacc0 | ttyl all!! |
| 14:35.26 | Aut0ExeC | jacc0: later |
| 14:42.28 | *** join/#asterisk tuxx- (tuxx@vps460.directvps.nl) |
| 14:42.30 | tuxx- | plop |
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| 15:33.31 | *** join/#asterisk m_tadeu (~quassel@89-180-193-162.net.novis.pt) |
| 15:34.09 | m_tadeu | hi |
| 15:34.39 | m_tadeu | I'm getting lots of these warnings in the log "xmldoc.c: Couldn't find function QUEUE_MEMBER_PENALTY in XML documentation" |
| 15:59.16 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-180-175.pools.arcor-ip.net) |
| 16:07.00 | nite613 | anyone have success with transmit_silence in * 1.4? |
| 16:08.42 | Freeaqingme| | nite613, why would you want to transmit mere silence? #curious |
| 16:09.07 | nite613 | Good question! :) It's because the system accepts voice recordings, so it's just 1-way audio |
| 16:09.22 | nite613 | and our provider hangs up the call if they detect 300s of silence in either direction |
| 16:09.43 | nite613 | rtpkeepalive doesn't help as they don't accept comfort noise as rtp media |
| 16:10.10 | nite613 | I've turned on transmit_silence and transmit_silence_during_record, they don't seem to work |
| 16:10.44 | Freeaqingme| | in either direction? so a voicemail message could never be longer than 300s? |
| 16:10.56 | nite613 | Yes, that appears to be the case :( |
| 16:11.08 | Freeaqingme| | lol |
| 16:11.22 | nite613 | Part of me just wants to switch to a provider that doesn't have this restriction, but management is not as keen ;) |
| 16:13.47 | Freeaqingme| | nite613, I understand. I have no idea as to what a answer to your question would be though. Perhaps someone else does |
| 16:16.50 | EugeneKay | nite613 - send a beep once a minute? |
| 16:18.56 | nite613 | EugeneKay: Not entirely a bad idea, but there is already an option for this. Just trying to figure out if it's a bug in 1.4 or if I'm doing something wrong |
| 16:19.43 | EugeneKay | I haven't a clue about Asterisk, I'm just trying to be helpful :-p I would guess that it is all your provider's fault. |
| 16:20.51 | nite613 | Well at this point it's *'s fault since it fails to transmit silence when asked ;) But yes, we may have to implement a onece-a-minute beep as a workaround |
| 16:23.00 | *** join/#asterisk The_Boy_Wonder (~manbearpi@asterisk/batman-developer/dvossel) |
| 16:34.55 | *** join/#asterisk DrDamnit (~michael@highpoweredhelp.com) |
| 16:34.56 | DrDamnit | How dialplan logic would I use to call and extension, and then when that extension is picked up, transfer them into a meetme conference? |
| 16:35.56 | leifmadsen | DrDamnit: on 1.6.2 or later, use app_originate |
| 16:35.59 | leifmadsen | Originate(...) |
| 16:36.10 | leifmadsen | you can control what the end point is connected to on answer |
| 16:36.21 | DrDamnit | can I do that via agi? |
| 16:36.28 | DrDamnit | phpagi has an originate method... |
| 16:37.27 | psilikon | DrDamnit, I do something similiar will a python script |
| 16:37.58 | DrDamnit | psilikon: I must be making this harder than it is. Would you be willing to share the relevant portions of your python script? |
| 16:38.26 | psilikon | DrDamnit, yeah i'll paste it. |
| 16:38.32 | DrDamnit | awesome. Thank you. |
| 16:39.09 | psilikon | DrDamnit, http://pastebin.com/uDVG9tLh |
| 16:39.17 | m_tadeu | I'm getting lots of these warnings in the log "xmldoc.c: Couldn't find function QUEUE_MEMBER_PENALTY in XML documentation...what can I do about this? |
| 16:39.20 | *** join/#asterisk ddickenson (~ddickenso@67-198-0-5.static.grandenetworks.net) |
| 16:39.25 | psilikon | DrDamnit, I have the same thing as a perl script if you wanna see it too |
| 16:39.37 | *** join/#asterisk Jasnejac (kvirc@81.91.107.236) |
| 16:41.27 | psilikon | DrDamnit, you'll have to extract what you need from that. That script is designed to call two people and dump them in a conference. Once both parties are in the conference sound clips can be played to the conf. |
| 16:41.30 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v000-152.mobile.uci.edu) |
| 16:41.41 | DrDamnit | psilikon: The more examples, the better. :-) |
| 16:43.02 | leifmadsen | m_tadeu: sounds like something got mucked up in your branch because that documentation is automatically generated when you compile asterisk |
| 16:43.55 | leifmadsen | DrDamnit: read the Call Control chapter in the Asterisk Cookbook |
| 16:44.03 | leifmadsen | http://ofps.oreilly.com/titles/9781449303822/ |
| 16:44.08 | leifmadsen | ~asteriskcookbook |
| 16:44.13 | DrDamnit | leifmadsen: Thanks! On it. |
| 16:44.16 | m_tadeu | leifmadsen: I installed from the packages that digium supplied for ubuntu |
| 16:44.24 | leifmadsen | pabelanger: ^^^^^^ |
| 16:44.30 | leifmadsen | m_tadeu: sounds like a bug then |
| 16:44.43 | leifmadsen | m_tadeu: you'll need to open an issue at issues.asterisk.org |
| 16:44.57 | m_tadeu | leifmadsen: is there a way to disable this documentation? |
| 16:46.24 | leifmadsen | infobot: asteriskcookbook is reply The Asterisk Cookbook is available for purchase as an eBook from O'Reilly at http://oreilly.com/catalog/0636920018551 or via Amazon. The book is available under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and can be read online at http://ofps.oreilly.com/titles/9781449303822/ |
| 16:46.24 | infobot | leifmadsen: okay |
| 16:46.29 | leifmadsen | ~thebook |
| 16:46.29 | infobot | i guess thebook is Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342. |
| 16:47.26 | leifmadsen | infobot: no, thebook is reply Asterisk: 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://ofps.oreilly.com/titles/9780596517342/ |
| 16:47.26 | infobot | okay, leifmadsen |
| 16:47.31 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 16:47.32 | leifmadsen | ~newbook |
| 16:47.33 | infobot | Asterisk: The Definitive Guide. A preview of the upcoming release is available at http://ofps.oreilly.com/titles/9780596517342/. The book is being released under a Creative Commons license: http://creativecommons.org/licenses/by-nc-nd/3.0/us/. Please consider supporting the authors generosity by purchasing a copy at http://oreilly.com/catalog/9780596517342. |
| 16:47.49 | DrDamnit | leifmadsen & psilikon: Thank you both. Knowledge obtained, issue solved. |
| 16:47.58 | leifmadsen | infobot: no, newbook is reply Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 16:47.59 | infobot | okay, leifmadsen |
| 16:48.02 | leifmadsen | ~newbook |
| 16:48.02 | infobot | i guess newbook is reply Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 16:48.06 | leifmadsen | grrr |
| 16:49.00 | jaytee | arghh |
| 16:49.25 | leifmadsen | what is the format for telling infobot how to repond? |
| 16:49.27 | leifmadsen | respond* |
| 16:49.42 | leifmadsen | I thought it was: infobot: foo is reply <description> |
| 16:50.31 | tvc123 | looking at the output I would say it is infobot: foo is <description> |
| 16:50.51 | leifmadsen | yes, but that adds some things like, "i guess newbook is" at the beginning |
| 16:50.58 | leifmadsen | there is a way to make it skip that extra prefix stuff |
| 16:51.07 | leifmadsen | infobot: no, newbook reply Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 16:51.11 | leifmadsen | hmmmm |
| 16:51.42 | leifmadsen | infobot: no, newbook is <reply> Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 16:51.42 | infobot | leifmadsen: okay |
| 16:51.44 | leifmadsen | there we go |
| 16:51.58 | leifmadsen | infobot: no, thebook is <reply> Asterisk: 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://ofps.oreilly.com/titles/9780596517342/ |
| 16:51.58 | infobot | leifmadsen: okay |
| 16:52.13 | leifmadsen | infobot: no, asteriskcookbook is <reply> The Asterisk Cookbook is available for purchase as an eBook from O'Reilly at http://oreilly.com/catalog/0636920018551 or via Amazon. The book is available under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and can be read online at http://ofps.oreilly.com/titles/9781449303822/ |
| 16:52.13 | infobot | leifmadsen: okay |
| 16:52.16 | leifmadsen | much better |
| 16:52.19 | leifmadsen | ~newbook |
| 16:52.19 | infobot | Please see ~thebook for more information about Asterisk: The Definitive Guide |
| 16:52.24 | leifmadsen | ~thebook |
| 16:52.24 | infobot | Asterisk: 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://ofps.oreilly.com/titles/9780596517342/ |
| 16:52.27 | leifmadsen | bingo |
| 16:52.35 | tvc123 | nice! |
| 16:52.53 | leifmadsen | lunchtime! |
| 16:53.01 | tvc123 | thanks again for putting it online it has been really helpful I will be picking up a hard copy as well |
| 16:53.22 | jaytee | I want a Kindle edition soon! |
| 16:57.12 | *** join/#asterisk RickB17 (~RickB17@pat.recoverynetworks.com) |
| 16:57.38 | RickB17 | how can i have a dialplan logic check if a sip agent is registered? |
| 16:58.27 | DrDamnit | RickB17: AGENT AGENT(agentid[:item]) Gets information about an Agent |
| 16:58.41 | DrDamnit | core show function AGENT |
| 16:58.42 | RickB17 | sorry i didn't mean agent like queue agent |
| 16:58.48 | RickB17 | i meant like a sip client |
| 16:58.54 | RickB17 | a softphone or hardphone |
| 16:59.51 | RickB17 | I want to provide sip users the ability to register to either or two asterisk boxes, but I need dialplan logic to work with it. |
| 16:59.54 | RickB17 | any suggestions? |
| 17:00.09 | zamba | somewhere i can get the script for the different prompts? |
| 17:00.20 | zamba | i'm thinking about creating a new set of norwegian prompts |
| 17:01.29 | Aut0ExeC | awesome link to the definitive guide.. is there an ebook yet? |
| 17:03.29 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 17:04.55 | *** join/#asterisk citywok (~citywok@67-134-194-33.dia.static.qwest.net) |
| 17:08.13 | Aut0ExeC | how does one go about selling the idea of opensource pbx systems to companies that are used to cisco and avaya? |
| 17:09.01 | Freeaqingme| | Aut0ExeC, "it works + better + cheaper"? |
| 17:09.02 | citywok | tell them it works just as well but it's free, just costs to configure & install |
| 17:09.26 | Freeaqingme| | besides, why tell them it's opensource? |
| 17:09.37 | Freeaqingme| | you just say that you're licensed to sell them products developed by digium |
| 17:09.48 | citywok | Freeaqingme: legally i believe you're required to tell them, AND provide the source code :) |
| 17:10.01 | citywok | you can't sell open source software under the guise that you wrote it. |
| 17:10.15 | Freeaqingme| | citywok, you have to put somewhere that it's gpl'ed indeed |
| 17:10.18 | Aut0ExeC | lol Freeaqingme| licensed? |
| 17:10.27 | Freeaqingme| | Aut0ExeC, licensed by the gpl ;) |
| 17:10.33 | Aut0ExeC | hahaaha |
| 17:10.36 | Freeaqingme| | "you are allowed to distribute" |
| 17:10.37 | Aut0ExeC | funny bro |
| 17:10.41 | Aut0ExeC | ok |
| 17:10.41 | Freeaqingme| | I'm not joking |
| 17:10.50 | citywok | i just leave the source code directly on the pbx when i install it |
| 17:10.50 | Aut0ExeC | i guess thats a way of saying your licensed i guess |
| 17:11.01 | Freeaqingme| | you cant say you're a digium partner |
| 17:11.06 | Freeaqingme| | but you are licensed to distribute it |
| 17:11.16 | Aut0ExeC | i see |
| 17:11.34 | _Corey_ | you could also sign up to become a Digium partner.... :) |
| 17:11.42 | Aut0ExeC | nice |
| 17:11.44 | Freeaqingme| | and most companies simply want a solution to their problems, whether it's opensource or not wont bother them. As long as it works, and they can get the appropriate support (from you) for a nice price they're fine wit all of it |
| 17:11.52 | Aut0ExeC | citywok: u dont setup switchvox systems?> |
| 17:12.28 | Aut0ExeC | wouldnt switchvox be easier to sell? |
| 17:12.36 | Aut0ExeC | sell as in sell the idea |
| 17:12.53 | Aut0ExeC | vs cisco etc etc |
| 17:13.20 | Aut0ExeC | instead of this dusty box in a corner with asterisk and a digium card? |
| 17:13.22 | citywok | well, it's easier to sell b/c it's a bundled product |
| 17:13.25 | _Corey_ | Aut0Exec: Go to the Astricon website and play some of the presentation videos... there are a bunch of sessions on selling PBXs |
| 17:13.37 | Aut0ExeC | _Corey_: ok thanks |
| 17:13.46 | Freeaqingme| | Aut0ExeC, if the gpl is a problem for you, you can also get asterisk with a proprietary license. I suppose you need to pay for that, no idea how much |
| 17:14.16 | Aut0ExeC | ok |
| 17:14.34 | Aut0ExeC | companies are weird... something is free /opensource and they dont trust it |
| 17:14.38 | Aut0ExeC | they pay and they feel better |
| 17:14.54 | Aut0ExeC | I've seen this over and over |
| 17:15.18 | Freeaqingme| | Aut0ExeC, yeah, so just charge them for it |
| 17:15.28 | Aut0ExeC | Freeaqingme|: :) i know right |
| 17:15.34 | Aut0ExeC | Freeaqingme|: overcharge on the service lol |
| 17:15.45 | Aut0ExeC | to make up for the software |
| 17:16.01 | Freeaqingme| | you can charge for the software just as well |
| 17:16.09 | Freeaqingme| | though a monthly license may be more lucrative ;) |
| 17:16.17 | Aut0ExeC | k |
| 17:16.57 | Aut0ExeC | hey is there a way to tie rates to cdr records? for like billing purposes? |
| 17:17.22 | Aut0ExeC | like if there was a cdr record for example price_per_minute |
| 17:17.54 | Aut0ExeC | then you can use a reporting software to print off nice reports to present for billing |
| 17:18.05 | Freeaqingme| | _Corey_, tnx for the astricon suggestion, got some interesting vids on there |
| 17:19.39 | _Corey_ | good luck |
| 17:21.13 | *** join/#asterisk Arsenick (~y@fedora/Arsenick) |
| 17:24.03 | *** join/#asterisk SiNGLer (~singler@78-60-139-121.static.zebra.lt) |
| 17:24.28 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 17:33.45 | nite613 | For anyone reading the record, I figured out what transmit_silence isn't working for me. |
| 17:34.40 | Freeaqingme| | tell us :D |
| 17:34.50 | nite613 | Our application calls the internal Dictate() app in asterisk, which does not have support for generating silence. If all else fails with my provider, I will be patching app_dictate.c so that it does silence generation similar to how it is done in app_record.c |
| 17:35.19 | nite613 | If it comes to that I will seek approval from management to release my patch into the bug tracker |
| 17:36.03 | Freeaqingme| | hmm, what's the difference between _record and _dictate? |
| 17:36.14 | Freeaqingme| | the names suggest they do the same |
| 17:36.23 | nite613 | dictate has some extra DTMF features I think |
| 17:37.03 | Freeaqingme| | ah,kk |
| 17:46.41 | *** join/#asterisk ihabtawfig (~ihabtawfi@41.95.12.67) |
| 17:53.21 | ihabtawfig | Hi , I have a problem with asterisk crash using Dahdi with TE410P |
| 17:53.36 | ihabtawfig | AM I on the right place? |
| 17:54.23 | leifmadsen | ~debugging |
| 17:54.23 | infobot | if debugging is the process of removing bugs, then programming must be the process of putting them in. |
| 17:54.26 | fauxalliance | ihabtawfig, those are supported directly from Digium non? |
| 17:54.30 | leifmadsen | ~asteriskdebug |
| 17:55.03 | leifmadsen | infobot: asteriskdebug is <reply> Information about how to collect debug information is available at https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 17:55.03 | infobot | leifmadsen: okay |
| 17:56.02 | ihabtawfig | yes but I think my problem is with asterisk rather than the hardware |
| 17:56.24 | fauxalliance | ihabtawfig, IMHO, they are both hopelessly intertwined |
| 17:57.20 | fauxalliance | saunters off singing the theme tune for "Married with Children" |
| 17:58.43 | ihabtawfig | @leifmadsen: I have collected that Core backtrace, where shall I go from here? |
| 17:58.54 | leifmadsen | http://issues.asterisk.org |
| 18:00.38 | ihabtawfig | Not openning :( thanks any how, glad to talk to Madsen himself in my first entery here :) |
| 18:01.37 | pabelanger | ~collectdebug |
| 18:01.37 | infobot | rumour has it, collectdebug is a method of collecting logs allowing others help troubleshoot an issue. Refer to https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 18:01.40 | fauxalliance | ihabtawfig, read HIS book.. make MY day |
| 18:01.42 | pabelanger | leifmadsen: ^ |
| 18:02.04 | leifmadsen | pabelanger: aha that's what I wanted... wonder how I delete an entry |
| 18:02.13 | leifmadsen | ihabtawfig: https://issues.asterisk.org |
| 18:02.17 | leifmadsen | https, not http |
| 18:02.32 | pabelanger | infobot: forget asteriskdebug |
| 18:02.33 | infobot | pabelanger: i forgot asteriskdebug |
| 18:02.37 | pabelanger | ~asteriskdebug |
| 18:02.40 | ihabtawfig | fauxalliance : |
| 18:02.59 | ihabtawfig | yes, now he has asterisk coockbook , cant wait to get it |
| 18:03.22 | fauxalliance | shudders at the thought |
| 18:04.41 | ihabtawfig | @leifmadsen : ahhhaa , thaanx, i know I can get something from you :) |
| 18:04.44 | ihabtawfig | working now |
| 18:20.23 | *** join/#asterisk nix8n82 (~nate@24.143.27.157) |
| 18:26.41 | *** join/#asterisk Diffen (~diffen@c-fc73e555.042-17-73746f11.cust.bredbandsbolaget.se) |
| 18:27.03 | *** join/#asterisk SaiSoma (~chatzilla@client105.jdcc.edu) |
| 18:37.35 | *** join/#asterisk agroman (agroman@otaku.freeshell.org) |
| 18:38.51 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 18:39.48 | agroman | is chan_gtalk in 1.8.3.2 still working? I'm having some issues and have followed the instructions in the wiki exactly. |
| 18:41.21 | Aut0ExeC | agroman: I use it |
| 18:43.28 | Aut0ExeC | it rox basically... i love it |
| 18:44.02 | EugeneKay | Here's a question. How in the HECK do call-in numbers work? |
| 18:44.18 | EugeneKay | IE, a 1-800 SIP number. |
| 18:44.55 | EugeneKay | It's a concept I've not figured out yet |
| 18:45.24 | Freeaqingme| | EugeneKay, 1-800 are the numbers that by default are collect call, right? |
| 18:45.38 | EugeneKay | I mean, multiple calls in to the same number |
| 18:45.55 | EugeneKay | (yes, they are, but I'm ignoring billing for now) |
| 18:46.29 | Freeaqingme| | what's there to get about it? you just have a number that points to a certain pbx, and the pbx simply handles all the calls at once |
| 18:47.05 | EugeneKay | I'm still learning the basics of PBX |
| 18:47.30 | EugeneKay | And SIP, for that matter |
| 18:47.36 | Freeaqingme| | EugeneKay, it's just a like a website that can handle simultaneous visitors |
| 18:48.02 | EugeneKay | So there's no concept of a busy signal, it just routes the signal over a second SIP channel automagically |
| 18:48.17 | EugeneKay | s/signal/call/ |
| 18:48.29 | EugeneKay | No you stupid bot, the other "signal" |
| 18:48.52 | Freeaqingme| | hey, the bot is trying, gotta give t credits for that ;) |
| 18:49.00 | Freeaqingme| | EugeneKay, yes, correct |
| 18:49.15 | Freeaqingme| | the pbx may however choose to return a busy signal and exit the connection |
| 18:50.05 | EugeneKay | Except I'd be returning an auto-answer system and routing it to voicemail. |
| 18:50.18 | EugeneKay | Makes a bit more sense. |
| 18:50.48 | Freeaqingme| | EugeneKay, yes, of course |
| 18:50.58 | Freeaqingme| | but there's no concept of a busy signal with voicemails ;) |
| 18:51.57 | agroman | I'm getting this from chan_gtalk. I've checked over my jabber connection name and that it's referenced in my gtalk.conf. :-\ http://pastebin.com/6MgAndLq |
| 18:52.09 | agroman | dialing out... |
| 18:52.12 | EugeneKay | Well, for a soft-voicemail. I'm still at the stage of unlearnign concepts like an answering machine is what is attached to my phone |
| 18:53.31 | Freeaqingme| | EugeneKay, you may also want to read on ss7 if you want to know how telephony works in general |
| 18:54.10 | Freeaqingme| | http://en.wikipedia.org/wiki/Signaling_System_7 |
| 18:54.13 | EugeneKay | Care to link me to a good guide? I can make do with the Wikipedia article |
| 18:54.48 | Freeaqingme| | I think that link is a good one already |
| 18:55.10 | benngard | dont forget about q.sig ;) |
| 18:55.39 | Freeaqingme| | lol |
| 18:55.48 | benngard | :) |
| 18:55.56 | benngard | couldnt resist |
| 18:57.12 | Aut0ExeC | how do you tie billing into cdr? |
| 18:57.34 | Aut0ExeC | like to you create another record for it in cdr? |
| 18:57.58 | Aut0ExeC | if i wanted to bill on every call based on area code etc etc? |
| 19:06.22 | agroman | anyone around to answer some questions re: chan_gtalk? |
| 19:06.53 | Freeaqingme| | benngard, you could also tell him about the globally used ss5 |
| 19:09.26 | agroman | is it possible to register asterisk with OCS? I'm trying this: "register => tls://first.m.lastname@domain.com:PASSWORD:AD\username@sip.domain.com/from-ocs |
| 19:10.07 | agroman | but I'm getting timeouts (no response from sip.domain.com) |
| 19:10.49 | agroman | I'm trying to get all OCS calls routed through asterisk to my sip ata |
| 19:37.43 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 19:37.43 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.2 (2011/03/17), 1.6.2.17.2 (2011/03/17), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.1 (2011/04/01), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 19:37.51 | *** join/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 19:38.06 | *** part/#asterisk mbrevda (~mbrevda@unaffiliated/mbrevda) |
| 19:38.09 | gruvfunk | @agroman - that's fine.. it's there.. hmm |
| 19:41.44 | gruvfunk | @agroman: the only difference between your config and mine is 1 |
| 19:42.25 | gruvfunk | though I'm unsure if this is required, or legacy from previous configurations - in gtalk.conf [general], I have an externip= statement with my external IP address |
| 19:42.39 | gruvfunk | do you have 5222 UDP open on your firewall(s) ? |
| 19:43.07 | *** join/#asterisk saliak (~kailas@68.9.228.184) |
| 19:43.41 | *** join/#asterisk hjoe (~hjoe@209-255-165-97.ip.mcleodusa.net) |
| 19:45.54 | *** join/#asterisk Guifort (~Administr@78.112.90.5) |
| 19:49.48 | *** join/#asterisk m4xx (4b909aa5@gateway/web/freenode/ip.75.144.154.165) |
| 19:50.57 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 19:51.06 | m4xx | hello again =] |
| 19:51.47 | m4xx | i'm trying to use the phpagi classes |
| 19:51.55 | hjoe | hello. any favorite SIP providers? |
| 19:52.21 | m4xx | i doubt they're of great quality but i just found didforsale.com they're cheep |
| 19:52.25 | *** join/#asterisk agroman (agroman@otaku.freeshell.org) |
| 19:53.12 | m4xx | i'm testing the ping script, and when i attempt to enter an IP to ping it doesn't capture anything in the get_data function |
| 19:53.28 | hjoe | thx. i have been using voipstreet.com metered service. seems to be ok |
| 19:53.55 | m4xx | dfs has unlimited in coming and .8 cents per minut outgoing to us 48 states |
| 19:54.15 | m4xx | only up to 20 connections per did though |
| 19:54.31 | agroman | @gruvfunk: missed your response. i lost my connection |
| 19:54.33 | m4xx | but i dont know anything |
| 19:55.15 | agroman | is there an irc log for this channel? |
| 19:55.24 | m4xx | i dont suppose matt a or david e hang out in this chan do they? |
| 19:55.41 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 19:55.47 | hjoe | i'll have to check that one out. |
| 19:56.02 | gruvfunk | @agroman - I said: gtalk module looks good, my gtalk.conf [general] ans an externip= statement in it, otherwise our confs are near identical |
| 19:56.18 | gruvfunk | agroman and check your port 5222 UDP across your firewall(s) if any |
| 19:56.21 | agroman | strange. wonder if there's some other config I'm missing elsewhere |
| 19:57.13 | agroman | I'm running it on a vps without any netfilter rules |
| 19:57.16 | agroman | :-\ |
| 19:57.55 | agroman | maybe externip is significant. i didnt think i'd need to use it if there's not nat. |
| 19:59.05 | agroman | nope, still getting "could not find recipient" |
| 20:00.59 | *** join/#asterisk DrDamnit (~michael@highpoweredhelp.com) |
| 20:01.17 | DrDamnit | Was Originate removed from the AMI in 1.8.3.2? http://pastebin.com/yN6rRCzc |
| 20:02.45 | Chainsaw | DrDamnit: It's now channel originate |
| 20:03.15 | Chainsaw | DrDamnit: Other then that your dial plan should work verbatim. |
| 20:03.16 | DrDamnit | So it should be Action: Channel Originate? |
| 20:03.24 | Chainsaw | nods |
| 20:03.33 | DrDamnit | bows to Chainsaw |
| 20:03.37 | DrDamnit | thanks! |
| 20:03.42 | Chainsaw | Any time :) |
| 20:05.18 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-145.cablep.bezeqint.net) |
| 20:05.26 | gruvfunk | @agroman so outbound calls are not working through google voice? who is making the call, what extension, did not see that in your configs |
| 20:05.58 | gruvfunk | I see a SIP/sip-phone |
| 20:06.10 | agroman | that's for incoming calls, which I havent tested yet |
| 20:06.17 | *** join/#asterisk [loy] (~nobody@95.72.238.2) |
| 20:07.02 | DrDamnit | Chainsaw: No Joy. http://pastebin.com/CmtEGCS1 |
| 20:07.19 | agroman | my sip.conf dumps my sip ata into context default which has the extensions for dialing out to pstn through gtalk in extensions.conf |
| 20:08.00 | Chainsaw | DrDamnit: system ("$ASTERISK -rx \"channel originate Local/*LO$a_agent#$a_location\@local/n application NoOp\" >/dev/null 2>&1"); |
| 20:08.22 | Chainsaw | DrDamnit: That works here; I appreciate it's in a slightly different context... but adding channel would normally suffice. This is AMI? |
| 20:08.54 | gruvfunk | have you configured the google voice number correct? have you enabled the gtalk user and performed 1 call via gtalk first? |
| 20:08.54 | DrDamnit | This is weird. manager show commands lists Originate, but ListCommands via the AMI does not: http://pastebin.com/W2wY4Q8P |
| 20:10.03 | *** join/#asterisk minaguib (~mina@modemcable098.129-202-24.mc.videotron.ca) |
| 20:10.27 | agroman | gruvfunk, aha... maybe that's it. I have setup the google voice number and it's been forwarding to my cell phone for months prior to me trying to get asterisk connected. |
| 20:10.40 | agroman | gruvfunk, what would I need to do to set it up for gtalk? |
| 20:11.08 | DrDamnit | Chainsaw: This is weird. manager show commands lists Originate, but ListCommands via the AMI does not: http://pastebin.com/W2wY4Q8P |
| 20:11.20 | agroman | gruvfunk, Google Voice Settings -> Forward calls to "Google Chat"? |
| 20:12.03 | gruvfunk | yes |
| 20:12.23 | gruvfunk | login from a gtalk client, or gmail web plugin, call your GV # to test |
| 20:12.30 | gruvfunk | if it rings in your gtalk user or gmail client.. you're good |
| 20:12.34 | Chainsaw | DrDamnit: Sounds like a genuine bug then. |
| 20:12.42 | Chainsaw | DrDamnit: Have you had a search through issues.asterisk.org at all? |
| 20:13.02 | DrDamnit | searches.. |
| 20:13.15 | agroman | gruvfunk, that was it! ha... |
| 20:13.32 | *** join/#asterisk carloimperia (~carloimpe@109.117.228.215) |
| 20:13.55 | agroman | gruvfunk, thank you! what a simple thing to miss.. |
| 20:14.13 | agroman | now to figure out if I can get asterisk to register with MS OCS |
| 20:14.57 | gruvfunk | @agroman: awesome, glad to help |
| 20:20.38 | DrDamnit | Chainsaw: Don't see it as a bug, should I report it? |
| 20:20.58 | Chainsaw | DrDamnit: It seems inconsistent, that's for sure. |
| 20:21.10 | drfreeze | Trying 1.8 for the first time (1.8.3.2) - but using a TDM card |
| 20:21.12 | drfreeze | Since I'm new to 1.8, I'm not sure if there is a way to show this in 1.8 or if it is due to using TDM and not PRI |
| 20:21.14 | Chainsaw | DrDamnit: Does the actual command work, or does it fail? |
| 20:21.15 | drfreeze | Since I'm new to 1.8, I'm not sure if there is a way to show this in 1.8 or if it is due to using TDM and not PRI |
| 20:21.24 | Chainsaw | drfreeze: Is there an echo in here? |
| 20:21.34 | DrDamnit | Chainsaw: I'll report it. The worst that can happen is they will ban me. :-) |
| 20:21.48 | Chainsaw | DrDamnit: I doubt that very much :) |
| 20:21.56 | *** join/#asterisk joshaidan (~brianj@24.109.210.41) |
| 20:23.03 | drfreeze | Chainsaw: already addressed? |
| 20:26.37 | drfreeze | tell me there is a fix |
| 20:26.43 | drfreeze | or do I drop back a version? |
| 20:29.09 | agroman | anyone know if it's possible to register asterisk as a sip client with MS OCS? I'm trying this http://pastebin.com/sau97MKe and it's not working. sip registration timeout. |
| 20:35.38 | *** join/#asterisk saxa (~sasa@host242-95-static.223-217-b.business.telecomitalia.it) |
| 20:37.12 | drfreeze | http://forums.digium.com/viewtopic.php?f=1&t=76980&start=0 |
| 20:38.05 | waterfoul | I keep getting a segmentation fault when calling out, how can i figure out why? |
| 20:48.27 | agroman | <PROTECTED> |
| 20:48.27 | agroman | REGISTER 10 headers, 0 lines |
| 20:48.27 | agroman | Reliably Transmitting (no NAT) to 144.230.168.10:5061: |
| 20:48.27 | agroman | REGISTER sip:sprint.com SIP/2.0 |
| 20:48.28 | agroman | Via: SIP/2.0/TLS 96.45.183.92:5061;branch=z9hG4bK0c1fbb6d |
| 20:48.28 | agroman | Max-Forwards: 70 |
| 20:48.28 | agroman | From: <sip:pat.m.padgett@sprint.com>;tag=as3d8b8c33 |
| 20:48.29 | agroman | To: <sip:pat.m.padgett@sprint.com> |
| 20:48.29 | agroman | Call-ID: 7f26d5ce62a8745716a9ed2a62917fd2@96.45.183.92 |
| 20:48.30 | agroman | CSeq: 104 REGISTER |
| 20:48.30 | agroman | User-Agent: Asterisk PBX 1.8.3.2-1digium1~lucid |
| 20:48.31 | agroman | Expires: 120 |
| 20:48.31 | agroman | Contact: <sip:from-ocs@96.45.183.92:5061;transport=TLS> |
| 20:48.32 | agroman | Content-Length: 0 |
| 20:48.33 | agroman | --- |
| 20:48.33 | agroman | [Apr 14 16:48:05] NOTICE[16267]: chan_sip.c:12232 sip_reg_timeout: -- Registration for 'pat.m.padgett@sip.sprint.com' timed out, trying again (Attempt #3) |
| 20:48.44 | SunTsu | well deserved |
| 20:51.46 | *** join/#asterisk agroman (agroman@192.94.73.16) |
| 20:52.01 | agroman | at least I didnt paste my f'ing password... dammit. |
| 20:54.29 | EugeneKay | "yet" |
| 20:55.12 | *** join/#asterisk JonathanRose (~jonathan@nat/digium/x-mnqkinqojimessyp) |
| 21:01.20 | jaytee | ~pb |
| 21:01.20 | infobot | [~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. |
| 21:04.06 | agroman | yeah, my right-click is mapped to paste. fumble finger == mess. i need to change that setting. |
| 21:04.25 | agroman | http://pastebin.com/1xgvsJ31 |
| 21:05.42 | *** part/#asterisk wesphillips (~wphill04@137-237-233-124.harris.com) |
| 21:08.15 | waterfoul | how do i find out why asterisk is segmentation faulting |
| 21:12.14 | Chainsaw | waterfoul: Provided you build it with debugging symbols, you could attach a debugger. |
| 21:12.25 | leifmadsen | waterfoul: https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace |
| 21:15.30 | *** part/#asterisk hjoe (~hjoe@209-255-165-97.ip.mcleodusa.net) |
| 21:16.05 | *** join/#asterisk chandoo (~chandoo@ool-4a5a2824.dyn.optonline.net) |
| 21:17.55 | waterfoul | where do i find the core dump file |
| 21:19.53 | agroman | find / -name core -mtime -1 (will find any file named core that was modified in the past day) |
| 21:22.43 | waterfoul | ti only found /proc/sys/net/core |
| 21:22.46 | waterfoul | ti only found /proc/sys/net/core*it |
| 21:22.49 | waterfoul | grr |
| 21:22.53 | leifmadsen | infobot: tell leifmadsen about asteriskcookbook |
| 21:31.06 | leifmadsen | I'm a little disheartened at the lack of "Likes" on the OFPS site for the Cookbook and The Definitive Guide (in relation to the rest of the books there) |
| 21:33.27 | *** join/#asterisk minaguib (~mina@modemcable098.129-202-24.mc.videotron.ca) |
| 21:34.14 | leifmadsen | Actually I guess the number comes from the tweets, but still :) |
| 21:34.53 | *** part/#asterisk The_Boy_Wonder (~manbearpi@asterisk/batman-developer/dvossel) |
| 21:38.43 | Chainsaw | All these people worrying whether their "friends" will "like" them. |
| 21:38.50 | Chainsaw | It's Second Life all over again. |
| 21:39.31 | EugeneKay | What, there's been a flying genitalia attack? |
| 21:40.38 | *** part/#asterisk waterfoul (~chatzilla@67.129.121.92) |
| 21:43.31 | drfreeze | Is there a way to get callprogress/logging/debugging on in asterisk 1.8? |
| 22:00.30 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-155-146.cablep.bezeqint.net) |
| 22:03.38 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 22:04.22 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 22:13.14 | *** join/#asterisk karmst (~kevin@s1.acclaimpc.com) |
| 22:13.22 | karmst | Hello |
| 22:13.32 | karmst | anyone in here? |
| 22:18.27 | fauxalliance | karmst, a whole lot even |
| 22:18.36 | *** join/#asterisk Not_shmaltz (~Athens@74.92.102.230) |
| 22:18.43 | karmst | I've got a weird issue |
| 22:19.05 | karmst | there doesn't seem to be a root cause for it |
| 22:21.32 | Not_shmaltz | I have several DIDs from one provider, -- one trunk, and I sell an IVR to clients, and I would like to limit the number of concurrent calls based on the DID. I can't set the calllimit in the sip trunk. |
| 22:21.55 | Not_shmaltz | Is there any way to know what calls are active based on the DNID |
| 22:21.57 | Not_shmaltz | ? |
| 22:25.39 | Not_shmaltz | anyone here? |
| 22:34.20 | *** join/#asterisk cyford (~cyford@96-25-169-243.gar.clearwire-wmx.net) |
| 22:39.44 | *** join/#asterisk KavanS (~KavanS@LINBIT/KavanS) |
| 22:40.52 | KavanS | trying to get polycom 330 to work with polycom 650 in attendant or "buddy" mode for BLF functionality, running asterisk 1.4.32 |
| 22:45.01 | *** join/#asterisk eject_ck (~eject_ck@62.205.134.210) |
| 22:48.46 | *** join/#asterisk mythicalbox (~mythicalb@adsl-99-99-151-110.dsl.lsan03.sbcglobal.net) |
| 23:03.31 | *** join/#asterisk michael-i (~michael@204.11.230.58) |
| 23:03.49 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
| 23:08.14 | *** join/#asterisk Dovid (~Dovid@213.8.121.90) |
| 23:08.20 | Dovid | anyone in Austria here ? |
| 23:19.48 | *** join/#asterisk pinoyskull (~pinoyskul@112.198.64.81) |
| 23:21.26 | *** join/#asterisk slum (~s@173-9-8-170-BusName-boston.ma.boston.hfc.comcastbusiness.net) |
| 23:22.21 | slum | I have a problem in my dialplan, without using WaitForSilence(), outbound SIP calls are being picked up immediately by a Local channel, not alllowing the SIP call to actually ring on the other side |
| 23:22.49 | *** join/#asterisk ariel_ (~chatzilla@unaffiliated/abatista) |
| 23:27.01 | *** join/#asterisk slum (~s@192.sub-174-254-162.myvzw.com) |
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| 23:49.53 | *** join/#asterisk Schreiber1337 (cee4b465@gateway/web/freenode/ip.206.228.180.101) |
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