IRC log for #asterisk on 20081110

00:01.11*** join/#asterisk shinao1 (n=shinao1@62.173.48.42)
00:02.36*** join/#asterisk timburke (i=timburke@unaffiliated/timburke)
00:05.50*** join/#asterisk squish102 (n=squish10@cpe-098-024-067-184.carolina.res.rr.com)
00:06.36squish102could any1 give me advice on what to use for the following simple home setup?
00:07.04squish102i have family in bouth south africa and the US. i live in US and would like to make a skype to voipbuster gateway
00:10.55*** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com)
00:19.34*** join/#asterisk thing1 (n=Dwayne@64.42.227.97)
00:20.42thing1hi, i have a cisco 186 ata, which connects to a call manager, i would like to plug phone1 and 2 into my asterisk box is this possible, if so is it fxs or fxo signalling?
00:20.58*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)
00:23.42*** join/#asterisk fors1 (n=forsen@pat-tdc.opera.com)
00:23.51*** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk)
00:35.35*** join/#asterisk etm124 (n=edmiller@216.37.164.100)
00:37.00etm124hello all. im having bit of trouble making outgoing phone calls. ive set up asterisk servers before, but was using zaptel.
00:37.11etm124i 'believe' i have set up my TDM400p card correcty.
00:37.33etm124i am getting now a     -- Executing [s@macro-trunkdial:2] Dial("SIP/208-08e34c18", "Zap/g1/17174211011") in new stack
00:37.33etm124[Nov  9 14:13:03] WARNING[7469]: channel.c:3051 ast_request: No channel type registered for 'Zap'
00:37.34etm124error
00:37.43etm124can anyone be of assistance on where to start checking?
00:37.49*** join/#asterisk lowlevel (n=Stuart@lowlevel.ca)
00:48.45lmadsenetm124: are you using DAHDI now?
00:49.04etm124i believe so.
00:49.13beeketm124: If you're using DAHDI then you'll need to use DAHDI/g1..
00:49.17lmadsenetm124: core show modules like dahdi
00:49.35*** join/#asterisk Kumbang (n=dsp@rusnas.paume.itb.ac.id)
00:49.49lmadsenetm124: or read the Zaptel-to-DAHDI.txt file in the root of your asterisk source directory for how to make it understand 'Zap'
00:50.10etm124ill check it out, thanks lmadsen.
00:51.38etm124DAHDI       DAHDI Telephony Driver w/PRI             no           yes          no
00:51.42etm124looks like i am using DAHDI
00:51.52etm124however, Devicestate is 'no'.
00:53.45*** join/#asterisk km2 (n=x@mobile-166-217-117-167.mycingular.net)
00:54.33beeketm124: You'll probably need to do a dahdi_cfg to configure that card.
00:55.04beeketm124: Basically anything you did with zaptel_* will need to be done with dahdi_*
00:55.49etm124interesting. thank you beek.
00:56.28TuxguyIs it possible to use the Vonage ATA for another network?
00:59.46thing1is this possible:     analog phones ----------- channelbank(FXS) ---------- asterisk ----------- channelbank(FXS) ---------------- Cisco ATA ----------- Callmanager
01:01.12thing1want to use call manager for outgiong calls
01:02.26ManxPowerthing1: it will work, but calls won't disconnect properly all the time, maybe never at all
01:03.09thing1will the FXS signalling work for the ATA?
01:03.42ManxPowerAnother option is Asterisk -> SIP -> Call Manager
01:03.53etm124beek: thanks again for your help. i think dahdi is correctly configured. i took your suggestion and changed Zap/g1 to DAHDI/g1, but still get the same error. COuld ayou point me somewhere else to look?
01:03.59ManxPoweror maybe Asterisk -> T-1/E-1 crossover -> Call Manager
01:04.19beeketm124: You're restarted * and dahdi, right?
01:04.25etm124Yessir.
01:04.27ManxPoweror heck Telco -> PRI -> Asterisk -> Call manager
01:04.45beekwhat does "dahdi show channels" give you?
01:05.01etm124asterisk01*CLI> dahdi show channels
01:05.01etm124<PROTECTED>
01:05.20thing1i have asterisk -> T1 -> Adtran Channel Bank -> ATA -> Call manager, thats the only way my provider will allow me to connect with thier call manager
01:06.06beekYou should have two files:   /etc/dahdi/system.conf    and /etc/asterisk/chan_dahdi.conf.   Are they present and populated?
01:06.29ManxPowerthing1: ATAs don't provide the correct signal to tell Asterisk or Call Manager the call has been disconnected.
01:06.49ManxPowerWho, exactly, is your provider?
01:07.12ManxPowerIn any case, you have my recommendation.  Take it or leave it.
01:07.18etm124yes they are. system.conf: http://pastebin.com/d58879e6f
01:08.04etm124chan_dahdi.conf: http://pastebin.com/d2f1e0f7b
01:08.18*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-fce1e3c18a4465ac)
01:09.03beeketm124: Are the modules for that card loaded?
01:09.53etm124as far as i know. i did a modprobe wctdm
01:10.02etm124is there a command to double check that?
01:11.55thing1i run a vonager box no problem
01:12.21beeketm124: Does "core show channeltypes" show DAHDI?
01:12.39etm124yes: DAHDI       DAHDI Telephony Driver w/PRI             no           yes          no
01:12.51etm124the Devicestate says no, however.
01:13.46beekHow about "dahdi show channel 1"
01:14.00etm124asterisk01*CLI> dahdi show channel 1
01:14.00etm124Unable to find given channel 1
01:14.02etm124:(
01:14.34beekHmmm... then it's definitely not configured properly yet.  I hate to sound microsoftesk, but perhaps a quick reboot may get everything loaded and discovered properly.
01:14.43etm124gaaaaasp. a reboot?
01:14.51etm124ha, let me give it a whirl. thanks beek.
01:14.52beekI know... I know.
01:16.24*** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM0014045acc3c.cpe.net.cable.rogers.com)
01:18.00*** join/#asterisk kamanashisroy (n=kamanash@119.30.34.4)
01:21.34etm124im afraid the channel was still not found beek.
01:21.42etm124asterisk01*CLI> dahdi show channel 1
01:21.42etm124Unable to find given channel 1
01:22.14beeketm124: Hmmm... well that's not good.   I don't understand -- a reboot always fixes a MS machine.  ;-)
01:22.21etm124haha.
01:22.29etm124im running centos 5
01:22.33orkidwhere as a reboot always efs up a lnux one
01:22.39*** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com)
01:22.46beekYou did install CentOS5 and turned off SELINUX, right?
01:23.01etm124asterisk 1.4.22
01:23.30beekorkid: That's because we all start/stop things manually on a Linux box and forget to update the startup.
01:23.39beeketm124: You turned off SELINUX?
01:23.46etm124are you referring to iptables, beek?
01:23.51beeketm124: No.  SELINUX.
01:25.00Kattylmadsen: do you twitter?
01:25.07beekdo a cat /etc/sysconfig/selinux and look for the line that says:  SELINUX=[enforcing|permissive|disabled]
01:25.08lmadsenKatty: I do... sometimes
01:25.09etm124just turned it off
01:25.25lmadsenKatty: leif_madsen
01:25.34beeketm124: That will kick you in the privates.
01:25.47etm124ha, lets try.
01:26.09Kattylmadsen: following. you know there's an app that will update your twitter to facebook, right?
01:26.18beekJust change /etc/sysconfig/selinux to "SELINUX=disabled" so that it won't start again on your next boot.
01:26.22lmadsenI do... I choose not to use it :)
01:26.26Kattykk
01:26.29lmadsenbut twitter now loads on my server :)_
01:26.35lmadsens/server/website/
01:26.55etm124hmph. still nothing beek. still getting the same 'Unable to find given channel 1'.
01:27.21beeketm124: Reboot once again.
01:27.22lmadsenlspci ?
01:27.38lmadsencomputer sees the card?
01:27.53Kattyi'll card YOU in a minute.
01:28.09lmadsensounds kinky
01:28.23etm124wow. no signs of my card when lspci
01:28.48lmadsenmight want to reseat it then
01:31.27mankashwhat is the reason for this error:
01:31.28Kattyhm.
01:31.28mankashUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
01:31.33etm124taking a break from this. thanks for your help lmadsen and beek. ill be back on a bit later.
01:31.44etm124mankash, asterisk isnt started.
01:31.55etm124just type asterisk
01:31.55lmadsenmankash: means asterisk isn't running
01:31.58beeketm124: you're welcome.   Good luck.
01:32.06*** join/#asterisk BeeBuu (n=beebuu@219.130.254.164)
01:32.09mankashit is started I can see it running on  the console
01:32.24mankashbut from remote if I try to connect it gives me this error
01:32.32mankashmy sip phone is regsitered
01:33.10mankashI think it is a permission issue
01:33.56BeeBuuis Set(foo=${ODBC_USER_DATABASE(${EXTEN})}) put result to foo ?
01:34.04lmadsenyes
01:35.36Kattyi'm hungry.
01:36.33BeeBuulmadsen: how can i know the query runing? set verbose ?
01:38.58*** join/#asterisk joat (n=joat@ip70-174-79-200.hr.hr.cox.net)
01:39.30*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
01:40.03*** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200)
01:43.28lmadsenBeeBuu: with ODBC, it's pretty much obscured and you can't unless you enable logging on your sql server
01:43.51lmadsendo 'odbc show' to make sure your ODBC connection is connected
01:44.04lmadsenyou should see, "connected: yes"
01:45.34BeeBuui just get one line--->DSN: asterisk
01:46.54BeeBuulmadsen: what's that mean?is it oK?
01:47.14lmadsenyou should see, "connected: yes"
01:47.23lmadsenif not, then it is not connected
01:47.42lmadsenBeeBuu: follow the instructions in the book at www.asteriskdocs.org, I think chapter 12
01:47.47lmadsen"database integration"
01:48.05BeeBuuthanks,lmadsen,reading...
01:48.56orkidis the online pdf the final version? or has it been proofed afterwards? it has a few spelling mistakes, etc, iirc
01:49.06orkidthanks for the book btw, it's great :09
01:49.08orkid:)
01:51.26[TK]D-Fenderlmadsen: Given they app runs a pretty limited number of commands it'd be great if it could verbose them
01:51.50[TK]D-Fenderlmadsen: Not like its something that would flood CLI disproportionately
01:53.28Kattyello fender.
01:53.59etm124Katty: go eat. i just picked up some dinner. well, reheated :(
01:55.24Kattyetm124: had a bagel.
01:57.04*** join/#asterisk joesuffceren2 (n=Dell@75.143.183.26)
01:57.54joesuffceren2I have a cisco 7940 setup as a remote extension (or, that's what I'm trying). It will register, I can use it to call extensions connected to my asterisk server, and I can call it from the pstn via it's did, but I cannot use it to place outgoing calls to the pstn. This pastebin shows the CLI output (verbosity 10) when I try to place a call: http://pastebin.com/m7707bea2
01:57.54joesuffceren2incidentally, connecting to the extension via xlite produces the exact same behaviour, so I don't think it's a cisco 7940 thing
01:58.30ManxPowerjoesuffceren2: Sorry, I can't help with GUI setups.
01:58.39ManxPower~freepbx
01:58.39jbot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
02:00.24joesuffceren2I asked in freepbx, and I understand you're not wanting to dive into their dialplans, etc., but this isn't a freepbx specific issue...
02:00.26joesuffceren2*sigh*
02:00.57ManxPowerjoesuffceren2: Tell ya what.  If you remove all the freepbx crap form your dialplan for whatever number you are dialing then I'll take a look at it.
02:01.03lmadsen[TK]D-Fender: possibly... but seeing asterisk try and execute the commands isn't very useful if it isn't getting to the server. Basically if ODBC is connected via res_odbc.conf, and 'odbc show' shows connected, everything else can easily be debugged from outside of asterisk (and most likely should)
02:01.13ManxPowerBTW, why are you using something you can't even get support for?
02:02.27ManxPowerYou should simply have a Dial, and for troubleshooting a Noop(HANGUPCAUSE is ${HANGUPCAUSE} and DIALSTATUS is ${DIALSTATUS}) as the priority after the Dial line.
02:03.16ManxPowerThat should give less than 5 lines of output.  Far better than the EIGHTY NINE lines of output the FreePBX crap has.
02:04.42[TK]D-Fenderlmadsen: True but it'd be more than easy to provider that "what * parses" bit
02:05.26ManxPowerLooks like it will be about 9 lines of output.
02:05.30[TK]D-Fenderjoesuffceren2>I asked in freepbx, and I understand you're not wanting to dive into their dialplans, etc., but this isn't a freepbx specific issue... <- Yes, it IS>  If you can dial some things but not others then it IS dialplan and FreePBX's problem
02:05.43*** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200)
02:06.19ManxPowerOf course it's a FreePBX problem.  I just offered to help him troubleshoot it without FreePBX.   Then he stopped talking.
02:07.09joesuffceren2even though that happens only with my one extension that's offsite and not with the 9 that are, it's a diaplan issue and not a sip header issue?
02:07.16drmessanojoesuffceren2: You have x number of extensions working, one not.. with 2 different SIP clients on that extension
02:07.32drmessanoYour extension is screwed, and there's no way to troubleshoot it
02:07.39drmessanoDelete, recreate, and move on
02:08.03joesuffceren2drmessano, I am in the process of trying that, but the one extension that isn't working is outside the firewall, while all the others are inside
02:08.13ManxPowerfunny thing is, the pastebin he pasted indicates the call worked just fine.
02:08.16*** join/#asterisk terracon (n=garry@CPE001cf097a750-CM0012254076d6.cpe.net.cable.rogers.com)
02:08.25drmessanoSo it's a NAT audio problem
02:08.29drmessano~sipnat
02:08.30jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
02:08.30joesuffceren2yeah, it seems to, but it just gives me dialtone again
02:08.41joesuffceren2the call doesn't actually place (because I called my cell and it never rang)
02:08.44ManxPowerdialtone is generated by the local phone
02:08.46ManxPowernot Asterisk
02:08.52mankashwhy when I do ps aux | grep asterisk show me too many lines for asterisk
02:09.09ManxPower    -- Called 1/8005551212     -- Zap/1-1 is proceeding passing it to SIP/113-00b62d10     -- Zap/1-1 answered SIP/113-00b62d10     -- Hungup 'Zap/1-1'
02:09.11joesuffceren2right, but my point is when I call a local extension, I get ringing and the vm of the extension that I call
02:09.24ManxPowerThis is a VALID and WORKING call.  Now you are saying the call doesn't even get placed.
02:09.36joesuffceren2my cell phone never rings
02:09.37*** join/#asterisk darkskiez (n=mbryars@195-11-205-216.suip.mezzonet.net)
02:09.41joesuffceren2I replaced my cell number with 5551212
02:09.58ManxPowerjoesuffceren2: Your cell phone number is 8005551212?
02:10.04ManxPowerdon't do that.
02:10.14*** join/#asterisk prodyan (n=ian@124.104.71.66)
02:10.33ManxPowerso we really don't know if it dialed a 1 before the area code or it did not dial a one before the area code.
02:10.41drmessanoYour cell never rings because the calls hangs up before it makes it to the cell
02:11.03joesuffceren2sorry if that offends you, manxpower. I'm new to IRC and I wasn't sure about the safety of displaying my personal phone number to 260 people I've never met before. forgive me if I'm paraniod
02:11.20joesuffceren2paranoid*
02:11.22drmessanoOk, i've already said twice how to fix it
02:11.25drmessanoLet him keep pasting
02:11.30ManxPowerdrmessano: It could be a reinvite problem or a localnet/externip problem, or a "dial before telco is ready problem"
02:12.12ManxPowerMy bet is localnet/externip, but you already gave him the link to fix that.
02:12.25drmessanoYeah, I think thats the solution
02:12.39*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
02:12.54ManxPowerjoesuffceren2: in the future, if you are concerned replace JUST the last 4 digits of the number.  Generally that info is not important.
02:13.11ManxPowerwhen pasting config files mask ONLY the passwords.
02:13.27joesuffceren2will do. do you prefer fake numbers for those last 4 or xxxx?
02:13.45ManxPowerdrmessano: I wonder if he can figure how to adapt the "generic asterisk" instrucitons into "works in freePBX"
02:14.00ManxPowerjoesuffceren2: xxxx is best as we know it's fake.
02:14.12*** join/#asterisk nikko (n=nikko@12-218-250-137.client.mchsi.com)
02:14.12etm124welp, found out that my card was listed when i did an lspci, just nothing i've heard of. Was recognized as Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
02:14.36BeeBuui passed  follow command:  echo "select 1" | isql -v asterisk-connector, but still get odbc connect,why?
02:14.37ManxPoweretm124: that just means you have an older version of the pci vendor id library for your distro
02:14.43etm124i tried calling the number in which the phone line is registered, and im not seeing anything in the cli
02:15.01etm124thanks ManxPower. i am used to seeing it as TDM Wildcard...
02:15.25ManxPoweretm124: upgrade your libpci and you might see it again
02:16.17etm124when u say upgrade, do you mean to a newer version?
02:16.25ManxPoweretm124: correct.
02:16.32joesuffceren2drmessano, the contents of my sip_nat.conf are:
02:16.44joesuffceren2externip=66.83.116.154
02:16.46joesuffceren2localnet=10.0.0.0/255.255.255.0
02:16.52ManxPowerjoesuffceren2: We don't use sip_nat.conf.
02:17.55etm124ManxPower: im running 1.4.7
02:17.55etm124i believe that's the latest.
02:17.55ManxPoweretm124: This is an OPERATING SYSTEM issue.
02:18.01etm124my mistake.
02:18.06joesuffceren2and deleting and recreating the extension didn't change the situation unfortunately
02:18.35ManxPowerjoesuffceren2: I'm sure it didn't.  Now are you going to edit extensions.conf so we can do it the old fashioned way?
02:18.43*** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal)
02:19.21ManxPowerjoesuffceren2: how many ports did you forward in your router?
02:19.32ManxPowerwell, what ports and what protocols?
02:29.08*** join/#asterisk joesuffceren2 (n=Dell@75.143.183.26)
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02:29.22*** join/#asterisk lolipops (n=lolipops@modemcable238.118-82-70.mc.videotron.ca)
02:30.05joesuffceren2drmessano and manxpower, thanks for the help. I'll keep working on it. I won't bother you any more. Have a good night
02:32.33lolipopsim trying to wait for user input with Background();, but it seems that it ignores the response timeout and just exists as soon as the message is done playing. any ideas?
02:32.42lolipopsexits*
02:34.27*** join/#asterisk BhaalWK (i=bhaal@freenode/staff-emeritus/bhaal)
02:34.47[TK]D-Fenderlolipops: "autofallthrough=no" <- should be under [genera] in extensions.conf
02:34.53[TK]D-Fenderlolipops: "autofallthrough=no" <- should be under [general] in extensions.conf
02:35.21lolipopsill try that. thanks.
02:38.15squish102is there any free skype -> asterisk gateways? i only need one skype extension
02:39.34drmessanono
02:39.56Mark_LoganI second that "no".
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02:41.09*** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200)
02:41.09squish102thanks
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02:46.19*** mode/#asterisk [+o d3wayne] by ChanServ
02:48.28Mark_LoganLike "Thanks!" thanks or "Thanks for nothing" thanks? :P
02:50.40squish102lol, no, like thanks that someone just gave me an answer
02:51.02Kattyjbot: hi
02:51.03jbothello, katty
02:51.27Katty[TK]D-Fender: have you ever had a tuna melt>
02:51.49[TK]D-FenderKatty: Yup, oven baked open faced goodness
02:52.22Kattyoh? never tried the oven baked variety.
02:52.24Kattydo you have a recipe?
02:52.31lolipops[TK]D-Fender, i guess ill fix my little problem with silence/10.wav. easier and cleaner.
02:55.37[TK]D-FenderKatty: Basiacally just a tuna-salad sandwich, open faced... maybe a slice of cheese.
02:55.54RB2Has anyone here used lylix to run a hosted asterisk instance?
02:56.58*** join/#asterisk rcahilig (n=sysad@202.78.75.246)
02:57.49Katty[TK]D-Fender: what kind of bread to you use?
02:57.51Katty[TK]D-Fender: broil?
03:02.05lolipops[TK]D-Fender, why is it TIMEOUT(response) seems to apply to Read(); but not Background();?
03:02.35[TK]D-FenderKatty: For bread we've got a Quebec made brand whose 12 grain  variety is un-fucking-believeably good.  Having visited around I've never seen anything like it anywhere in standard bagged form
03:02.46[TK]D-Fenderlolipops: You are mistaken
03:03.01Katty[TK]D-Fender: pity. hard to get good bread around here.
03:04.15Kattymaybe i should start making my own mini loaves
03:04.17lolipops[TK]D-Fender, i have an extension here that uses Read(); and it waits the whole default 10 seconds, this other one with Background(); falls out as soon as it's done playing
03:05.13[TK]D-Fenderlolipops: Yes and I told you what to add and you aren';t showing me the problem.
03:05.21[TK]D-Fenderlolipops: I know full well how timeouts work.
03:06.05lolipopscalm down. im just pointing out something that seems illogical to me here.
03:06.39Kattyfender is always calm.
03:06.41[TK]D-Fenderlolipops: PB <-
03:06.48Kattyexcept when he chops a digit nearly off.
03:07.13[TK]D-FenderKatty: thats right, and if 1 more person slanders me like that I'LL KILL THEM!
03:07.22[TK]D-Fender:D
03:07.30*** join/#asterisk BeeBuu (n=beebuu@125.95.28.194)
03:07.40Kattypats [TK]D-Fender
03:07.43Kattyyou do that dear.
03:08.00Mark_LoganJeez, we're having a real emotional crash going on here.
03:08.05BeeBuuwhere can i set the transfer press key time?
03:08.27Mark_Loganmaybe next time lets go with "asterisk -rvv" and not "asterisk -rvvvvvvvvvvvv" alright?
03:10.18[TK]D-FenderKatty: On the thumb topic I need to see a doc about having that bit removed entirely.  the muscle bit that was left there curled into a bit of a knot and it useless, and a little painful under pressure.  Its like it not even "me"
03:10.49Katty[TK]D-Fender: sounds icky.
03:11.10etm124ouch.
03:11.13*** join/#asterisk RobertLaptop (n=rmiddle@164.sub-75-196-9.myvzw.com)
03:11.48[TK]D-FenderKatty: if they can do it with confidence of success I have NO issue with it.
03:12.07[TK]D-FenderKatty: its a quality of life issue for me.
03:12.16Kattyyou're also not a hypochondriac
03:12.22Kattyiw ould have already been to the doctor half a dozen times ;)
03:12.26KattyOMG IT"S CANCER
03:12.27KattyA TUMOR
03:12.30Kattyetc.
03:13.26[TK]D-FenderKatty: No, its a "healed-over" lump of useless muscle thats F-ing up my thumb!
03:13.27lolipopshttp://pastebin.com/d37c04282
03:13.51Katty[TK]D-Fender: indeed.
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03:15.42[TK]D-Fenderlolipops: failed call please
03:15.54lolipopsim sorry?
03:16.21Katty[TK]D-Fender: also, pork chops.
03:16.39Katty[TK]D-Fender: with chicken broth, honey, soy sauce, ketchup, ginger, and a lil garlic.
03:16.54[TK]D-Fenderlolipops: Show me the failed call
03:17.10`Sean:o, TASTES yummy
03:17.26Katty`Sean: oh yes, yes it does.
03:17.40Katty`Sean: especially when you simmer the pork chops alllll day long.
03:17.56Katty`Sean: with some rice and veggies, and cornstarch with the leftover sauce.
03:18.20etm124is getting hungry now.
03:19.48Kattyetm124: http://bp1.blogger.com/_WqvyAw872Ko/RnGWugLUiOI/AAAAAAAAAXM/A4UL6qTxzkA/s320/PorkChopsYumYum.jpg
03:20.22Kattyetm124: tell me that does not look good, after a long day of work and cold weather.
03:20.30etm124mmmmmmmmm.
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03:21.30lolipops[TK]D-Fender, i feel retarded, but i dont know what you want to see exactly.
03:22.37lolipops[TK]D-Fender, autofallthrough=no DOES work, but it just strikes me as odd that with autofallthrough=yes, Read(); waits 10 seconds but background returns immediately.
03:24.02[TK]D-Fenderlolipops: because autofallthrough says what happens at the end of an EXTEN <-
03:24.19[TK]D-Fenderlolipops: When you run out of priorities then you fall through
03:24.37[TK]D-Fenderlolipops: Which would be blatantly visible in the CLI of the call.
03:24.38lolipopsthe same behavior applies if I put a Playback(vm-goodbye); after Background();
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03:29.55[TK]D-Fenderlolipops: Do not put playbacks after backgrounds.
03:30.03lolipopswhy not?
03:33.07[TK]D-Fenderlolipops: because it devalidates * waiting at the end.
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03:41.08prodyanguys, how do i get the parameter of an agi script call like this AGI(test.agi|2222) - i want to get 2222
03:41.42[TK]D-Fenderprodyan: Go read the docs for whatever language you wrote your script in
03:42.01prodyanhmm oki let me check
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03:42.30prodyanthanks btw D-Fender
03:42.33lolipops[TK]D-Fender, i guess i got myself an okay setup for this now. thanks for all.
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03:49.30etm124ok. ive been fiddling with my setup for a bit now. it seems like my channels aren't being recognized...
03:49.53etm124DAHDI Tools Version - 2.0.0
03:49.53etm124DAHDI Version: 2.0.0
03:49.53etm124Echo Canceller(s): MG2
03:49.53etm124Configuration
03:49.53etm124======================
03:49.54etm124Channel map:
03:49.56etm124Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
03:49.58etm124Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
03:50.00etm124Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03)
03:50.02etm124Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)
03:50.04etm1244 channels to configure.
03:50.06etm124Setting echocan for channel 1 to mg2
03:50.08etm124Setting echocan for channel 2 to mg2
03:50.10etm124Setting echocan for channel 3 to mg2
03:50.12etm124Setting echocan for channel 4 to mg2
03:50.16drmessanoSTOP
03:50.24etm124sorry for the big paste.
03:50.30drmessanoDont do it
03:50.31drmessanoEver
03:50.45drmessano~pb
03:50.46jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
03:51.17etm124i am, however, getting an unable to find given channel 1 error.
03:51.56drmessano....
03:52.11[TK]D-Fenderetm124: You didn't show us any error
03:52.54etm124[TK]D-Fender: Unable to create channel of type 'DAHDI'
03:53.12[TK]D-Fenderetm124: Did you compile * AFTER you installed DAHDI?
03:53.27[TK]D-Fenderetm124: and please pastebin the entire failed call attempt
03:54.27etm124[TK]D-Fender: yes, i did. and http://pastebin.com/d6a16e3bd
03:54.54[TK]D-Fenderetm124: And now your configs....
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03:57.11Kattyhai cunning
03:57.44CunningPike_Katty: Hai!
03:57.44etm124[TK]D-Fender: chan_dahdi.conf: http://pastebin.com/d2f1e0f7b
03:57.56CunningPike_How are you?
03:58.46[TK]D-Fenderetm124: "dahdi show channels"
03:59.19etm124[TK]D-Fender:    Chan Extension  Context         Language   MOH Interpret
03:59.26etm124no other info from that.
03:59.43[TK]D-Fenderetm124: http://pastebin.com/d2f1e0f7b <- line 7.  You have NOT set the group you are tryig to dial
04:00.00[TK]D-Fenderetm124: 1st slear error.  next, try to reload chan_dahdi after fixing it.
04:00.03[TK]D-Fenderclear*
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04:01.47etm124[TK]D-Fender: would that be a reason my channels aren't showing up?
04:01.59[TK]D-Fenderetm124: 2 separate issues
04:02.41[TK]D-Fenderetm124: next if the fact that your spam above lists *4* channels, and here I see you trying to configure *8*
04:02.58[TK]D-Fenderetm124: Someone needs their head screwed on straight
04:08.21etm124[TK]D-Fender: your two suggestions fixed it.
04:08.22etm124thank you.
04:08.33[TK]D-Fenderetm124: You're welcome
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04:45.35*** join/#asterisk wnspark (i=0cd8f6bd@gateway/web/ajax/mibbit.com/x-6da09c6c410e8907)
04:46.00wnsparkWhat directory is asterisk installed into when you install it via the ports collection in FreeBSD?
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04:54.24postelwnspark: which asterisk or use the ports command (man ports) to query the system for where it put the files
04:54.57postelwnspark: most things go under /usr/local
04:55.27[TK]D-Fenderusuallt /usr/local/etc/asterisk
04:55.36[TK]D-Fenderthe config anyways
04:55.48wnsparki found it
04:55.49[TK]D-Fender"which asterisk" should answer the other.
05:12.19thing1how can i do an autoanswer on a zap channel
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05:13.58thing1i have a ata hooked into a channel box, now i would like to be able to dial 9 to get that line and it just keeps ringing
05:18.00[TK]D-Fenderthing1: What is a "channel box"?  And what do you mean "autoanswer"?  Describe the call from beginning to end
05:18.52thing1i want to dial an extension, like 9 and asterisk will bridge me with the line i have the ata plugged into, zap/48 right now i have dial zap/48, and it rings, but never picks up
05:19.08thing1i assume i have to have asterisk pickup the call or something
05:19.31thing1i have a 24 port adtran channel bank hooked into asterisk via t1
05:19.51[TK]D-Fenderthing1: Please start from the BEGINNING
05:20.24[TK]D-Fenderthing1: don't just say "I dial".  Start from the description of exactly what phone plugged into what device, calling on what channel.
05:20.36thing1i ahve a cisco ata which connects to a remote callmanage, i would like to be able to use the two lines for my pbx
05:21.28[TK]D-Fenderthing1: in your zapata.conf or chan_dahdi.conf you specify the context the channels will send incoming calls to.
05:21.28thing1i've connected the phone one and two into zap ports and would like to dial an extension like 9 and have asterisk give me a dial tone
05:21.43thing1how about outgoing calls?
05:22.05[TK]D-Fenderthing1: lets be clear on this since you have failed again.
05:22.17[TK]D-Fenderthing1: what is the phone you are holding in your hand plugged into?
05:22.35thing1its a sip phone plugged into the network
05:22.46[TK]D-Fenderok, SIP phone direct to *, correct?
05:22.52thing1yes
05:23.51[TK]D-Fenderthing1: Good.  then make an exten => 9,1,Dial(Zap/1) for example to pull that zaptel channel which should pull you dialtone from the ATA attached to it
05:24.24[TK]D-Fenderthing1: What is the ATA using to talk to the CM?
05:24.43[TK]D-Fenderthing1: because this is a lot of A>D>A for nothing...
05:24.52[TK]D-Fenderthina fugly setup for sure...
05:25.02thing1it's used skinny
05:25.05thing1using
05:25.21[TK]D-Fenderthing1: You have no SIP license available?
05:25.36[TK]D-Fenderthing1: It'd be far better if you could jsut send the call direct from * to CM
05:26.06thing1well the idiot on the other end is making me do it with a ata, i could do it straight with astierks i know but some head-techs are propietery idiots
05:26.19thing1i know
05:26.48[TK]D-Fenderthing1: Its doable, but its just a stupid expense and loss of functionailty, etc
05:27.05[TK]D-Fenderthing1: Pass it on that others "in the know" agree that they are morons.
05:27.07thing1right now when i dial the zap channel the ata is plugged into it just  rings
05:27.33thing1i have an fxs signal card
05:27.54thing1i also have 4 telco lines coming in on fxo lines and they work perfectly
05:28.07thing1but this is diff because it has diff signalling
05:28.50[TK]D-Fenderthing1: you need an FXO signal card, not theother way around
05:29.19thing1what is the one for telco, i always get mixed up
05:29.27[TK]D-Fenderthing1: Your ATA acts as FXO.  You therefore need an FXO card for your adtran, not an FXS.
05:29.53thing1ok
05:29.58thing1i guess thats the problem
05:30.03[TK]D-Fenderthing1: Adtran FXO port = act like a phone (uses fxs_ls signalling in Zapata)
05:30.15[TK]D-Fenderthing1: Great so retarded AND backwards
05:30.18[TK]D-Fender:)
05:31.02thing1thanks
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05:49.35orkidubuntu or centos?
05:49.37orkid:)
05:49.42orkidor debina :O
05:49.44orkiddebian
05:50.39orkid?
05:52.03[TK]D-Fenderorkid: Thoughts your coherent very aren't much.
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05:59.11drmessano[TK]D-Fender: He is amusing, despite being useless
06:00.52[TK]D-Fenderdrmessano: Talks doe funny Yoda hmmmmmmMMMMMM!??!?!??!
06:00.53troy-drmessano, i concur
06:01.04[TK]D-Fenderdoes*
06:02.43troy-[TK]D-Fender, it would seem the only way i'm getting inbound SMS is with a USB GSM Radio
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06:05.30orkid[TK]D-Fender: what distribution of linux would you recommend for asterisk. centos, debian, ubuntu server ?
06:05.32[TK]D-Fendertroy-: Which still seems to have nothing to do with *.
06:05.36orkida basic asterisk install
06:05.40orkidnot business
06:05.41orkidjust home
06:05.49[TK]D-Fenderorkid: Whichever you are capable of using to fulfil *'s requirements with
06:06.08orkidall this talk of 'realtime' and 'transcoding' makes me scared
06:06.11orkid:L
06:07.34[TK]D-Fenderorkid: Almost everybody does transcoding.  Its always a question of what to what, and how.
06:07.43orkidit seems like centos is preferred in these circles? i mean, thats what pbxin a flash uses right?
06:08.00drmessanotroy-: chan_mobile?
06:08.00[TK]D-Fenderorkid: And realtime... you don't need to care about, and more than half of those that use it probably shouldn't
06:08.07orkid[TK]D-Fender: really? i would think many people stay ulaw only
06:08.17orkid[TK]D-Fender: at least for the call portion, not the ivr sounds
06:08.22[TK]D-Fenderorkid: Who cares what PBIAF uses.  Look at what YOU can manage
06:08.23troy-drmessano, yeah
06:08.42drmessanoWhy do you need a USB GSM radio?
06:08.54drmessanoProper Bluetooth phone should do it
06:09.34troy-drmessano, much less reliable
06:09.48drmessanoHow so?
06:11.46[TK]D-Fenderorkid: Any sound not in the codec of your call is transcoded.
06:11.52[TK]D-Fenderorkid: the load is the same
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06:12.19[TK]D-Fenderorkid: Naturally its best if you aren't transcoding
06:12.49[TK]D-Fenderok, checkout tie.  Later all
06:12.52[TK]D-Fendertime*
06:12.56orkidbye
06:13.01drmessanoWho the hell wants to use ULAW for everything?
06:13.30troy-bluetooth is largely irrelevant - you are relying on a consumer phone for transmit/receive versus a purpose built radio chipset
06:14.36drmessanoSame device, metal box
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06:21.26orkidpeople with packet loss?
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06:22.47CrazyTuxDoes anyone here know how AMI works with using Originate and queues?
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06:35.17ManxPowerdrmessano: um, we use ulaw for almost everything.
06:37.27denonas do we
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06:38.31CrazyTuxHey guys, sorry on a real flaky connection, so any responses I did not see.  I was asking if anyone knows a way to view / access the AMI originate CALL queue, when sending numerous originate calls
06:40.37Miccwhat is AMI?
06:41.36MiccIf your talking about the api that astman uses, I had to modify the source to send variables with each event.
06:41.50MiccSo I could know what originate went with what newexten
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06:43.36MiccCrazyTux, I thought that was fixed in 1.4, it should send the unique id with each channel.
06:44.02CrazyTuxMicc unique ID of the channel?
06:44.22Miccno, its just a unique id you can use for each originate I think.
06:44.24drmessanoAMI is the asterisk management interface
06:44.36MiccOr set a variable that gets passed back to you in each channel.
06:44.50MiccWith the originate I think you send a setvar too.
06:45.13MiccThen it passes that var back with the events.
06:45.53MiccCrazyTux, I haven't done anything with that for more than a year. I'd have to look into that to remember exactly what I did.
06:46.21MiccThere is a way, you might have to modify the asterisk source, but there is a way.
06:47.12CrazyTuxI'm thinking from more of a global view, so I can see it all, by issuing a command possibly? core show channels perhaps, but that is not 100% reliable
06:47.24CrazyTuxwithout keeping consistent state/information on each channe
06:47.58Miccoh, then I don't know.
06:48.11MiccCrazyTux, you'll have to make your own AMI function.
06:51.33drmessanohttp://www.voip-info.org/wiki/view/AstManProxy
06:51.35drmessanoTry that
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08:19.55phpboyhey all
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09:04.08pifhi, does 1.4.22 fix some 100% CPU peaks + random crashes that 1.4.21.2 has ?
09:05.20mort_gibpif: I use 1.4.21.X and have no such issues.....
09:05.51alirajaHi all, i want to hangup all calls after 5 minutes coversation b/w customer and agent ...any idea of how i call achieve it
09:07.29aliraja*can achieve it.
09:08.18pifmort_gib: debian?
09:08.18tzafrir_laptoppif, reproducable?
09:08.36pifoh hi tzafrir_laptop I just sent you a query about 1.4.22 debs
09:09.03mort_gibpif: both Debian and CentOS
09:09.11pifnot really reproducable yet, but very annoying (production server upgraded last week)
09:09.13mort_gibBut compile from source
09:09.43pifmort: amd64 arch ?
09:11.27mort_gibpif: No Xeons
09:13.50piftzafrir_laptop: our two newly installed debian/sid dell servers with TE410P card have the same symptoms
09:14.29tzafrir_laptoppif, looking into this
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09:14.44tzafrir_laptop(well, at least for something that builds)
09:20.28pifthanks!
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09:25.12CrazyTuxDoes anyone know if AMI sends back event/action data to the same socket that created the action, or does it broadcast it to all sockets reading?
09:28.06creativxCrazyTux: AMI broadcasts it.. doesnt matter where it came from
09:28.55CrazyTuxcreativx: are you sure about that?
09:29.24CrazyTuxcreativx: I'm listening on another socket, aside from the invoking client request, and not getting anything through
09:29.37creativxCrazyTux: mine does in 1.2.. sending "originate" events from one client, amiproxy reads ami, and another client reads the amiproxy output..
09:30.43CrazyTuxcreativx: this is 1.4.21.2 so I'd imagine hmmm
09:31.56creativxbut
09:32.00creativxwhen i think about it
09:32.23creativxwell.. ive forgotten
09:32.27creativxand dont have time to check source atm :)
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10:19.42masusanyone know where i can buy an opensource predicitive dialer script ?
10:19.56masusor recommend one ?
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10:33.05kippihey
10:33.27kippihas anyone got a SoundStation IP 6000 working with openser/asterisk?
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11:01.47viraptoris there some way to get sip "From:" domain in AGI?
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11:12.35beniwtvHi all... I've got an AGI script in which I execute a DIAL command. This works fine, and I would like to execute a DeadAGI when the channel is hung up, so I put a h extension in my dialplan. However, once my AGI script has finished sending the DIAL command to Asterisk, it finishes executing. That somehow causes Asterisk to execute the h channel inmediaely, when the channel is not really hung up. Do I have to wait in the AGI scr
11:12.36beniwtvipt for the call to finish? Is there any signal Asterisk sends?
11:12.59beniwtv(sorry for the long explanation)
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11:41.25piftzafrir_laptop: I'm trying to debuild 1.4.22 (removed all patches) and it fails with "dpkg-source: error: cannot represent change to asterisk-1.4.22/doc/lang/hebrew.ods: binary file contents changed"
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11:41.42scruzgood day
11:42.18scruzi'm an asterisk newbie...while my copy of the book is downloading, how can i turn off debug mode in asterisk?
11:42.59viraptorscruz: core set debug 0
11:43.07viraptoror remove debug from your logger.conf
11:44.08phpboy[Nov 10 13:43:34] WARNING[30475]: res_agi.c:2129 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI
11:44.21phpboyWhat kind of problems would this cause?
11:45.27scruzthanks...set debug 0 worked
11:45.42scruzi think it's an old version :)
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11:48.39scruzor it didn't
11:58.19phpboy<PROTECTED>
11:58.24phpboythat does not look right
11:58.25phpboythe time :(
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12:10.11ifnotwhynothi there any pri span experts active?
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12:20.03plantseekerHow all
12:20.09plantseekerHi all
12:21.09plantseekercan anyone recommend a good site  to learn more about call recording.
12:22.28plantseekerI am particular interested to find out more about both identifying and dealing with recordings that are out of sync
12:22.32Meawgoogle can recommend you one
12:22.52phpboyifnotwhynot: what do you wanna know?
12:23.31plantseekertried asterisk calling recordings out of sync
12:23.49phpboyplantseeker: what do you mean?
12:23.54phpboy'out of sync'?
12:24.36plantseeker2 legs joined but out out of sync
12:24.50plantseeker2nd out =of
12:26.41plantseekerI have 39666 reocrdings to check
12:27.03phpboyI'm still not with you on the out of sync part
12:27.33Meawour billing system records the call.. I just simple click to listen
12:27.34plantseekerok phpboy I will try to explain what I mean
12:27.43Meawbut it's too evil to record a 39666 calls
12:27.59Meawwe record only when a customer complain about bad signal :)
12:28.10plantseekerMeaw lol
12:28.25plantseekerI have listen to a recording
12:28.48plantseekeralthough everything ok at the beginning
12:29.23plantseekeras the call progresses the two callers start talking at the same time
12:29.47plantseekerout of sync
12:32.14*** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk)
12:32.37plantseekermy question is how do I fix an out of sync recording and what is the best way to identify an out of sync recording. I obviously don't want to listen to 39666 recordings
12:34.36plantseekerI don't expect to be be given the answers just some pointer to the solutions. some good google search terms would be nice :-)
12:34.50plantseekerpointer=pointers
12:35.16plantseekerperhaps a touch typing tutorial as well :-)
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12:46.39ultrav1oletI have two questions: if asterisk supports that, and if there's software which can do that
12:47.29HeManit can do that with some subsets of that
12:47.33tzafrir_laptoppif, nice . so we need to find why that file changes at build time
12:47.54ultrav1oletthat is: imagine such a situation: someone calls into your office, with a call automatically redirected to a IP phone or SIP/IAX2 account, there's a conversation, then I would like to transfer that call to another person.
12:48.22ultrav1oletI tried playing with Zoiper and I couldn't do that
12:59.03magronezis away: cliente
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13:05.52ifnotwhynothi there waht is the best way to reload zap drivers once one make changes to the zaptel.conf?
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13:06.52ifnotwhynotphpboy : sorry had to run are u still available?
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13:08.14hi365anyone ever get this?
13:08.14hi365app_queue.c: The device state of this queue member, Local/230@from-internal/n, is still 'Not in Use' when it probably should not be! Please check UPGRADE.txt for correct configuration settings
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13:16.58clintcplantseeker: does this sound like your problem: http://bugs.digium.com/view.php?id=12837
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13:18.32clintcplantseeker: here is a bug we have filed with digium: http://bugs.digium.com/view.php?id=13745
13:20.36plantseekerThanks clintc . I will look at both the links provided.  Appreciate it.
13:21.10clintcplantseeker: one the recording has been done, I don't think you can fix it as both sides have been mixed into a mono signal
13:22.19clintcplantseeker: a workaround we are thinking about until this is fixed is to use the monitor app instead of mixmonitor... monitor can put each side in it's own left right channel... then you could fix after the fact
13:22.34phpboyI really wish I could figure out why this silly asterisk install is dropping zap calls every now and then :(
13:26.17*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
13:26.58hi365i think its a 'feature'
13:26.59*** join/#asterisk astrOdz (n=astrOdz@119.111.96.121)
13:27.09astrOdzhey
13:27.15astrOdzhas anyone tried asterisk with amazon's ec2?
13:28.26hi365there have been reports and how-to's. Want more info? Ive got a friend that know how to do it. His name is... (hint: it begins with a big blue G)
13:28.42lmadsenanyone know how I can check to see why asterisk is blocking and won't return any data from the CLI when I type in a command?
13:29.47*** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
13:30.12lmadsenhrmm... interesting, other commands will come back as long as I don't run  "sip show peers" first... if I do that, then the console blocks until I exit and jump back in
13:31.39*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
13:32.02lmadsenoh now other commands don't work anymore either
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13:36.37stoffellis there any way to keep the ringing volume to maximum on a polycom 320? (it resets after every reboot)
13:37.24stoffelllmadsen, maybe dns issues?
13:38.35lmadsenstoffell: never seen that happen.. but entirely possible
13:40.11masushi all , does anyone know where i can find the (Mysql SQL) db structure for asterisk 1.6
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13:47.46hi365how can i find which revision are included in which rleases?
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13:54.22feedshi
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13:54.54feedsI have a problem with permissions of my created user asterisktest
13:56.30feedsI put him into the asterisk group, and changed the asterisk.conf so, that he is the astctl owner, and the runuser, without changing the rungrp or astctl own group from asterisk, assuming that he already is in that group
13:57.03feedsbut when I try running asterisk as asterisktest, it says: Unable to install capabilities.
13:57.08feedswhy is this?
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13:59.15Kattymorning
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13:59.51feedsgood morning
14:00.26jayteeKatty good morning
14:00.40Kattyjaytee: allo
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14:04.05Kattyjaytee: how're the new cubs settling in?
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14:11.09jayteeok, time for me to head off to Digium for 1st day of class
14:11.11jayteelater all
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14:15.22*** part/#asterisk killfill (n=killfill@200.63.96.244)
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14:19.34Kattyso quiet this morning.
14:19.35Kattywho died?
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14:19.56anonymouz666Katty!
14:20.10anonymouz666how was the weekend?
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14:21.15[TK]D-Fendergoes off to hide the bodies
14:21.39anonymouz666[TK]D-Fender: Dexter
14:21.46*** join/#asterisk skyphyr (n=quassel@92-234-8-140.cable.ubr07.hari.blueyonder.co.uk)
14:21.58[TK]D-Fenderkicks skyphyr in the nads
14:22.06Kattyhugs anonymouz666
14:22.21Kattyanonymouz666: pretty good. took the pup to mom's house and we all raked leaves.
14:22.29Kattyanonymouz666: well riddick didn't. he jumped and chased them around.
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14:24.18tzangerodd, this te405 doesn't get detected in this old p3 pc
14:24.23lmadsenneato
14:24.28tzangerindeed
14:24.33Kattyhugs lmadsen
14:24.40lmadsenI believe I will make pancakes for breakfast...
14:24.47lmadsenhugs the Katty
14:25.07[TK]D-Fendertzanger: OLD PCI spec perhaps?  Not 2.1?
14:25.13Kattyoh? pancakes?
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14:27.26Kattygingerbread muffin mix makes excellent pancakes.
14:27.35Kattyas does the blueberry and chocolate muffin mixes.
14:27.59[TK]D-FenderKatty: Pie-Rat pup! http://xs433.xs.to/xs433/08450/peglegdog862.png
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14:28.09*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
14:28.18*** kick/#asterisk [skyphyr!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender)
14:28.44[TK]D-Fenderhopefully that'll wake up his client...
14:29.16*** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender
14:29.34Katty[TK]D-Fender: that did not parse.
14:29.48Kattymaybe it's too early.
14:29.55[TK]D-FenderKatty: Keep reading and looking at the pic....
14:30.03[TK]D-FenderKatty: Go caffeinate
14:30.18Kattyoh
14:30.20Kattyi didn't see the leg.
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14:30.29Kattyheh
14:30.32Kattygoes to get soda.
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14:33.18Kattyheh. on my way back from getting a soda, the receiptionist asked me if the company had a copy of 'microsoft' she could take home and load on her personal computer so she wouldn't have to go buy it.
14:33.29tzanger[TK]D-Fender: that's kind of what I am thinking
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14:34.39hi365Katty: I need a copy too. Y'know, for my ipod.
14:35.17Katty[TK]D-Fender: this is the same receiptionist who blogged Obama being the antichrist.
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14:37.06*** part/#asterisk gambler1 (n=mene_sun@dragan.eunet.yu)
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14:38.21[TK]D-FenderKatty: there's a great word for people like that : sombitch
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14:44.04Kattyhow incredibly annoying.
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14:46.51clintcplantseeker: after doing a little more digging this might be the answer to our problem: http://bugs.digium.com/view.php?id=12296
14:47.17clintccan someone tell me if these patches at http://bugs.digium.com/view.php?id=12296 have been merged into the 1.4.22 release?
14:47.30clintcor even just a pointer for figuring it out myself
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14:54.59kaldemarclintc: seem to be. 1.4.22 ChangeLog 2008-04-08 15:03 says the issue is closed.
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14:56.20clintckaldemar: yes, was just looking at revs in 1.4.22... but we are still having mixmonitor sync problems with recordings longer than 3 to 5 minutes or so
14:56.37disposabledoes anyone have an init.d startup script for asterisk 1.6 on debian? the one from 1.4 only starts astcanary. asterisk doesn't start but prints out Unable to open pid file '/var/run/asterisk.pid': Permission denied
14:56.39*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
14:57.26Kattyhugs anthm
15:02.46*** join/#asterisk dynamite-- (n=patrick@212.112.255.47)
15:03.12dynamite--hey all. im running asterisk 1.4.18 and want to upgrade to the last stable version of asterisk. how do i proceed?
15:03.45[TK]D-Fenderdisposable: thats not a lack of init script error, thats a file permissions.  Check what user you're running as and who owns the PID
15:03.53[TK]D-Fenderdynamite--: Just compile over
15:04.01[TK]D-Fenderdynamite--: (withing 1.4 series)
15:04.08*** join/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net)
15:04.09dynamite--okay
15:04.13fcois93hello all
15:04.14dynamite--simply get new code
15:04.20dynamite--recompile and reinstall?
15:04.32[TK]D-Fenderdynamite--: Yes.  As long as you don't do "make samples" you'll be fine
15:04.39fcois93I need to know how asterisk can forward some headers ?
15:04.39dynamite--right, cheers..
15:06.14*** join/#asterisk jer (n=jer@unaffiliated/jer)
15:06.14tzangerhmm I'm pretty sure the PIIX4 is PCI 2.1 compliant
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15:08.23hi365im having a weird situation where queues is showing a member as avalible, while in truth he's not
15:08.27IsUphi ya
15:08.50hi365there seems to be some bugs about this, but I think I applyed all the lates patches - and its still not reporting the proper state
15:09.00stintel\
15:09.29IsUpcan anyone explain how can i use G option in Dial? i am trying to transferring caller and called party to extension and want to bridge calls. but call is dropping after execution of dialplan.
15:09.34IsUpany ideas?..
15:11.21guaxIsUp: show application Dial
15:11.32guaxooops
15:11.44guaxwrong anderstand of the question sorry
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15:12.15IsUpi did
15:12.32plantseekerThanks clintz
15:12.40IsUpbut call is not bridging, its dropping after execution. and
15:12.49disposable[TK]D-Fender: i've added user asterisk, group asterisk, chowned /etc/asterisk to asterisk:asterisk, i'm using the init script from 1.4 (on 1.6). astcanary is started under root, and i can start 'asterisk' under root with no problem. only when i do it with the startup script, i get the permissions error. how do i start it as asterisk?
15:12.57IsUpits just execute dialplan for 'called' party, i cant set any variables to caller
15:13.09guaxIsUp: why you need to transfeer them?
15:13.10[TK]D-FenderIsUp: That option does not bridge after
15:13.23IsUpwell, let me explain my problem.
15:13.56IsUpfor example, i am sending a call with Dial at 14:01, but call is answered on 14:03, and caller is hangs up. how can i get actual 'answered' time?
15:14.03*** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
15:14.44IsUpso theres 2 mins lost, on connection state, on ringing etc... i want to know exact time when channel answered.
15:15.02guaxIsUp: theres a variable ANSWEREDTIME
15:15.04IsUpand i am using AGI, i want to save this information to my database. but i cant read a dead channel.
15:15.25guaxyou can read a dead channel on DeadAGI call
15:15.28lmadsenIsUp: DeadAGI()?
15:15.39IsUpyeah, i am using DeadAGI
15:15.54jameswfokay good news and bad news
15:15.59jameswfNortel Cuts 1,300 Jobs and Lowers Its Outlook  <----good
15:15.59IsUpwhen caller hangs up, i need the actual answered time =)
15:16.18guaxIsUp: after dial get the variable ANSWEREDTIME
15:16.40jameswfGm SUV production plant working over time to keep up with demand after drop in gas prices <--bad... god america is stupid
15:17.04*** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-0-69.phlapa.east.verizon.net)
15:17.27IsUplet me tell if its working.
15:17.28coppicejameswf: there is nothing good about nortel having problems. it means the entire telecoms outlook is gloomy
15:17.32*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
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15:19.08jameswfThe proprietary telecom industry going down means people are looking for cheaper alternitives.... in flys asterisk with tights and a cape
15:19.36coppiceat times like thing it generally means people are not looking for anything at all
15:19.38jameswfCircuit city is bankrupt that kinda sucks
15:19.48phpboy:(
15:19.55jameswfI dont shop there but still sucks...
15:19.57phpboythis is happening with most companies these days :T
15:20.14jameswfno more state of the art service
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15:23.10jameswfthats funny apple and rim battle it out to be second to nokia
15:23.19IsUpyeah, i can get with ANSWEREDTIME but, how can i get it without using DeadAGI?..
15:24.48[TK]D-FenderIsUp: same way you would normally.
15:25.04*** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com)
15:25.33IsUpbut when i am hangup it does nothing. and i am using AGI script.
15:26.38mark_csiHello, anyone know if you can buy hardware echo cancellation after you've already got a digium analogue card?
15:26.52hi365belives so
15:30.15mark_csiThanks hi365 - just found it.  VPMADTO32
15:30.22[TK]D-FenderIsUp: Clrealy should be using DeadAGI
15:30.29disposablels
15:30.39disposablesorrywrong window
15:31.29*** join/#asterisk Bad_Robot- (n=BadRobot@cpe-76-173-219-25.socal.res.rr.com)
15:35.57mark_csiexit
15:36.01*** part/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com)
15:36.09*** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com)
15:36.21mark_csihehe wrong window
15:36.23*** part/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com)
15:36.52IsUplol
15:36.54*** join/#asterisk mbranca (n=matteo@93-62-230-243.ip24.fastwebnet.it)
15:36.57IsUpLOL
15:40.55*** join/#asterisk sergee (n=serg@voip1.west-call.com)
15:44.52*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)
15:54.46getsql08silly question but how to i start asterisk
15:54.53getsql08via the CMD prompt
15:55.02*** join/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com)
15:55.31IsUpCMD prompt?
15:55.49getsql08command prompt
15:55.51getsql08in linux
15:56.05getsql08IE /usr/sbin/asterisk start?
15:56.20[TK]D-Fendergetsql08: "asterisk -gvvvvvvvc"
15:56.34[TK]D-Fendergetsql08: if * is not already running as a daemon
15:56.55[TK]D-Fendergetsql08: Which is normally the case.  At which point you jsut access CLI via "asterisk -r"
15:57.39getsql08[Nov 10 09:57:04] WARNING[6478]: manager.c:3159 init_manager: Unable to bind socket: Address already in use
15:57.58[TK]D-Fendergetsql08: Looks like * is already running.
15:58.01[TK]D-Fendergetsql08: try conencting to it.
15:58.14*** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU)
15:58.56getsql08[Nov 10 09:58:34] WARNING[6487]: pbx.c:4693 add_pri: Unable to register extension 'abn', priority 3 in 'gdincoming', already in use
15:59.00getsql08how
15:59.06getsql08unable to register add_pri EXT
15:59.15[TK]D-Fendergetsql08: Go look at your dialplan.  You've clearly duplicated things
15:59.33[TK]D-Fendergetsql08: its telling you to your face exactly what.
16:01.04getsql08how about this
16:01.05getsql08<PROTECTED>
16:01.05getsql08* make sure you read the INSTALL doc and apply the one of the following patches       *
16:01.05getsql08*        channel.c.hangup_callerid_ast12.diff                                         *
16:01.05getsql08*        channel.c.hangup_callerid_ast14.diff
16:01.13getsql08how do i apply these patches
16:01.30[TK]D-Fender~pb
16:01.31jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
16:01.32*** join/#asterisk gazzerh (n=garryh@host-212-158-227-149.dslgb.com)
16:01.32[TK]D-Fender^^^^
16:01.38[TK]D-Fendergetsql08: do not spam in here please.
16:01.59[TK]D-Fendergetsql08: And where did you see that?
16:03.42*** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com)
16:04.31getsql08[TK]D-Fender: one sec.. i am back to my dial plan
16:04.36getsql08i have clearly duplicated things
16:04.41getsql08but I do not see any duplicate entries
16:04.45getsql08in manager.conf
16:04.48*** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com)
16:04.58[TK]D-Fendergetsql08: that isn't the dialplan <-
16:07.45*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
16:07.49*** part/#asterisk SteelSide (n=SteelSid@217.76.87.68)
16:13.14[TK]D-FenderBRB, rebooting.
16:15.15*** join/#asterisk nikko (n=nikko@69.57.49.100)
16:16.00ManxPowerBTW, why do you have an extension called "abn">
16:16.02ManxPower?
16:16.52*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
16:17.16*** join/#asterisk l2trace99 (n=jr@static-71-251-65-16.tampfl.fios.verizon.net)
16:17.25*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
16:21.36IsUpbrb reboot
16:22.03Kattyponders lunch
16:22.05*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
16:22.11KattyZEEEK!
16:22.15Zeeekhey there ho there ho there
16:22.15Kattyhugs Zeeek
16:22.28Zeeekair kisses Katty
16:22.48ZeeekLadies and gentlement, for the first time ever on this channel
16:22.57ZeeekI HAVE AN asterisk QUESTION
16:23.00Kattyomg
16:23.01Kattyfaitns
16:23.04Kattyoh.
16:23.06*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
16:23.07Kattylet's try that again.
16:23.09Kattyfaints.
16:23.14Bad_Robot-heh
16:23.15Zeeekwhere does SipAddHeader() go in the dial plan? Right beofre the dial() ?
16:23.18*** join/#asterisk keulin (n=cray@bne75-5-82-231-224-34.fbx.proxad.net)
16:23.19Kattyyes.
16:23.23[TK]D-FenderZeeek: Any time before
16:23.30Kattyi can pastebin something, if you need it.
16:23.44Zeeekyou mean like in another file? </joke>
16:24.03Zeeekno I think that's all I need
16:24.12[TK]D-FenderZeeek: Sure... as long as its executed before the dial in the call you want it to apply to :)
16:24.32Zeeekso to call Talkshoe, apparently they've added a SIP "failsafe" way to defeat DTMF timing problems
16:24.37Kattyoh noes. i'm out of checks.
16:24.49Zeeekhow about balances? Out of those too?
16:25.10[TK]D-FenderKatty: Executive Branch may be right for you!
16:25.21ZeeekSo if I wanted to send a Subject header:
16:25.35ifnotwhynotwhere can i list of hangup causes on PRI for asterisk, got hangup cause 88 can't seem to google it right
16:25.52*** join/#asterisk sdaniels (n=chatzill@216.65.195.52)
16:26.07ZeeekSipAddHeader(Subject: <passocde>12345</passcode><pin>1</pin>)
16:26.12ZeeekCorrect?
16:26.13*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:26.18[TK]D-FenderKatty: If it makes you feen any better, MO is looking at a recount that you can hold over your receptionist :)
16:26.25[TK]D-FenderZeeek: Sure
16:26.34Zeeekno quotes or anything silly like that
16:26.40Katty[TK]D-Fender: i'm not going to even go there.
16:26.58Zeeekof course, there is an error in my call above
16:27.02[TK]D-Fenderifnotwhynot: http://www.google.ca/search?hl=en&q=isdn+code+88&btnG=Google+Search&meta=
16:27.18[TK]D-Fenderifnotwhynot: Funny I see a lot of answers on that first page without even DRILLING the links
16:27.36ZeeekDrill Baby Drill
16:27.40[TK]D-FenderISDN cause codes help you diagnose problems with calls. ... 88. Incompatible destination. 89. Non-existent abbreviated address entry ...
16:27.54Zeeekshudders at the memory evoked by that line
16:27.58*** join/#asterisk astrOdz (n=astrOdz@119.92.213.100)
16:28.02Kattyseanbright: LIAR
16:28.36seanbrighti is
16:28.56seanbrighti'll update to something more appropriate
16:28.59ifnotwhynotthx
16:29.06seanbrightthere.
16:29.30Kattyseanbright: that i can believe.
16:29.33Kattylights seanbright on fire.
16:29.42[TK]D-Fendergrabs some marshmallows
16:30.17[TK]D-FenderSeans-nuts roasting on an open fire!
16:30.22[TK]D-Fendercarols...
16:30.31Kattyhaha
16:30.33Kattyyou're awful
16:30.35Kattyalso!
16:30.35Bad_Robot-feels bad for sean
16:30.40Kattyseanbright: what is 'twitterBar'?
16:31.07ManxPowerThere's nothing wrong with twittering, as long as you do it in private and wash your hands after.
16:31.25Kattyseanbright: something for a browser?
16:31.48Kattyoooh. firefox addon. schnazzy.
16:32.39*** join/#asterisk Greek-Boy (n=email@41.222.89.77)
16:33.46*** join/#asterisk SwK (n=SwK@freeswitch/developer/swk)
16:33.59*** join/#asterisk CunningPike (n=arodgers@204.239.8.157)
16:36.12KattySwK: GET OUT
16:36.14Kattyhugs CunningPike
16:36.22Zeeekhttp://food4wine.ning.com/page/vuc-and-other-news
16:36.28Zeeekthanks everyone
16:36.42Zeeekair kisses Katty again
16:37.02Kattyi never really got air kisses.
16:37.03Qwelloh god, another sip extension?
16:37.05Kattythe point of them, that is.
16:37.14Zeeekwait, you're Americans. {{{{{{Katty}}}}}
16:37.27KattyZeeek: are you french?
16:37.37CunningPikehugs Katty right back
16:37.42ZeeekKatty: I am today
16:37.51*** join/#asterisk synchris (n=synchris@athedsl-86032.home.otenet.gr)
16:37.56ZeeekQwell: referring to my header?
16:38.16Kattyi'm lost. that's okay tho. i stay lost.
16:38.31ZeeekLost is good. Don't ever change
16:38.32KattyCunningPike: how're you today, deary?
16:38.49CunningPikeKatty: Excellent, thanks - et vous?
16:39.07ZeeekQwell: was that comment directed at I ?
16:39.16QwellZeeek: yes
16:39.18KattyCunningPike: hungry, but plotting lunch.
16:39.27ZeeekIn that case it deserves an answer
16:39.33CunningPikeKatty: Cool - just after breakfast here :)
16:39.35ZeeekIt isn't an "extension"
16:39.40KattyCunningPike: must get some groceries on the way home from work. I'm making pork chops, homemade mashed potatoes, and some sort of veggie for dinner.
16:39.57CunningPikeKatty: I'll be right over
16:40.04Kattyplans to make extra.
16:40.07CunningPike;)
16:40.17ZeeekTalkshoe's SIP client uses it to send "pseudo" callerID and conference number info
16:40.23KattyCunningPike: want a tip on the mashed potatoes?
16:40.30KattyCunningPike: used diced canned potatoes. already cooked.
16:40.36CunningPikeAh
16:40.43KattyCunningPike: just mash and mix with the milk and butter--insta homemade goodness!
16:40.53Kattycomplete with those mashed potato chunks.
16:41.09Zeeekso, the faithful participants of VUC who constantly complain about Inband/RFC DTMF won't need to anymore
16:41.15CunningPikeis Irish - not sure I could use canned spuds - it's against religion or something
16:41.16QwellZeeek: it's an extension
16:41.17Zeeekya see?
16:41.35ifnotwhynothi there i am lost,,  i am receiving a call to a PRI span 1 for some reason(http://pastebin.com/m261bb11d) i wan to forward the call to another pri span 2 connected to a legacy pbx.. any help welcome.
16:41.36KattyCunningPike: well i'm only half irish, so it's okay if i cheat
16:41.38putnopvutThat seems like an odd use of the Subject header
16:41.44CunningPikeKatty: ;)
16:41.49ZeeekQwell I prefer to see it as the opposite of "deprecated"
16:41.57Qwellputnopvut: probably violates some rfc :p
16:42.00Zeeekmaybe it's "reprecated"
16:42.04Qwelllike, "don't put meaningful crap in here"
16:42.24Zeeek"If it works, use it"™
16:42.35QwellYou just described SIP in 5 words.
16:42.42Zeeekindeed
16:42.50putnopvutQwell: Well, it's not really violating anything. But I would have expected some sort of X- header to be used.
16:43.07*** join/#asterisk italorossi (n=italoros@201.76.154.111.intranet.digi.com.br)
16:43.09Zeeekwhich is where logic forbidding masturbation falls short
16:43.25putnopvutLike Zeeek said, "if it works, use it" :)
16:43.35Zeeek"If I shouldn't do this, why is it there?"
16:43.52*** join/#asterisk fogo (n=Paul@69.169.132.35.provo.static.broadweave.net)
16:44.04KattyZeeek: then how do you explain shock collars :<
16:44.09ZeeekIf RFC is a "request" then who said I said it was ok to use? huh?
16:44.35ZeeekKatty:  tu m'intéresses! You're into that stuff? I wish I wasn't so far away
16:44.44ManxPowerifnotwhynot: Cause 88 means "incompatible destination"
16:44.52*** join/#asterisk pif (n=ldm@zenon.apartia.fr)
16:44.59KattyZeeek: erm.
16:45.06pifis it still possible to use zapte with 1.4.22 ?
16:45.10pifzaptel
16:45.15putnopvutpif: yes
16:45.24Zeeekbut it's desecrated
16:45.37ManxPowerifnotwhynot: But I have no idea where this message comes from "-- Requested transfer capability: 0x10 - 3K1AUDIO"
16:45.41Arsenick-Zeeek, t'es en amour ? ;)
16:45.45*** join/#asterisk Ropeguru (n=yeah@adsl-067-035-094-144.sip.mia.bellsouth.net)
16:45.51Zeeeknon, en chaleur
16:45.59Arsenick-haha
16:46.04Zeeekà mon âge avancé
16:46.05Kattyoh wow. i just read that obama's chief of staff wants people through the age of 18 to 25 to serve in the military
16:46.12Qwellyay for being 26
16:46.13tzafrir_laptopZeeek, I guess you read the SIP 4.0 RFC
16:46.22KattyQwell: i need another year STAT
16:46.22ZeeekGreat idea. Cuts way down on anyone wanting to approve of wars
16:46.26ManxPowerKatty: change.gov is their new web site
16:46.31putnopvuttzafrir_laptop: that is the best RFC
16:46.32pifwhere is the last version of 1.4.x zaptel?
16:46.35KattyManxPower: this makes me want to strangle someone.
16:46.40Qwellpif: same place as the version before that
16:46.47pifcan't get it from digium
16:46.49ManxPowerKatty: Maybe so, but I bet we won't have as many wars
16:46.51[TK]D-Fenderpif: in the channel topic
16:46.51Qwellpif: why not?
16:47.01KattyManxPower: not if it's manditory.
16:47.10KattyManxPower: there will be an uprising in the religious communities tho.
16:47.20Zeeekyeah right
16:47.32KattyManxPower: Jehovah's Witnesses would rather go to jail than serve in the military
16:47.37ManxPowerKatty: Sure it will.  It's one thing for people to volunteer -- if they want to be killers that's up to them, but when people are drafted their parents get pretty upset
16:47.47Kattynods
16:47.49[TK]D-Fenderpif: http://downloads.digium.com/pub/zaptel/ <- sure as heck works for me.
16:47.56Zeeekand therefore war becomes very unpopular
16:48.01ManxPowerKatty: there are and have always been options for people who object to being drafted.
16:48.08pif[TK]D-Fender: thx
16:48.11ZeeekIn fact, it has been said the ending the raft was a way to stiffle protest
16:48.26Zeeekwhen you think about that, it makes a little more sense
16:48.26[TK]D-Fenderpif: Might want to try a tiny hit harder next time.
16:48.35KattyManxPower: never read about them. growing up as Jehovah's witness, and a female, has a way of turning you against all things military.
16:48.47ManxPowerKatty: Personally I don't like the idea of required military service -- but many, many, man countries around the world have it.
16:48.48pifkisses [TK]D-Fender 's feet
16:49.29KattyUnited States of <strike>America</strike> SPARTA!
16:49.59ZeeekI may be the only person here who has served in a US branch of milserv?
16:50.13KattyZeeek: my fiance served 10 years
16:50.14ManxPowerZeeek: There's a BIG difference between "Mom, Dad, this is what I want to do!  <put in reasons here> and I'm 18 and you can't stop me" and "I don't want to go to war Mom and Dad!  I'm scared! "
16:50.22[TK]D-FenderManxPower: You wouldn't mind it as much if not for who's running it :)
16:50.25ZeeekBUT HE's NOT HERE. NANANANANA
16:50.36Kattythank goodness ;)
16:50.37pifSparta didn't have mixed breeds as leaders
16:50.46pifhence their demise
16:51.34*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
16:51.55jblackThe US free market economy approach to waging war has been an interesting experiment.
16:51.56ZeeekWell, I got an asterisk answer here, possibly the first in the last three years. We now return you to an endless sequence of OT comments. THANK YOU ALL
16:52.10Kattyi live for OT
16:52.11WHYSOF the people I know, the ones not going to war were the brave ones.  There is a fine line between brave and stupid.
16:52.17ManxPowerjblack: It worked much worse than I expected.
16:52.19ZeeekKatty: me too. Asterisk is boring
16:53.01jblackOh?
16:53.39WHYSNot that they are the only two reasons for going to war.  I hear "adventure" is a big one.
16:53.44pifiraq, vietnam are not real wars, only bullying of a weaker nation that turned into quagmires
16:53.53WHYSa hollow voice says "plugh"
16:53.57fcois93I need to know how asterisk can forward some headers ?
16:54.00ManxPowerjblack: I felt that an all volunteer army was a good thing.  The problem with my idea is that there is much less resistance to war when the only people getting killed WANT to be there.
16:54.35*** join/#asterisk Toerkeium (n=Miranda@201.216.206.221)
16:54.43jblackOhhh, so it's not the direct results, but the implications. That's a very interesting point.
16:54.50WHYSManxPower, on our side anyway.
16:54.51ManxPowerfcois93: Asterisk is a B2BUA SIP device, it does not forward ANY headers.
16:55.01Toerkeiumguys, "show g729" shows "0/2 encoders/decoders of 10 licensed channels are currently in use". How/what can I check who is using those 2 decoders?
16:55.08ZeeekManxPower: unfortunately it lowers the quality both of the army and of the experience gained. (not including war, which is awful)
16:55.39ZeeekAnyway, we should stop attacking shit for oil and solve the problmes more intelligently
16:55.42jblackZeeek: Do you really think that? I think quite the opposite.
16:55.44Zeeekand there I meave you
16:55.54ZeeekI meave you bvery much
16:56.04ManxPowerZeeek: IF they started drafting congrespeople's and senator's kids this war would be over by now.
16:56.07*** join/#asterisk Defraz (n=T0tal@63.228.246.229)
16:56.10Zeeekjblack: what that war is awful?
16:56.11fcois93ManxPower: Is it possible to imagin a solution for that?
16:56.20jblackBut they never draft congress' kids.
16:56.21ZeeekManxPower:  so we seem to agree?
16:56.30Zeeekbye for now
16:56.45Zeeekand thanks for all the phish
16:56.48*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
16:57.02ManxPowerjblack: I would have to find a source to cite, but if Elvis was drafted, I doubt politician's kids were exempt.
16:57.10Carlos_PHXThe real solution to unpopular wars is to eliminate the government's ability to steal our money to fund it.  If they had to ask us to vote to pay for a war, it wouldn't happen unless it was truly necessary.
16:57.24WHYSLook up draft board 100
16:57.34coppiceElvis and Mohammed Ali were drafted as show pieces
16:57.37WHYSIt's where important peoples kids went.
16:57.50Kattyoff i go. cheerio.
16:58.11jblackCongressional children, unlike 85% of the population, usually go to college. There's a draft exception for that.
16:58.59jblackI don't think elvis is the example you mean it to be. Didn't the folks in congress dislike elvis, socially speaking, due to his gyrating hips?
16:59.09coppicepeople like Elvis ended up entertaining the troups, and were always kept away from danger. morale would have suffered if they were hurt
17:00.08jblackLook at george bush. Didn't he serve in the national guard, and weren't there questions as to whether he actually bothered to show up?
17:00.37coppiceno. no. nobody dares to even ask such questions :-)
17:00.53jblackAt leat not twice, right? :)
17:02.39*** join/#asterisk devilsoulblack (n=aandaluz@srv.ec-gye.internet.geainternacional.com)
17:02.52*** join/#asterisk chebyte (n=chebyte@host172.190-225-228.telecom.net.ar)
17:02.57chebytehi people
17:03.00chebytei have a question
17:03.12Carlos_PHXWe probably have an answer, want to see if they match?
17:03.20chebytei need buy a board asterisk with support for 4 lines
17:03.20jblackThat's good. Everyone should have a question.
17:03.20pifit better be about politics
17:03.31chebyteanyone can recommend me?
17:03.43*** join/#asterisk keith4_ (n=keith@lust.CC.Lehigh.EDU)
17:03.46jblackPOTS ?
17:04.07pifdidn't Dali say "computer are boring, they only have answers" ?
17:04.14chebyteI ' am viewing a openvox a400p
17:04.17ManxPowerchebyte: 4 of what kind of ports?
17:04.25chebyteo digium tdm400p
17:04.33ManxPowerchebyte: if you go with OpenVox almost nobody here will be able to help you.
17:04.36chebytefxo
17:05.29chebyteManxPower: ok what recommend me?
17:05.49pifcoppice: are you still contributing to callweaver?
17:06.01*** join/#asterisk djin (n=djin@84-104-110-116.cable.quicknet.nl)
17:06.08devilsoulblackhi any one have exprience with openr2
17:06.59*** join/#asterisk bijit (n=benji@190.241.15.48)
17:07.38coppicepif: yes
17:07.41chebytewhat kind of board is good ?
17:07.45*** join/#asterisk aiksa[LV] (n=root@mx.fiveplus.lv)
17:07.53aiksa[LV]beep, tzafrir_laptop there?
17:08.01pifcoppice: so the project is still alive and well ?
17:08.33*** join/#asterisk bijit (n=benji@190.241.15.48)
17:08.48tzafrir_laptopaiksa[LV], yes
17:08.52coppiceyes
17:09.08*** part/#asterisk WHYS (i=lpfm@137.28.94.209)
17:09.52devilsoulblacktry to make and make install over openr2 and get "make: *** No targets specified and no makefile found.  Stop."
17:10.18*** part/#asterisk mark_csi (n=csiadmin@host217-41-18-3.in-addr.btopenworld.com)
17:12.42*** join/#asterisk erth64net (n=erth64ne@mail.opensourcery.com)
17:13.18*** part/#asterisk fcois93 (n=Administ@bagnolet.acropolistelecom.net)
17:13.30*** join/#asterisk spokra (n=spokra@blockhead.sea0.speakeasy.net)
17:15.02tzafrir_laptopdevilsoulblack, looks like you need to run ./configure
17:15.12*** join/#asterisk wiscados (n=mint@81.25.184.155)
17:17.09*** join/#asterisk l2trace99 (n=asd@static-71-251-65-16.tampfl.fios.verizon.net)
17:18.13*** join/#asterisk lejun (n=lejunhu@66.178.134.235)
17:21.03[TK]D-Fenderchebyte: Sangoma A200d
17:22.13*** join/#asterisk Spydre (n=nbaker@75.147.255.17)
17:23.11ManxPowertzafrir_laptop: So basically he needs to read the INSTALL or README file.
17:23.43SpydreAnyone around that can offer some advice on videoconferencing?
17:23.54ManxPowerI must admit Digium putting almost all the docs in TeX format almost guarantees nobody will read them.  Great going Digium!
17:24.16QwellManxPower: RTFPDF
17:25.04ManxPower[root@bourbon asterisk-1.6.0.1]# find . -name "channelvariables*" -print
17:25.05ManxPower./doc/tex/channelvariables.tex
17:25.05ManxPower[root@bourbon asterisk-1.6.0.1]#
17:25.09ManxPowerWhere was that PDF again?
17:25.50[TK]D-FenderManxPower: Yeah add me to the picketer's list...
17:26.02tzafrir_laptopManxPower, the README has nothing about it. The INSTALL is the generic autotools one
17:26.13*** join/#asterisk rwaite (n=richard@rrcs-74-218-125-86.central.biz.rr.com)
17:26.14*** join/#asterisk freakazoid0223 (n=mattc@pool-72-81-0-69.phlapa.east.verizon.net)
17:26.33hardwireKeizer: hows things?
17:26.37ManxPower[TK]D-Fender: They must have been drunk or high when they made that decision.
17:26.41hardwirerolls Keizer
17:27.14SpydreIs there any active development of video conferencing solutions for asterisk?  I've tried app_conference but that doesn't seem very complete, and hasn't been updated lately.
17:27.50*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
17:28.05*** join/#asterisk CrazyTux (n=brandon@user-vcauh95.dsl.mindspring.com)
17:28.15ManxPower[TK]D-Fender: The decision will make ME stop telling people to look at the docs included in the tarball.
17:28.45ManxPowertzafrir_laptop: and INSTALL doesn't tell you to run ./configure?
17:29.00*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
17:29.24*** join/#asterisk boolean12 (n=boolean1@tandem.uplinktel.com)
17:29.52ManxPowerPeople complain about us being "unix elitists"  No better elitism than making all the poor sods figure out how to read TeX formatted files.
17:30.02tzafrir_laptopit does. Though it could have been a bit clearer
17:31.01ManxPower~mailinglist
17:31.02jbot[~mailinglist] The Asterisk mailing lists can be found at http://lists.digium.com , http://www.asteriskguru.com/archives Search the archives by adding "site:lists.digium.com" to your Google search.
17:35.41[TK]D-FenderManxPower: I still haven't figured out how...
17:38.26ManxPower[TK]D-Fender: Maybe the marketing people said "The docs are too easy to find!  Lets make it harder!"
17:38.28*** join/#asterisk grantm (n=grant@68.142.138.4)
17:38.45[TK]D-FenderManxPower: What does TeX really add to our lives?
17:38.57ManxPower[TK]D-Fender: maybe for 1.8 they will require EMACS in order to edit the config files!
17:39.13*** join/#asterisk Wayhigh (i=wayhigh@www.kevinlynn.com)
17:39.21*** join/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
17:39.32ZeeekI'm back with question #2! 2 in one day
17:39.47ManxPower[TK]D-Fender: it's a vendor/platform independent document format.  Much like oh say HTML or TEXT.
17:39.49ZeeekHow can I find out who this number belongs to (provider): Call 1-800-949-1284
17:40.15ZeeekI just got a domain scam and they want me to call Call 1-800-949-1284 and pay $75 for one name
17:40.39ZeeekI encourage you to get them tazlking, they must be real scumbags to do that
17:40.47[TK]D-FenderManxPower: I mean it isn't any more searchable or anything is it?  No in-lin pics or anything fo value?
17:41.09ManxPowerZeeek: http://www.google.com/search?hl=en&q=800-949-1284&btnG=Google+Search&aq=f&oq=
17:41.21ManxPower[TK]D-Fender: It's a desktop publishing format.
17:42.09ManxPowerformatting, pictures, generated table of contents, citations, everything you need for desktop publishing.
17:42.51[TK]D-FenderManxPower: And how much of that did we USE in converting to it?
17:44.04ManxPower[TK]D-Fender: No idea, as I'm not going to install 100MB of TeX and required libraries in order to find out.
17:45.19ManxPower[TK]D-Fender: The thing that really annoys me is that SOME of the docs are still in text format.
17:45.30ManxPowerjust not the ones I was looking for.
17:46.19*** join/#asterisk kotique (n=picachu@host-static-89-41-72-85.moldtelecom.md)
17:46.27ManxPowerI predict that UPGRADE.txt will become UPGRADE.tex before the next relase.
17:46.31kotiqueguys. do you know any good graphical SIP proto analyzer ?
17:46.37ManxPowerThat will raise the barrier even more.
17:46.39kotiquesomething like this http://bugs.digium.com/file_download.php?file_id=19827&type=bug
17:47.09magronezis back
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17:52.12pifis that a zaptel.conf problem? "ZT_SPANCONFIG failed on span 1: Invalid argument (22"
17:52.21[TK]D-Fenderpif: Clearly
17:52.41pifI just upgraded from zaptel versions without changing the conf
17:53.01pif1.4.11 to 1.4.12.1
17:53.26ManxPowerpif: did you also upgrade your kernel?
17:53.30pifyep
17:53.51ManxPowerpif: Did you recompiled zap after upgrading the kernel?
17:53.55pifwct4xxp loads
17:54.02ManxPowerThat was not my question
17:54.08pifyes, after reboot to new kernel
17:54.31ManxPowerpif: Odd.
17:55.00pifdidn't do distclean
17:55.09pifleftovers from older kernel?
17:55.22ManxPowerno idea
17:55.26piftrying
17:55.28ManxPowerdid you at least do a make clean?
17:55.38pifyep and it failed
17:55.44ManxPowerfailed?
17:55.47*** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net)
17:55.53ManxPoweras in error message or as in "no rule to make clean"
17:56.32l2trace99does anyone know how do bridge the channels once in attendend transfer  ?
17:56.41[TK]D-Fenderpif: Did you upgrade * as well?
17:56.49pifyes, 1.4.22
17:57.01[TK]D-Fenderl2trace99: huh?
17:57.02ManxPowerl2trace99: however your phone docs say to do it.
17:57.22l2trace99it is a function of the phone  or *  ?
17:57.26*** join/#asterisk bbryant (n=brett@68.208.65.50)
17:57.29[TK]D-Fenderl2trace99: How are you doing it?
17:57.47l2trace99using atxfer in features
17:58.00ManxPowerl2trace99: That depends on the kind of transfer.  I recommend using phone based transfers rather than the ugly DTMF transfer hack people seem so fond of.
17:58.42l2trace99know any good softphones that allow for a warm tranfer ?
17:58.47pifis ztdummy supposed to load with wct4xxp ?
17:58.53pifdoes it interfere ?
17:58.53ManxPowerpif: no
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17:59.07Daejeoanyone have a iphone?
18:00.23[TK]D-Fenderpif: No, you should not load ztdummy.
18:00.41pifwithout it ztcfg completes
18:01.18pifbut I only have red alarms :(
18:01.28*** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com)
18:02.28pifwith 1.4.11 they were green
18:02.45Zeeekhow cool would it be if IRC had tags like the web 2.0 stuff?
18:02.58ZeeekHow large would ZTDUMMY be in this channel?
18:03.01ZeeekHIGE
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18:14.56Kattyah, screen, how i love thee.
18:14.57*** join/#asterisk Segnale007 (n=Pietro@host36-253-dynamic.36-79-r.retail.telecomitalia.it)
18:15.16pifdo alarms stay RED when asterisk is stopped?
18:17.02ManxPowerpif: try it and see
18:17.34pifthey stay RED with 1.4.22, but go to GREEN with 1.4.21.2
18:17.56DaejeoKatty: :) meow meow
18:19.22KattyDaejeo: murrrrrrrrow
18:19.22*** join/#asterisk Carlos_PHX (n=Carlos@64.129.235.200)
18:19.22Kattyherroes Carlos.
18:20.03DaejeoKattyt Meowing and What it Means
18:20.04Carlos_PHX'morning
18:21.00Carlos_PHXnotices new political discussion on file formats.
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18:22.55KattyDaejeo: mewtacular. http://www.youtube.com/watch?v=6HMIRDEYoEI
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18:24.20DaejeoKatty: beautiful :)
18:24.28pifget a room
18:24.33Kattyquite.
18:24.53Kattypif: play nice, or get out (=
18:26.10KattyCarlos_PHX: did you hear about Obama's chief of staff wanting to put all 18-25 year olds into active duty?
18:26.27Carlos_PHXwonders if I'm being set up for a joke
18:26.44jameswfis over 25 YAY
18:26.47Carlos_PHXSurely, no democrat would be such a fascist, no?
18:26.56KattyCarlos_PHX: REF: http://www.examiner.com/x-536-Civil-Liberties-Examiner~y2008m11d6-Obamas-chief-of-staff-choice-favors-compulsory-universal-service
18:27.00Zeeekok gad, still on that?
18:27.21jameswfexaminer == enquirer?
18:27.30Carlos_PHXHoly shit, they've already started.  I'm Cuban, and everything about the Obama campaign reminds me of another socialist whose rule I lived under.
18:27.31pifuniversal service is _not_ military service and is an excellent idea
18:27.33De_MonI heard he's going to review all of the executive orders made by the current president and recend them all as his first act in office too.
18:27.46pifKatty: stop casting false aspersions
18:27.47*** join/#asterisk rene- (n=renemend@200.34.66.137)
18:27.50rene-hey guys
18:27.50[TK]D-FenderDe_Mon: "looking at".  We'll see.
18:27.56KattyCarlos_PHX: it's basically 3 months of service.
18:28.05KattyCarlos_PHX: from the way i understand the news article.
18:28.10Carlos_PHXYes, that's how it started in Cuba
18:28.12pifso what big whoop
18:28.14De_Monyou can hear all sorts of stuff, I'll wait till he actually "does" something
18:28.16Carlos_PHXMy dad did 4.5 years.
18:28.20jameswfI think the US should have manditory service like all other countries
18:28.24KattyCarlos_PHX: eeesha.
18:28.39Zeeekbye again :)
18:28.41*** part/#asterisk Zeeek (n=Zeeek@pdpc/supporter/active/Zeeek)
18:28.44KattyCarlos_PHX: i can't even begin to imagine the ammount of religious uprising about all this.
18:28.52Carlos_PHXI think the US should have filtered speech, it works for China.
18:29.02jameswfAmerica should also require a foriegn language like all other countries (not spanish)
18:29.06De_Monbut speaking of hearing things, someone told me at lunch the credit card companies are raising interest rates to something like 20%
18:29.14De_Monstarts googling
18:29.16Kattyjameswf: i agree with the foreign language.
18:29.30De_Monjameswf ahh hahaha not spanish?
18:29.33rene-where should i be looking at when calls  from a VOIP provider drop at random times but not all the calls drop at once, it looks totally random i dont think it is an audio issue since quality is very good
18:29.44Carlos_PHXI agree everyone should know it, but that's very different from abrogating liberty to force people to learn.
18:29.48Carlos_PHXWhich is just retarded.
18:30.04KattySame with military service.
18:30.19jameswfspanish would lean towards having two national langues like canada which would be completely stupid but you should have to pick one and learn it
18:30.35Carlos_PHXchecks to see if he still lives in "the land of the free."
18:30.36pif"liberty" is a red herring to keep ppl into submission, freedom is often obtained through compulsion
18:30.40jameswfI went to a private school foe k-6 and we did french
18:30.43KattyThe concept is okay, but quite another when Mommy Dearest has her child taken away.
18:31.02KattyCarlos_PHX: heh.
18:31.07KattyCarlos_PHX: not since the patriot act :/
18:31.11De_Monjameswf guys or girls?
18:31.12jameswfCarlos_PHX: Freedom is no excuse for ignorance
18:31.13Carlos_PHXI learned French (already knew Spanish), but got stuck at that whole white flag thing.
18:31.33Carlos_PHXFreedom includes the freedom to choose ignorance.
18:31.40pifwhat white flag?
18:31.48Carlos_PHXIf you can't choose what you put in your brain, you are not free.
18:31.53De_Monpif surrender?
18:32.03pifwho surrendered?
18:32.09jameswfCarlos_PHX: On that line of thinking we should eliminate all warning labels so those who choose to be stupid dont last long :)
18:32.19Carlos_PHXI fully agree.
18:32.24Carlos_PHXDarwin
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18:32.44jameswfI totaly see America turning it to idiocracy (the movie)
18:33.20Carlos_PHXThat movie was a view of the future.
18:33.30Kattyi'm picturing more civil war.
18:33.31De_Monwhite flag.. nevermind
18:34.13Carlos_PHXI mean, Idiocracy pretty much predicted Youtube.
18:34.28Kattydon't think i've ever seen the movie.
18:34.37De_Monme either
18:34.50Carlos_PHXRecommended, it's funny.
18:34.50Kattywell there we go, somethign to do this week (=
18:35.01De_Monsounds like a religulous and expelled and the like though
18:35.03jameswfObama won a popularity contest and like most winners of American Idol people will be like "Who?" in 3 months
18:35.15KattyTerry Fator!
18:35.18KattyPaul Pots!
18:35.23Kattyor was that Britian's got talent?
18:36.02Kattythose two guys have skills (=
18:36.51De_MonI was watching comedy central last night and the guy on stage said the same thing. "america doesn't vote for president, but they do vote for american idol" they should come up with a similiar show for electing presidents to get the public involved he went on to say... funny, yet very sad.
18:37.03jameswfI was floored Friday i think it was  America is going to a bad place in a hand basket, Obama lays out his plan to fix it then opens up for questions where they are like ummmmmokay thats awesome but what america really want to know is what Dog are you getting
18:37.30pifjameswf: everybody needs lighter moments, you know
18:37.44pifit's not a sign of stupidity to have them
18:37.46De_Monhas he picked a dog yet?
18:37.48sdanielsHow can I set * to send all the info that I usually see in an asterisk -r window to syslog?
18:37.58De_Monsdaniels edit logging.conf
18:38.00jameswfhe wants a Mut like him
18:38.00KattyDe_Mon: i believe they are adopting a puppy
18:38.05De_Monlogger.conf?
18:38.16De_MonKatty but what Kiiiind of puppy
18:38.31KattyDe_Mon: Mix breed, i'd presume.
18:38.41KattyDe_Mon: Peeing on American History.
18:38.48De_Monhahaha
18:38.56Kattythat should be a movie.
18:38.58jameswfunless he is going to sell the dog to the chinese for sweet and sour and make cash to fix things I DONT CARE ABOUT HIS DOG  :)
18:38.58De_MonI thouth it was because he was of mixed race
18:39.21sdanielsDe_Mon: Found it, thanks.
18:39.28Kattyjameswf: i'm glad he's getting a dog.
18:39.30De_Monthe first african american president has the firs dog of a mixed breed!
18:39.33Kattyjameswf: those children are going to have a very hard life.
18:39.42Kattyjameswf: the dog will be a comfort to them.
18:39.53pifaw, cry me a fucking river
18:39.56jameswfObama should perpitrate the sterio type and get a pit bull
18:39.57pifpoor kids
18:40.05Kattyjameswf: haha.
18:40.19jameswfKatty: he could save his kids and quit
18:40.34De_Monthat would put biden in charge, right?
18:40.47*** join/#asterisk lucky|aba (n=lucky@ip70-177-7-204.sb.sd.cox.net)
18:40.51Kattymaybe we could put mister paul in office.
18:41.02jameswfIs bidens daughter a lesbian... oh wait that was cheney
18:41.05Kattyit's nice to think about, anyway.
18:41.14[TK]D-Fenderone of the few dislikes I have with Biden is some of his tech stance, esp on net neutrality.  Obama = for, Biden = against
18:41.51*** join/#asterisk ChkDigit (n=mike@static24-72-71-175.regina.accesscomm.ca)
18:43.02ManxPowerTime will tell.
18:44.09jameswfwaits for email postage
18:44.11KattyHow depressing. Time for a new OT.
18:44.22[TK]D-Fendercranks up "The Best of Asia"
18:45.04RypPnpr0n-time again tk, yum
18:45.23De_MonO_o
18:45.40[TK]D-Fenderbacks away slowly....
18:45.49RypPnnow we know where the lumpy thumb came from
18:46.24[TK]D-FenderRypPn: came from a nasty sword accident
18:46.35jameswfthe government should tax internet porn.... no more $$$ worries
18:47.38De_Monwhy not just tax the internet!
18:47.59jameswfwell taxing porn taxes 98% or the net
18:49.36jameswfhttp://www.examiner.com/x-668-TV-Examiner~y2008m11d10-Fox-Orders-Full-Season-Of-The-Cleveland-Show <<Cleveland has his own show...
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18:52.45De_Monyada yada yada king of the kill got the axe too
18:52.50rgsteele||workHey folks.  Is there a way to force a particular sip peer to re-register?  I've got a phone acting up, and I'd like to be able to handle that server-side.
18:52.57rgsteele||workInstead of having the user power cycle the phone.
18:53.35De_Moni didn't think you could "request" a registration from the server
18:53.53*** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-1ad46cafb405ae51)
18:53.57*** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir)
18:54.16De_Monthe phone should have a registration timeout and it'll just reregister when it expires
18:54.42rgsteele||workDe_Mon: No way to pre-empt the timeout?
18:55.33rene-what could cause randomly dropped sip calls not all calls from the trunk drop at the same time and not all of them drop at all. audio quality is very good
18:56.49De_Monnone that I've ever heard of
18:57.59*** join/#asterisk killfill (n=killfill@200.63.96.244)
18:58.03killfillhey...
18:58.05hardwirepoor fill
18:58.15killfilldoes zopier support g729 in iax?...
18:58.26killfillcannot find the codec.. :S
19:00.53killfillah..it requires to pay.. :P
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19:04.28zchaosif i'm getting bad sound quality with my voip phone is it due to a poor internet connection?
19:05.19rene-yes
19:05.30zchaosit was so choppy
19:05.31Kattypossibly.
19:05.33zchaosand cutting out
19:05.51rene-probably you dont have enough upstream bandwidth
19:06.01Katty[TK]D-Fender: what is your opinon of this? http://www.newegg.com/Product/Product.aspx?Item=N82E16830120239
19:06.50zchaoswhat is the min upstream required
19:07.32rene-it depends on how many calls you need
19:07.40Daejeoping [TK]D-Fender:
19:07.46Daejeodig [TK]D-Fender:
19:07.51Daejeohost [TK]D-Fender:
19:07.53zchaosi can't even make 1 call
19:07.54zchaosits so choppy
19:07.56rene-for one call 100 kbps should be enough if not using compression, 50ks will do with g729
19:08.05rene-but then again are u sharing the connection?
19:08.11rene-running servers, sending email
19:08.15rene-uploading stuff? gaming?
19:08.27*** join/#asterisk Bilano (n=no@66.54.249.50)
19:08.40zchaosactually
19:08.42zchaosi think my bittorrent was going
19:08.43zchaos:P
19:08.45[TK]D-FenderDaejeo: Yes?
19:08.51[TK]D-Fenderzchaos: SMRT
19:08.52zchaosnot that you mentioned it
19:08.59BilanoGuys, I got me a small newbie issue.
19:09.01zchaosnow*
19:09.42BilanoI can get incoming calls, but only if my first extension has the Direct DID filled out. I can't seem to send an incoming call into the IVR...
19:10.38*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
19:10.46BilanoIf I remove the Direct DID from my extension, I get something in the full log that says I've sent the call to an invalid extension.
19:10.53[TK]D-FenderKatty: OUCH.. pricey
19:10.54StephenFBilano pastebin your extensions.conf
19:10.57BilanoThe extension it thinks I've sent the call to is the incoming DID.
19:11.11BilanoI've told the IVR to send it to extension 200, however.
19:11.15BilanoOkay, one sec.
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19:14.21BilanoMy extensions.conf is the default, it has not been touched.
19:14.34Keithdizzlecan someone help me? i don't understand how to apply a patch i'm trying to use.
19:14.56Katty[TK]D-Fender: do you have a nice lil camcorder recommendation?
19:15.01Keithdizzlethis patch in particular: http://bugs.digium.com/view.php?id=8824
19:15.29[TK]D-FenderKatty: Ask yourself what it offers considering there is stuff out there thats 1/2 the price
19:15.37[TK]D-FenderKatty: And how muc you're really going to use it
19:15.43Keithdizzlei looked up how to apply the patch and i applied it and recompiled, but it doesn't seem like it did anything.
19:15.56Keithdizzlei can't find the functionalit that it should have added.
19:16.01Keithdizzlefunctionality.
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19:18.30Keithdizzleif anyone could help me, i'd really appreciate it.
19:19.18Keithdizzlei mean, first of all, are the files on that bugs.digum.com page what i'm supposed to be applying? am i supposed to be checking it out from an svn repository?
19:19.46Keithdizzlethere doesn't seem to be much of a faq on that site.
19:20.32*** join/#asterisk bbryant (n=brett@68.208.65.50)
19:20.48[TK]D-FenderKeithdizzle: the bug entry says what versiont he patch is for.  It may or may not be compatible with a newer release
19:21.52Keithdizzlewell in the attachments, he has a file for 1.4.21.2, which is the version i'm using.
19:22.19Keithdizzlei applied that file to the 1.4.21.2 source and the output indicated that everything went well.
19:22.39Katty[TK]D-Fender: it's just for riddick, and family events.
19:22.46Keithdizzlethen i did "make clean" "./configure" "make" and then "make install".
19:23.10Katty[TK]D-Fender: i've no idea what i'm looking for in a camcorder, the one listed above was the highest rated on newegg.
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19:26.17Keithdizzleonce i started up asterisk, i don't see the function listed when i do "core show functions" and any attempt to use it in extensions.conf fails.
19:27.41jerrecommendations on moving a production 1.4 pbx to 1.6 this early?
19:28.19*** join/#asterisk iCEBrkr (i=icebrkr@cyberdyne.org)
19:28.33*** join/#asterisk CrazyTux (n=brandon@user-vcauh95.dsl.mindspring.com)
19:28.33Kattyhugs iCEBrkr
19:28.36iCEBrkryo
19:28.42Keithdizzleso uh yeah, if anyone could help me or maybe point me to what i might be doing wrong, i'd sure appreciate it...
19:28.51CrazyTux[TK]D-Fender: Hey, quick question in 1.4.2.*, using AMI (asterisk manager), can I write a command to one client connected to the server, and read from another client connected to the server at the same time, i.e. does AMI broadcast the events / dialog, or does it send back to the same client socket.
19:29.15*** join/#asterisk nicoAMG (i=asgalt@201.203.96.42)
19:29.37iCEBrkrCrazyTux: It broadcasts
19:30.02tzafrir_laptopKeithdizzle, what version of Asterisk do you use now? asterisk -V
19:30.04iCEBrkrI mean, it'll spit back the results.
19:30.21CrazyTuxiCEBrkr: to all clients connected to socket server with events ON?
19:30.34CrazyTuxiCEBrkr: or only back to the requesting client
19:30.46iCEBrkrYeah
19:30.57CrazyTuxiCEBrkr: which one?
19:31.07iCEBrkrCrazyTux: If one client sends an Action:  All the other clients will see the Events:
19:31.20*** join/#asterisk bijit (n=benji@190.241.15.48)
19:31.42iCEBrkrSpeaking of AMI events.
19:31.58Keithdizzlei'm using asterisk 1.4.21.2
19:31.58iCEBrkrI wrote a quick and dirty v2 of my callerID app with URL launching! weeee!
19:32.03Keithdizzleand i applied 1.4.21.2.patch from http://bugs.digium.com/view.php?id=8824
19:32.57Keithdizzleit seemed to go just fine and i recompiled alright, but it didn't seem to do much.
19:33.06Keithdizzleit certainly didn't seem to add anything new...
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19:33.23tzafrir_laptopKeithdizzle, what do you mean by " any attempt to use it in extensions.conf fails."? What do you do? What happens?
19:35.50[TK]D-FenderKeithdizzle: Doesn't list something you say... PASTEBIN your "list" and tell us what you think you should be seeing...
19:37.34Keithdizzleexten => 701,1,Noop(Parked Test ${CALLERID(ALL)})
19:37.35Keithdizzleexten => 701,n,Set(CALLEDID(all)="Voice Mail" <500>)
19:37.35Keithdizzleexten => 701,n,Answer()
19:37.43Keithdizzleso that's what i've got for extensions as a test.
19:38.07Keithdizzle[Nov 10 11:36:55] ERROR[9639]: pbx.c:1564 ast_func_write: Function CALLEDID not registered
19:38.11Keithdizzleand that's what i get.
19:39.25[TK]D-FenderKeithdizzle: http://bugs.digium.com/view.php?id=8824 <-- where do you see that function listed in this link?
19:40.07Keithdizzleas far as my list, what i mean is when i do "core show functions" i don't see calledid in there.
19:40.16KeithdizzleThe CALLEDID() function replaces the RemoteParty() application, it can be used
19:40.16Keithdizzleto name channels that otherwise have no set callerid such as trunks and other
19:40.16Keithdizzleinternal applications (VoiceMailMain, MeetMe etc.)
19:40.22Keithdizzlegareth (reporter)
19:40.22Keithdizzle2007-08-26 05:49
19:40.26[TK]D-FenderKeiwhere did you even get the idea that that function name is VALID>
19:40.40Keithdizzlethat post is on the bugs page.
19:41.37Keithdizzleam i misunderstanding what he's saying?
19:41.51Keithdizzlei tried RemoteParty() as well, but to no avail.
19:42.00[TK]D-FenderkeithPB <-
19:42.22Keithdizzle???
19:42.26[TK]D-FenderKeithdizzle: and that function you saw referenced was a SUGGESTION
19:42.29[TK]D-FenderPASTEBIN
19:43.58Keithdizzlehow...uhhh...do i use pastebin?
19:44.57[TK]D-Fenderkei~pb
19:44.59[TK]D-Fender~pb
19:45.00jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
19:45.19Keithdizzleok. thanks.
19:46.35Keithdizzleso what are you looking for me to paste then? just the function list or...?
19:49.26[TK]D-FenderkeithWhat you are trying to do, your lists at CLI, etc
19:49.44Keithdizzleyeah i was just doing "core show functions".
19:50.15[TK]D-FenderKeithdizzle: Include that
19:50.19Keithdizzlei thought that i should see "remoteparty" or "calledid".
19:50.33[TK]D-FenderKeithdizzle: Keeping in mind no document I see says that function should even exist
19:50.46Keithdizzlehttp://pastebin.com/d302d6b94
19:51.12Keithdizzlewhat about the description at the top? it talks all about using remoteparty.
19:51.17[TK]D-FenderKeithdizzle: Now pastebin the doc that says that function should exist
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19:51.42[TK]D-FenderKeithdizzle: what "description"?  Sho me the exact statement that says "this is the thing to call"
19:52.04Keithdizzlehttp://pastebin.com/d1f8732
19:52.39[TK]D-Fenderkeithdo YOU see a reference to a FUNCTION in there?\ I sure don't
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19:54.03Keithdizzlewhat about step 1? i guess that would be an application and not a function?
19:54.06jfarrellgreetings all
19:54.46jfarrelli have a question, this may or may not be relevant to this channel, if it isnt, please let me know
19:54.50*** join/#asterisk masus (i=masus@88.248.14.186)
19:54.53jfarrelldoes anyone know if the possibility of interop is available for .NET with respect to the HUDclient
19:55.11jfarrellthat is, can an external application tie into HUD at all and extended it?
19:55.14jfarrell*extend
19:56.03[TK]D-FenderKeithdizzle: Step 1
19:56.16[TK]D-FenderKeithdizzle: Step 1 - follow the INSTRUCTIONS
19:58.56Keithdizzlei...guess i don't understand. i didn't see anything besides his list of current functionality.
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20:00.16rwaitecan i nest $[] blocks? so can i go GotoIf($[ $[] | $[]]?this:that)
20:00.22Keithdizzleam i missing something about his description? i don't really see any instructions.
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20:00.55[TK]D-FenderKeithdizzle: exten => s,1,RemoteParty("Voicemail" <123>) <-- is this sample not blatant enough for you?
20:01.15[TK]D-FenderKeithdizzle: Its in your own pastebin for the instructions you claim to be following
20:01.37[TK]D-FenderKeithdizzle: Instead you seem to be pulling function names out of THIN AIR and ignoring the stated sample.
20:01.43Keithdizzleright right, i mean i tried that as well.
20:02.04Keithdizzle<Keithdizzle> i tried RemoteParty() as well, but to no avail.
20:02.04Keithdizzle<[TK]D-Fender> keithPB <-
20:02.38*** join/#asterisk Akiyuki (n=jimi@rrcs-70-63-90-226.midsouth.biz.rr.com)
20:02.44AkiyukiCan a vonage ATA be unlocked?
20:03.01AkiyukiOr, can you put the IP address it is using in /etc/hosts
20:03.05Keithdizzleso his description says to use RemoteParty but then his post later on says that "The CALLEDID() function replaces the RemoteParty() application, it can be used
20:03.05Keithdizzleto name channels that otherwise have no set callerid such as trunks and other
20:03.05Keithdizzleinternal applications (VoiceMailMain, MeetMe etc.)
20:03.06Keithdizzle"
20:03.11[TK]D-FenderKeithdizzle: Feel free to try again, this time following the instructions and showing that you are able
20:03.28[TK]D-FenderKeithdizzle: pastebin the entire entry and stop spamming the channel
20:03.39Kattydies from spam overload.
20:03.46*** join/#asterisk numbshot (n=numbshot@189.205.75.92)
20:03.50[TK]D-FenderAkiyuki: Some yes, others no.  You are asking for pain.
20:04.00Keithdizzlehttp://pastebin.com/m2c5c8194
20:04.44[TK]D-FenderKeithdizzle: Go look in your modules folder.
20:04.58Akiyuki[TK]D-Fender: Just trying to save $50.. I guess I coul djust shell it out and save myself some trouble. Or get a wireless SIP phone.
20:05.24[TK]D-FenderKeithdizzle: http://bugs.digium.com/view.php?id=6643 <-- that paragraph didn't come from here which is where you said you patched rfom.
20:05.41KattyDear Lord, please grant me the ability to punch people in the face over standard TCP/IP.
20:05.57[TK]D-FenderKatty: AT&T hasn't extended that far yet ;)
20:06.17Daejeo<PROTECTED>
20:06.19Keithdizzle<[TK]D-Fender> Keithdizzle: http://bugs.digium.com/view.php?id=8824 <-- where do you see that function listed in this link?
20:06.30Keithdizzleyou listed...the wrong bug report.
20:06.34Daejeo<PROTECTED>
20:07.18CrazyTuxKatty: :)
20:07.27*** join/#asterisk qdk (n=qdk@94.191.219.74.bredband.3.dk)
20:07.32[TK]D-Fender[14:14]<Keithdizzle>can someone help me? i don't understand how to apply a patch i'm trying to use.
20:07.34[TK]D-Fender[14:15]<Keithdizzle>this patch in particular: http://bugs.digium.com/view.php?id=8824
20:07.38[TK]D-FenderKeithdizzle: YOU said it.
20:07.50[TK]D-FenderKeithdizzle: You are massively inconsistent.
20:08.13[TK]D-FenderKeithdizzle: Get your head screwed on straight already
20:08.18sdanielsI musty be an idiot or something, can someone give me a hand getting all the console messages to syslog?
20:08.23Keithdizzle<[TK]D-Fender> Keithdizzle: http://bugs.digium.com/view.php?id=6643 <-- that paragraph didn't come from here which is where you said you patched rfom.
20:08.36Keithdizzlesee...you've got the wrong bug report that you listed.
20:08.58Keithdizzleand what you pasted from me above is 8824 and not 6643.
20:09.10rwaiteSIP/blah-082c6df0 is circuit-busy << would this be the same as 'CHANUNAVAIL'?
20:09.31[TK]D-FenderKeithdizzle: You'd referred to 2 patches at this point.  Just go look in your modules folder.
20:09.46[TK]D-Fenderrwaite: No.
20:09.53Keithdizzleok, i'm looking. want the output?
20:10.02Kattyanyone know a better gallery than gallery v2.2?
20:10.04[TK]D-FenderKeithdizzle: feel free
20:10.29Keithdizzlehttp://pastebin.com/d53f49a32
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20:11.17Keithdizzleyou know, i'm just looking for help. if you would like to see more people use asterisk, maybe you could try being a bit friendlier. i already said i've never done this before and the site says to come here for help, not to talk to assholes.
20:12.01[TK]D-FenderKeithdizzle: Doesn't look installed...
20:12.14sdanielsnevermind, I got it. thanks
20:12.34Keithdizzleyeah i know, i mean that's what i'm asking. his .patch file didn't seem to do much and i am trying to figure out if i just don't understand how to apply this patch or what.
20:12.57[TK]D-FenderKeithdizzle: Normally the pathc mods your source in place and you need to recompile to get the new apps
20:12.59italorossianyone use py-asterisk manager interface?
20:13.10[TK]D-FenderKeithdizzle: Have you done this?  You should see it in menuselect
20:13.40Keithdizzleyeah i went to recompile, but it's not in menuselect.
20:14.17[TK]D-FenderKeithdizzle: I'd verify that your source was actually patched.
20:14.46Keithdizzlewell if i try to apply the patch again, it sees it's already been patched. is there something else i can do to verify it?
20:15.00[TK]D-FenderKeithdizzle: Just trash your whole folder and extract again
20:15.06Keithdizzleok.
20:16.00devilsoulblackhi anyone have this error http://lists.digium.com/pipermail/asterisk-r2/2008-November/000138.html
20:16.08jfarrellahh kernel patching, memories
20:16.15jfarrellno longers users linux
20:16.20seanbrighthe's not patching the kernel
20:16.22jfarrell*uses
20:16.25seanbrighthe's patching asterisk
20:16.29jfarrellahh
20:16.38Keithdizzleok, i've deleted the whole /usr/src/asterisk-1.4.21.2 directory and decompressed the whole tar again.
20:16.38jfarrelli saw menuselect and patching - i asumed
20:16.42jfarrellmy apologies
20:16.49seanbrightthere's a menuselect in asterisk as well
20:16.52seanbrighti forgive you
20:16.52seanbright;)
20:16.57jfarrellsweet
20:17.15[TK]D-Fenderdevilsoulblack: Here's an error... your channel ranges don't match at all
20:17.33Keithdizzleso this is the file i should be using to patch, am i correct? http://bugs.digium.com/file_download.php?file_id=20203&type=bug
20:17.52seanbrightKeithdizzle: what bug is that attached to?
20:18.00[TK]D-FenderKeithdizzle: what are you running?
20:18.07Keithdizzlesean: this one: http://bugs.digium.com/view.php?id=8824
20:18.18Keithdizzleyou mean what version?
20:18.24[TK]D-Fenderos asterisk
20:18.27[TK]D-Fenderof
20:18.33seanbrightKeithdizzle: cd /path/to/asterisk/src ; wget -O - "http://bugs.digium.com/file_download.php?file_id=20203&type=bug" | patch -p1
20:18.40devilsoulblack[TK]D-Fender, the telco give that info
20:19.01Keithdizzlei'm using linux, centos 5 i believe.
20:19.13[TK]D-Fenderdevilsoulblack: dahdi_cfg is souwing a completely mismatched range from chan_dahdi.conf
20:19.21[TK]D-FenderKeithdizzle: ASTERISK <-------
20:19.30seanbright1.4.21.2
20:20.08Keithdizzlehere's the output of me patching: http://pastebin.com/d20506fc
20:20.12[TK]D-FenderOk, looks fin so far if thats the ccase
20:20.16*** join/#asterisk feeds_ChZ (n=chatzill@85-135-235-105.adsl.slovanet.sk)
20:20.23seanbrightKeithdizzle: looks perfect
20:20.26Keithdizzle[root@tobias asterisk-1.4.21.2]# asterisk -V
20:20.26KeithdizzleAsterisk 1.4.21.2
20:21.03[TK]D-FenderKeithdizzle: ok, and then you did "./configure" , "make menuconfig" (looked for the app/function?), "make" make install"?
20:21.03Daejeo<PROTECTED>
20:21.10[TK]D-FenderDaejeo: No idea
20:21.18Keithdizzlethat's what i'm working on right now.
20:21.18QwellDaejeo: why not just do it locally, and bypass the provider?
20:21.19seanbrightwhat app/func are you expecting to see?
20:21.37Keithdizzlei believe it should be app_remoteparty, but i'm not quite sure.
20:21.45devilsoulblack[TK]D-Fender, i change on chan_dahdi.conf the same number channels and have the same issue
20:21.52seanbrightKeithdizzle: not based on that patch
20:21.54DaejeoQwell: it did it, but sometimes trunk fail
20:22.18Daejeoi want to handle fail over
20:22.43Daejeoany educated idea?
20:23.12Keithdizzleseanbright: ok, so am i using the wrong patch? are the files attached to that bug report even the actual patches?
20:24.27Keithdizzlei have no idea how to use their bug site and their guidelines page doesn't have much information.
20:24.56seanbrighthold on
20:25.45seanbrightit looks like it adds options to the Dial() application
20:25.54*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
20:25.58seanbrightthe f, I, and N options
20:26.07seanbrightonce you get it all built and installed, do a 'core show application Dial'
20:27.10Keithdizzleok...i see those. if that's all it did, what the hell is all this discussion about RemoteParty and CALLEDID?
20:27.16lesouvageIs it true that with the last version of sox (SoX v14.1.0 ) sox -m came in the place of soxmix or is this bare nonsens. (sorry, kind of ot  but surely asterisk related)
20:27.31seanbrightKeithdizzle: it does more... but that was the first thing i noticed.
20:28.24Keithdizzleok, i mean i know c and i can see that it doesn't even reference either of those functions, but i know very little about diff and thought maybe i just didn't understand what it was doing.
20:28.37seanbrightKeithdizzle: i think func_connectedline.so is what you are interested in
20:28.53seanbrightKeithdizzle: core show function CONNECTEDLINE
20:29.15Keithdizzleok..i get some output from that.'
20:29.35Keithdizzlehttp://pastebin.com/d7b249cf6
20:29.40seanbrightthat function was added by the patch
20:29.47seanbrightso i assume it's relevant to what you are looking for
20:29.56Keithdizzleok, thanks. i bet that's what it's been renamed to.
20:29.58seanbrightbrb
20:32.27beek[TK]D-Fender: Sangoma's wanfig_dahdi created an interesting chan_dahdi.conf file.   Under "Channels" it has "echocancel=yes, echocancelwhenbridged=yes", yet under the individual spans it added "echocancel=no".   As a result, Asterisk throws warnings when reloading chan_dahdi.  Which should it be?  echocancelwhenbridged=no under '[channels]' or echocancel=yes in the span?
20:33.14[TK]D-Fenderbeek: Your choice
20:33.38*** join/#asterisk seaq (n=seaq@190.144.113.26)
20:34.06beek[TK]D-Fender: So bridging these PRIs are unlikely to require echocancelling?
20:34.36[TK]D-Fenderbeek: generally not on bridge.  Otherwise yes, they almost certainly will
20:35.05beek[TK]D-Fender: Alrighty -- I'll turn it off for the PRI-PRI bridge and on for the channel bank.  Thanks!
20:35.16[TK]D-Fenderbeek: np, and glad to hear its going well
20:35.31beek[TK]D-Fender: I'm having a blast working on this.
20:39.07seanbrightand back
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20:40.02Keithdizzlehey sean, i'm still having trouble, but i'm going to work on it for a few minutes. thanks for being so helpful.
20:40.23seanbrightKeithdizzle: no sweat.  let me know if you need more help.
20:41.28Keithdizzleseanbright: ok, i've got it to work, but i'm getting errors on the console and i'm not sure if i should be concerned about it.
20:41.36seanbrightpastebin them
20:41.45Keithdizzleyep, one second.
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20:43.17Keithdizzlehttp://pastebin.com/d4d96b49d
20:43.43seanbrighti don't think that is something you need to be concerned with
20:43.46seanbright(but i might be wrong)
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20:46.08seanbrightKeithdizzle: i think i am wrong, but i don't know how to resolve that particular issue.
20:46.49Keithdizzleyeah no problems there. i just needed to figure out how to use this.
20:47.35Keithdizzlethanks so much.
20:47.40seanbrightnods
20:48.19Keithdizzleand [TK]D-Fender, i guess i'll thank you for your help as well, fraught as it was with berating statements and the general attitude of a complete douche bag.
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20:52.55[TK]D-FenderKeithdizzle: I refarined from berating, just pointed out the incositant bits as you were working off 2 different sources.
20:53.19[TK]D-FenderKeithdizzle: We've come prety far so far, just a bit of a bump start
20:53.43Keithdizzleyeah that was an accident. i'm still learning to use this bug system and those posts are linked.
20:53.51[TK]D-FenderKeithdizzle: Do you see an app or func being compiled yet?
20:53.57Keithdizzleyeah, i got it working.
20:54.01*** join/#asterisk seanbright (i=seanbrig@asterisk/contributor-and-bug-marshal/seanbright)
20:54.09[TK]D-FenderKeithdizzle: how functional, and using what models?
20:54.52Keithdizzlefollowing the posts, it went from application RemoteParty() to function CALLEDID to function LINEID and finally it's now called CONNECTEDLINE.
20:55.15[TK]D-FenderKeithdizzle: Story gets better every time I hear it... can't wait to see what its called next week :)
20:55.29[TK]D-FenderKeithdizzle: So what phones had you tested this with?
20:55.49Keithdizzlewell it doesn't work with my grandstream, but it worked with a polycom.
20:55.58*** join/#asterisk synchris (n=synchris@athedsl-86032.home.otenet.gr)
20:56.20Keithdizzlealbeit with a shit ton of these errors:
20:56.20Keithdizzle[Nov 10 12:42:04] WARNING[21099]: chan_sip.c:3707 sip_write: Asked to transmit frame type 64, while native formats is 0x4 (ulaw)(4) read/write = 0x4 (ulaw)(4)/0x4 (ulaw)(4)
20:56.33[TK]D-FenderKeithdizzle: I'm not surprised at the former.... Polycom, cisco, Linksys, and Aastra have the best bets... Snom is probably a decent bet.
20:57.11[TK]D-Fender~gs
20:57.11jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
20:57.24Keithdizzleyeah i had to get it working on polycom phones else i'm not going to be eating soon.
20:58.30[TK]D-Fender~cpid
20:58.31jbot[~cpid] Called-Party ID is possible with * using patches on Mantis.  See : http://bugs.digium.com/view.php?id=8824
20:58.34Keithdizzlejbot: a digium rep recommended grandstream when we first looked into this, but it only took two of them to realize they weren't all that nice.
20:58.57[TK]D-FenderKeithdizzle: Get us a name... we have people on the inside who can deal with him ;)
20:59.23[TK]D-FenderKeithdizzle: And don't talk with the bot... its too early for that kind of humour :)
21:00.11Keithdizzleis jbot an actual bot?
21:00.18fileyes.
21:00.19[TK]D-Fender~areyouadog ?
21:00.20jbotBark! Bark!
21:00.23filejbot: botsnack
21:00.23jbotaw, gee, file
21:00.31[TK]D-Fenderfile: DAMN YOU!
21:00.34Keithdizzlewow...that's a lot of work put into something not that exciting.
21:00.53rob0It excites me ... shrug
21:01.07[TK]D-FenderKeithdizzle: jbot is quite useful.
21:01.17[TK]D-Fenderfile: How goes?
21:01.27Keithdizzlei've been working on multivalue database programming for 3 years. that's about as unexciting as it gets.
21:01.39file[TK]D-Fender: not bad! yourself?
21:01.40Keithdizzlei guess i would be happy to work on an irc bot instead.
21:02.10[TK]D-FenderKeithdizzle: You can always distract yourself on the "All snail-racing" channel, or the "Astro-turf growing competition".
21:02.29[TK]D-Fenderfile: Getting by... few med situations to clean up I'm hoping don't haunt me down the line.
21:02.39file[TK]D-Fender: >_<
21:02.52[TK]D-FenderKeithdizzle: Well jbot is done and well trained with all sorts of useful info.
21:03.26[TK]D-Fenderfile: that mal-healed cut I hope to have surgery to correct, fire off all the dental work before year end, etc...
21:03.39*** join/#asterisk klictel (n=klictel@nat/digium/x-f44a2ca44fc16edc)
21:04.03[TK]D-Fenderfile: fun, fun, fun...
21:04.16file[TK]D-Fender: eep
21:04.25fileklictel: Claude!
21:04.27[TK]D-Fenderfile: All good news if they're able to...
21:04.35klictelpresent
21:04.39[TK]D-Fenderpast
21:04.43klictelheh
21:04.58klicteli'm sitting jared class in huntsville
21:05.30fileklictel: I thought so, the digium hostname gave it away :D
21:05.44klicteloh well
21:06.42klictelfile: any snow yet in your corner?
21:06.52fileklictel: thankfully no! but lots of rain
21:07.06klictelwe had some in mtl a couple of weeks back
21:07.18klicteldepressing.... i started crying
21:07.20[TK]D-Fenderklictel: Lol... a tiny bit of slush 1 night...
21:07.39klictelwell yeah maybe but still IT WAS SNOW IN OCTOBER
21:07.46[TK]D-Fenderklictel: Nothing to complain abaout.  Now if ew get smashed like last year, thats another matter...
21:07.48klictelnext year it will start in july
21:08.09klicteli am out to get a snow blower next week
21:08.30klicteldepressing
21:08.40[TK]D-Fenderklictel: I got my winter tires done last friday... jsut about set... took down the shovel for the car this morning...
21:09.03[TK]D-Fenderklictel: I want the green Christmas we had 3 years ago...
21:09.12klicteli still have to put on my tires... and my wife's car
21:09.17[TK]D-Fendertakes a few more cans of hairspray outside...
21:09.24klictelheh
21:11.05[TK]D-Fender20 mins to checkout time...
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21:15.55Katty1hr20 for me :<
21:17.51Keithdizzlei have another question.
21:18.10Keithdizzleis there any way when i park a call to have the caller who parked it know which extension it was parked to?
21:18.22*** join/#asterisk seaq (n=seaq@190.144.113.26)
21:19.20*** join/#asterisk RMod_ (n=nicolasj@unaffiliated/rmod)
21:20.01[TK]D-FenderKeithdizzle: it gets read back to you when you park it.
21:20.13[TK]D-FenderKeithdizzle: Which is why you need to be doing it in an attended transfer
21:20.19Keithdizzlethanks.
21:20.52Keithdizzlewell i guess what i mean is if there is any way for the dialplan to know.
21:21.32[TK]D-FenderKeithdizzle: Well an exten is generated for the parked call... so technically yes its trackable
21:22.26devilsoulblackopenr2 need asterisk-addon ?
21:22.40Kattypouts at [TK]D-Fender
21:22.43Katty[TK]D-Fender: can i go home wif you?
21:22.51Keithdizzleok, would i just try and list extensions to track it or something?
21:22.57*** join/#asterisk Filippino (n=Filippin@151.61.84.181)
21:23.15[TK]D-FenderKatty: Sure... c'mon up :)
21:23.34[TK]D-FenderKeithdizzle: Well what exactly are you trying to accomplish?
21:23.52Filippinohi.
21:24.02Filippinojust a problem with a grandstream bt200
21:24.09Filippinoanyone can help me pls ?
21:24.38Keithdizzlebasically my problem is that if a call comes in and gets parked, i need to display the original incoming caller id on the display of the user who actually picks up the parked call.
21:24.52Keithdizzleand not the extension that parked it?
21:25.14[TK]D-FenderKeithdizzle: IIRC thats what you get...
21:25.35Keithdizzlehuh?
21:25.40[TK]D-FenderKeithdizzle: Even through its an attended transfer it keeps the original CID.
21:26.16*** join/#asterisk baliktad (i=baliktad@c-24-17-254-250.hsd1.wa.comcast.net)
21:26.31[TK]D-FenderKeithdizzle: AH, but you pick up the call and you want it displayed BACK...
21:26.35Filippinoi have my asterisk running with a problem. sometimes I start a phonecall with my BT200 than when I hangup, asterisk doesn't ... after a lot lot lot progblems
21:26.45[TK]D-FenderKeithdizzle: yeah, that'd be trick, but possible with the CPID patch
21:26.57[TK]D-Fendertricky*
21:27.04baliktadI have a provider who wants to send me calls from an entire IP range (a /24) - how can I specify the SIP peer to accept calls from a whole range of IP's?
21:27.15Keithdizzleok, let me look up the cpid patch. thanks fender.
21:27.24[TK]D-FenderKeithdizzle: You'd have to build your own custom exten to do the pickup.
21:27.36[TK]D-FenderKeithdizzle: thats the patch you just spent all this time installing
21:27.52[TK]D-FenderKeithdizzle: this is what is required to see it when you pick them up.
21:28.27Keithdizzleright ok, sorry. i understand what you meant now.
21:28.38[TK]D-FenderKeithdizzle: You'd ahve to do some real scripting though.  Scan for the channel parked on the lot # you are calling for, set the CPID, then do the pickup in a local channel.  Rather complex, but possible
21:29.07[TK]D-FenderKeithdizzle: And it's look nasty from a CDR perspective.
21:29.11[TK]D-Fenderit'd
21:29.19[TK]D-FenderOk, well its checkout time here... back later.
21:29.23[TK]D-Fendergood luck to all.
21:31.25diegowsany spa 3102 (firmware 5.1.7) user here?
21:33.23*** join/#asterisk kusznir (n=kusznir@isg-grad-02a.eecs.wsu.edu)
21:33.57kusznirHi all:  I've got a hardware question:  I need to acquire a hardware phone (or something similar) that will function somewhat like an intercom.  Basically, I need a speakerphone that will auto-answer.
21:34.16*** join/#asterisk shinao1 (n=shinao1@62.173.48.42)
21:35.06kusznirIt needs decent mic pickup as well.  It will be used for announcements followed by comments from whomever is near it.  I'm currently doing it with a basic softphone, but the PC I have available is busy doing other stuff and the audio is frequently garbled and such.
21:35.26kusznirOh, and it would be nice if it was less than $200, preferably in the $100 range.
21:37.16baliktadI use the SPA-9xx phones, you can set a SIP header to have it auto answer and playback over the speakerphone
21:38.08baliktadI've set it up so users can dial the phone directly (extension 2xx) to have the phone ring; to page the phone, they dial 8+extension
21:41.36*** part/#asterisk seaq (n=seaq@190.144.113.26)
21:42.38moydevilsoulblack: no, it's not needed
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21:55.50Kattylmadsen: snow :<
21:58.39kusznirbaliktad: thanks!
21:59.01*** join/#asterisk riddlebox (n=adfad@75-128-170-26.static.stls.mo.charter.com)
21:59.23riddleboxdoes anyone use vicidial? does it work well with zap channels?
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22:08.12*** join/#asterisk littlepinkdot (n=thedot@69.7.43.20)
22:08.24littlepinkdotWhat would be the recomended echo cancellor?
22:08.37JThardware
22:08.56littlepinkdotI'm currently using OSLEC but finding that it doesn't autostart/load before the zaptel drivers do so it doesn't work 90% of the time.
22:08.57tzafrir_laptoposlec
22:09.13tzafrir_laptoplittlepinkdot, what do you mean?
22:09.45tzafrir_laptopthis should be fixed by a proper installation, I believe .
22:09.54littlepinkdotOn system boot, zaptel loads, oslec kernel module loads, freepbx/asterisk/etc loads. Since zaptel loaded before oslec did, oslec can't perform the function its designed to since it can't insert itself into the kernel.
22:10.11tzafrir_laptopwhat is the output of:   modinfo zaptel | grep ^depends
22:10.22littlepinkdotmodinfo zaptel | grep ^depends
22:10.24littlepinkdotErr
22:10.29littlepinkdotcrc-ccitt
22:10.44tzafrir_laptopfind /lib/modules -name oslec.ko
22:11.21littlepinkdotMy copy resides in /usr/src/oslec/kernel/oslec.ko, should I move it?
22:11.22*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
22:11.41tzafrir_laptopthat's not a proper install
22:11.59tzafrir_laptopfind /lib/modules -name zaptel.ko
22:12.00littlepinkdotI patched/compiled as the guide suggested
22:12.18littlepinkdot/lib/modules/2.6.18-8.el5/misc/zaptel.ko
22:12.18littlepinkdot/lib/modules/2.6.18-53.1.21.el5/misc/zaptel.ko
22:12.51tzafrir_laptoplittlepinkdot, kernel modules should be under /lib/modules/`uname -r`
22:13.06littlepinkdotHmm
22:13.07tzafrir_laptopprobably just put it in the same misc/ subdirectory
22:13.13tzafrir_laptopand then run depmod
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22:13.46tzafrir_laptoponce you do that, modinfo would show that zaptel depends on oslec
22:13.56AkiyukiIs it possible to tell asterisk to "call xxx , then once connected, transfer/confernce to yyyy" ?
22:14.12AkiyukiI do not want to build this app in perl Net::SIP modules if possible
22:14.48littlepinkdottzafrir_laptop, still shows crc-ccitt
22:15.06tzafrir_laptoponly that?
22:15.11littlepinkdotCorrect
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22:17.30tzafrir_laptopdid you run depmod?
22:17.40littlepinkdotYes
22:17.54tzafrir_laptopuname -r
22:19.11littlepinkdot2.6.18-53.1.21.el5
22:19.11littlepinkdot[root@voip1 oslec]# find /lib/modules -name oslec.ko
22:19.11littlepinkdot/lib/modules/2.6.18-53.1.21.el5/misc/oslec.ko
22:19.38tzafrir_laptopmodinfo /lib/modules/2.6.18-53.1.21.el5/misc/oslec.ko | grep ^vermagic
22:20.15kfifeanyone want to weigh in on whether to use the new g711 codec in 1.6?  How much more expensive is it?  Docs say 'cleaner' but what practical benefit does it offer?  More reliable inband DTMF for example?
22:20.19littlepinkdotmodinfo /lib/modules/2.6.18-53.1.21.el5/misc/oslec.ko | grep ^vermagic
22:20.33littlepinkdotvermagic:       2.6.18-53.1.21.el5 SMP mod_unload 686 REGPARM 4KSTACKS gcc-4.1
22:26.21*** part/#asterisk galeras (n=galeras@190.159.51.107)
22:26.35littlepinkdotAny ideas tzafrir_laptop? =/
22:28.11tzafrir_laptopany chance you could test unloading zaptel and loading it (to see if modprobe actually somehow pulls zaptel)?
22:28.28tzafrir_laptoperr... pulls oslec
22:30.16littlepinkdotFATAL: Module zaptel is in use. when trying to remove
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22:30.41*** part/#asterisk marl (n=marl@78.149.78.173)
22:31.05AkiyukiIs it possible to tell asterisk to "call xxx , then once connected, transfer/confernce to yyyy" ?
22:32.51seanbrightAkiyuki: yes.  call files.
22:34.10Akiyukiseanbright: Do you have to set cronjobs on them?
22:34.20Akiyukiseanbright: Trying to build a real time auto-dialer for my CRM.
22:34.38seanbrightAkiyuki: do you want to call people on a scheduled basis?
22:34.48Akiyukino
22:34.48seanbrightor in an on-demand fashion
22:34.51Akiyukion demand
22:35.06seanbrightthen you could just write a script to generate the call file
22:35.12AkiyukiI want it to Call the customer at xxx then imediately have the phone ringing at yyyy, not whent hey answer.
22:35.23seanbrightor you could connect to asterisk via AMI and send an Originate
22:35.23AkiyukiOk, so as soon as i place the .call file in the directory, asterisk will read and dial it?
22:35.29seanbrightAkiyuki: correct
22:35.31AkiyukiAMI?
22:35.35AkiyukiIs that a perl module?
22:35.38seanbrightAsterisk Manager Interface
22:35.40seanbrightno
22:35.46seanbrightthere is a perl module that helps you connect to AMI
22:35.53*** join/#asterisk zchaos (n=none@CPE0080c828f609-CM0019474d28a8.cpe.net.cable.rogers.com)
22:35.53seanbrighttake a look at the wiki for more details
22:35.58seanbrighthttp://www.voip-info.org/
22:36.05AkiyukiAh ok. trying to do this in php. Might be better off to do it w/ creating the .call files?
22:36.15seanbrightAkiyuki: yeah
22:36.48AkiyukiMan, that is sweet. I  have been trying to do it with Net::SIP and getting no where fast :P
22:36.49zchaoscan anyone tell me of any good canadian VOIP service providers? i was going to use acanac... but i heard they only allow you to call 50 different telephone numbers before they force you to go to the business package
22:37.02zchaoswhat is that crap
22:37.10seanbright~itsp
22:37.11jbot[~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs
22:37.23seanbrightzchaos: /msg jbot ~itsplist-ca
22:37.31zchaosthanks
22:37.36zchaosis PSTN sip?
22:37.42seanbrightAkiyuki: yeah, call files are trivial in comparison.
22:37.50rob0~pstn
22:37.50jbotfrom memory, pstn is Public Switched Telephone Network, or "please stop the nonsense"
22:38.55Akiyukiseanbright: Can a call file have multiple numbers to attempt? Reading the wiki abuot them now but do not see that
22:39.13*** join/#asterisk jer (n=jer@unaffiliated/jer)
22:39.25seanbrightAkiyuki: you can have the call file call a local channel, which in turn can dial multiple people
22:39.26zchaosdo all VOIP service providers allow you to only call 50 different numbers a month?
22:39.29zchaosbefore charging you extra?
22:39.34seanbrightAkiyuki: local channels are also detailed on the wiki
22:39.40rene-riddlebox: i think it was designed with zap channels in mind
22:41.48littlepinkdottzafrir_laptop, think I just have to get oslec to load before zaptel does, right now it loads via modprobe.conf but that gets read after zaptel is already loaded.
22:42.49*** join/#asterisk chazz (n=chazz@nat/digium/x-d5445bf82d964ad5)
22:46.33zchaosdo all VOIP service providers allow you to only call 50 different numbers a month?
22:46.50*** join/#asterisk Bilano (n=no@66.54.249.50)
22:46.54BilanoHowdy gents.
22:47.01littlepinkdotzchaos, where'd you hear that?
22:48.37zchaosacanac said that
22:48.52zchaosthats what i said
22:48.54zchaosi was like wtf
22:48.55zchaostahts gay
22:49.46*** join/#asterisk ManxPower (n=manxpowe@123.sub-75-203-80.myvzw.com)
22:54.07littlepinkdotBlegh...anyone have experience with OSLEC?
22:55.02YournameLOL zchaos that's hilarious!
22:55.52*** part/#asterisk masus (i=masus@88.248.14.186)
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22:57.15zchaosi'm not kidding
22:57.18zchaosi ordered it then if ound out later
22:57.22zchaosand the first thing i said
22:57.29zchaoswas... cancel me and refund my shit you never told me that
22:58.08zchaoswtfff do i do
22:58.09zchaoshttp://www.babytel.ca/content_pages/babyPLANS.html
22:58.15zchaosi need 2 phone lines
22:58.30zchaosthats $40 a month for 2 lines
22:58.40zchaosthe canadian village package
23:03.46*** join/#asterisk jer (n=jer@unaffiliated/jer)
23:05.33ManxPowerlittlepinkdot: I have heard good things about OSLEC, but I've never used it.  I've use the HPEC before, but these days I just stick to Tellabs hardware EC devices.  You plug them in between the telco and Asterisk and never have echo again.
23:08.34*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
23:09.56*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)
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23:12.20`SeanHey, does anyone know where i can get a 1900 number with unlimted channels or lke 10 chanels?
23:13.51ManxPower`Sean: I doubt you'll fine ANYONE that sells 1-900 VoIP service.
23:14.01ManxPowerHeck, most telcos won't even offer the service.
23:14.09rene-manx what do u think about digium hardware echo canceller?
23:14.34jer_anyone recommend (or not) upgrading a production 1.4 * box to 1.6 ?
23:14.35ManxPowerrene-: "It works good enough for most people" is what I've heard.
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23:14.48ManxPowerjer_: did you read the upgrade notes?
23:14.52jer_ManxPower, yes
23:15.06ManxPowerWOW!  Someone finally read them!!!!!!
23:15.09jer_=]
23:15.14jer_but it doesn't tell me what i want to know
23:15.24ManxPowerjer_: 1.6.x is far too new for me to put it into production.
23:15.32jer_it's not about what's changed, i want to know from people actually using it, whether or not they'd recommend upgrading a production 1.4 box to 1.6
23:15.39jer_ManxPower, yeah that's my hangup too
23:16.02ManxPowerjer_: I figure I'll wait until Digium upgrades their corporate PBX to 1.6.  That would demonstrate Digium's confidence in 1.6
23:16.14jer_ah, heh, good point =]
23:16.35ManxPowerRight now it's more like "Hey!  We have this great new release!  Use it!  No, we don't use it, but you should!"
23:17.08StephenFWhat do you guys use to test your SIP connectivity to your ITSP? Users are complaining about call quality, trying to determine the problem
23:17.29ManxPowerStephenF: many of us don't even use an ITSP.
23:17.48StephenFManxPower due to quality?
23:17.56rene-StephenF: reliability
23:17.59rene-quality
23:18.05rene-sometimes cost
23:18.08ManxPowerStephenF: The internet is not reliable to send calls over for my clients.
23:18.14ManxPowerreliable enough, that is.
23:18.16Mark_LoganTrue.
23:18.20rene-it is almost there
23:18.23rene-but not quite
23:18.31Mark_LoganCall me when it is okay :P
23:18.33ManxPowerrene-: apparently your users have low expectations.
23:18.38Mark_Loganlol
23:18.41Daejeoanyone from austraila?
23:18.51Daejeoanyone from Australia?
23:19.01rene-actually they arent
23:19.12ManxPowerVoIPoInternet seems to be about as reliable as a cell phone -- most of the time.
23:19.13rene-but sometimes they are cheap
23:19.49rene-if u have good connectivity like > t1 on both ends should be ok, as long as both ends are properly setup for QoS
23:20.15ManxPowerMY problem with VoIPoInternet is if sometime breaks who are you going to call?  It could be your ISP, but the problem could be with ANY network between you and your ITSP and since you are not their customer they are not going to fix it for you.
23:20.15rene-but it does not beats TDM on quality or reliabilty only cost
23:20.20`Seandamn it this is hard finding a 900 number with like 10 channels
23:20.28littlepinkdotGuess I'm back to my orignal question...how can I get oslec to load before zaptel so it actually loads?
23:20.29ManxPowerYo can't do QoS over the public internet.
23:20.35rene-ManxPower i know
23:20.46*** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com)
23:20.46rene-but at least you should make sure your upstream bandwidth is managed
23:20.56*** join/#asterisk steliosk (n=Stelios@athedsl-318864.home.otenet.gr)
23:21.05ManxPowerrene-: and their upstream, and their upstream and their upstream.
23:21.09rene-hehe
23:21.21rene-yes you can only pray it is going to be ok all the time
23:21.23rene-and it wont
23:21.32rene-if u have fiber
23:21.44StephenFso for a small office with 5 channels, what would your bring in as far as non-voip
23:21.44rene-at the two points then voip works very well
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23:22.14ManxPowerOne of our carriers (XFone) is having trouble talking to Charter.net, but they go thru some 3rd ISP.  Nobody will even admit thereis'a problem.  My users don't want to hear "well you can't VPN in because someone we have no influince over has a broken network).
23:22.16StephenF5 POTS lines?
23:22.32ManxPowerThey would lynch me if I told them that about voice.
23:23.02ManxPowerStephenF: in most markets 8 channels is the min on a PRI to have about the same cost as 8 analog lines.
23:23.32rene-StephenF: u can try with voip, it works to some extent, it is cheap, it may be right for your application/costumer
23:23.33ManxPowerIf you MUST use analog, I'd make sure failover happened between the POTS and the VoIP channels.
23:24.02ManxPowerBut really, if you can't afford an 8 channel PRI then you really can't afford my consulting services.
23:24.09StephenFwe are on voip now, its been OK but every once in a while we have quaility issues
23:24.25StephenFwhat does an 8 channel PRI for on average?
23:25.00StephenFcost on average*
23:25.03Mark_Logancostwise?
23:25.06Mark_Loganah
23:25.17ManxPowerStephenF: $300/month to $1,200/month
23:25.28Mark_LoganSo, pocket change.
23:25.37ManxPowerThe correct answer is "there is no average, each market is different."
23:25.41*** join/#asterisk Mad|Cow (n=user@static-72-94-249-58.phlapa.fios.verizon.net)
23:25.42StephenFok, now what if the customer doesnt need 8 channels?
23:25.55ManxPowerMark_Logan: Business lines are usually about $50/month.
23:26.14StephenFok so thats the next step down just POTS
23:26.16ManxPowerStephenF: Then it's not really my problem because they are not big enough of a client for me to be interested in.
23:26.24Mad|CowDoes anyone have a example they can share with me from their dialplan on how to enable the "Call Forward on No Answer" feature in asterisk?
23:26.39ManxPowerReally, I don't see much point in doing something other than PRI as your primary PSTN service.
23:26.56StephenFwhy is that?
23:27.01ManxPowerMad|Cow: You mean build a call forward no answer feature on asterisk, don't you?
23:27.04jblackMad|Cow: Do two dials one after the other.
23:27.12ManxPowerStephenF: because nothing else is as close to as reliable.
23:27.50Mad|CowManxPower: Yes, do you have an example I could see?
23:28.03ManxPowerAs an example, a friend has a cablemodem and service with Vitelity.  He says about %5 of the time when he wants to use the phone it does not work.  Is that reliable enough for you?
23:28.06Mad|Cowjblack: What do you mean?
23:28.17ManxPowerMad|Cow: Press Forward on your polycom phone, follow the menus
23:28.24StephenFsure, and if you say about $50/mo for a pots line 6 lines = a PRI
23:29.04ManxPowerStephenF: not if your PRI is $1,200/montjh
23:29.13jblackMadcow:  123, 1, Dial(Somephone)   \n  123,n,Dial(Someotherphone)
23:29.15ManxPoweri.e. most of Louisiana and much of the west.
23:29.16StephenFoh that was a range
23:29.18Mad|CowManxPower: I've tried, asterisk doesnt seem to be playing nicely; I was just on the phone with Digium and they say I have to add something to my dial plan to support it, but couldnt elaborate
23:29.42ManxPowerMad|Cow: Asterisk does not have a call forward feature.
23:29.43jblackYou have a paid contract with digium, and they wouldn't answer that?
23:29.48*** join/#asterisk seanbright (n=sean@asterisk/contributor-and-bug-marshal/seanbright)
23:30.38ManxPowerMad|Cow: Me, and most people on this channel use the call forwarding features of their phone, they don't build one for Asterisk.
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23:31.26littlepinkdotManxPower, *72 isnt call forwarding?
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23:32.13ManxPowerlittlepinkdot: Asterisk is not really a PBX.  Asterisk is more of a PBX Toolkit that lets you build a PBX.  You could build a service on Asterisk that uses *72 for server based Call forwarding.
23:32.15Mad|CowManxPower: Call forwarding works fine with my Polycom phones; I'm interested in a Call forward on No answer. I have a new Polycom 301 that supports this; but asterisk doesnt seem to acknowledge it. I'm told by support I have to have soemthing in my dialplan.
23:32.23ManxPowerMy question is "why bother"
23:32.38littlepinkdotAh, forgot thats not a part of FreePBX by default.
23:32.49[TK]D-FenderWhy on Earth would anyone leave that decision up to a PHONE?
23:32.52ManxPowerMad|Cow: I have very large scripts that have support for ADMINISTRATOR set CF-NA
23:33.45ManxPowerbut now I am off to the hottub
23:33.51devilsoulblackopenr2 work fine with freepbx 2.5 ?
23:34.15Mad|CowManxPower: Thanks for the offer; but I would like to figure out how to make it work using the softkeys on the phone
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23:35.14littlepinkdotAgh I give up
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23:35.37littlepinkdotIll just use the built in echo cancellor...for whatever thats worth.
23:35.50kfifeHi Guys--Question: Compiling Asterisk 1.6: app_fax compile runs into an error.  http://pastebin.com/m4b78f187 Is this a know issue or have I missed a dependency
23:37.09[TK]D-FenderMad|Cow: Softkeys?  Where to go / what to do on on-answer isn't a KEY.  You don't push for it.  It just happens.  So why is this left up to a phone?  When is the decision ever going to be different?
23:37.42kfifeManxPower --> littlepinkdot: well said "Asterisk is more of a PBX Toolkit "
23:39.08voxterWhat headsets do you guys use with polycom phones (wired headsets)? I've had some trouble with Plantronics S11/S12 wired headsets + echo
23:39.17[TK]D-Fenderkfife: not even "more of a".  it IS a toolkit.  Nothing about it is inherently a PBX.  You do have to configure everything.
23:39.35[TK]D-Fendervoxter: I use Plantronics M22's + H261 binaural.
23:40.13kfife[TK]D-Fender: Even more well said.  We have one installation here that is NOTHING Like a PBX
23:40.46kfife[TK]D-Fender: but it's a lot like an amazing business tool
23:41.45voxter[TK]D-Fender: can you get those in a combo, or do you buy them separate for about $200 in total?
23:42.11[TK]D-FenderAll separate.  I get good pricing from my reseller here
23:45.23voxter[TK]D-Fender: yeah ill check those out. I buy direct too. the S11/S12 are a nice combo package but i think maybe they have their own amp, and so does the phone, and somewhere in the mix echo is introduced... very annoying.  Thx for the tip.
23:45.58[TK]D-Fendervoxter: I happily pay for quality
23:46.06voxter[TK]D-Fender: me too.
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23:58.42ManxPower[TK]D-Fender: I was trying to be nice.  I'm a pretty harsh critic of Digium's marketing department, I was just trying not say bad things about them today
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