IRC log for #asterisk on 20081105

00:00.44pepsemmm polycom
00:01.48nikkogot to hold the new adtran phones, they are nice
00:02.15nikko$249 and $299 is the street price i think
00:05.14drmessano~polycommunists
00:05.18drmessano~polycommunist
00:05.19jbotA polycommunist is someone who believes Polycom phones can do no wrong.. that Polycom's are so over and above anything else, that what you are using is surely crap, and the mere fact you mention another brand name is of great insult to the channel, the community, and the world.  They may also be getting a 10% kickback.
00:05.46*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
00:10.00trnzmetaman this chan is weird... go go palin!
00:16.56*** part/#asterisk jvanreij (n=chatzill@hoefnix.Stanford.EDU)
00:18.00*** part/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-6e72f799222c11d5)
00:28.08*** join/#asterisk ManxPower (n=manxpowe@2.sub-70-221-66.myvzw.com)
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00:51.13pcrane[Nov  5 13:44:42] NOTICE[27010]: chan_sip.c:2943 auto_congest: Auto-congesting SIP/7001-082c83e0
00:51.14pcrane[Nov  5 13:44:42]     -- SIP/7001-082c83e0 is circuit-busy
00:51.24pcranedoes anyone know what that means?
00:51.43pcraneor how to trouble shoot?
00:52.44*** join/#asterisk RobertLaptop (n=rmiddle@pool-72-81-212-249.bltmmd.fios.verizon.net)
00:58.15*** join/#asterisk DaveCanoe (n=Dave@strike.dclg.ca)
01:00.55*** join/#asterisk troy- (n=troy@worldnet.tauri.ca)
01:01.41troy-how can i make an extension ring for a specified number of seconds?
01:02.02*** join/#asterisk Micc (n=dotirc@c-67-183-169-202.hsd1.wa.comcast.net)
01:02.11jayteetroy, use the timeout in the Dial() apps
01:02.13jayteeapp
01:03.01troy-thx
01:03.16MiccAre there any active/active high availability asterisk distributions?
01:04.02ManxPowergenerally autocongestion means there was no response to the request
01:06.35drmessanoDistributions?
01:08.31drmessanoLet me translate and save some trouble
01:08.45jayteeadjusts his pants
01:09.36drmessanoDistributions = Trixbox
01:10.57drmessanoThere is no such thing has a high availability trixbox solution.  Matter of fact, "trixbox", "High", and "availability" should never be used in the same sentence, with one exception.. "Dude, I got high last night and installed trixbox, which reduced my availability to go to work the next day".
01:11.28Micchmmm... so what can be done?
01:11.46MiccHow can I make asterisk fail safe?
01:11.46drmessanoYou can come up with something on your own.. there is more to life than trixbox
01:12.15QwellMicc: how many 9s do you want?
01:12.32MiccQwell, as many as possible.
01:12.51MiccQwell, I'd like to start a phone company to compete with vonage in my local area.
01:12.54QwellI can sell you 1 9
01:12.56drmessanoI'll take a few 8's and a capital W
01:13.01MiccBut I need better call quality and better reliability.
01:13.18QwellSo, you want to be the next Vonage?
01:13.32drmessanoOh, and I need 3 J's and an %
01:13.33Micclocally, in a way.
01:13.49MiccI'd like to offer small businesses pbx and phone service.
01:13.59MiccSo a more personalized vonage.
01:14.03drmessanoUsing Trixbox to become the next vonage.  Thats original.  No, really... only heard that one 100 or so times.
01:14.06jaytee"And now Foreskin Systems has released the latest, most revolutionary VOIP PBX software of the 21st Century: JizzVox! Just listen to a sample comment from one of our satisfied customers. "Yo, wussup? DF in the house! I gots ta tell ya this JizzVox shit gonna fuck yo up!"
01:14.39drmessanoMicc: You won't get far on asterisk "distributions"
01:14.44QwellMicc: if you want that type of availability, you need to A) do it yoyrself
01:14.45drmessanoLearn Linux
01:14.50drmessanoLearn Asterisk
01:14.51QwellB) pay somebody
01:14.52drmessano????
01:14.55drmessanoProfit
01:15.02jayteepray to Cthulu
01:15.11MiccI can do it myself if need be. I've been working with asterisk since 2004.
01:15.23MiccI would just rather not if something already exists.
01:15.32*** join/#asterisk tacubo (n=tacvbo@189.146.192.181)
01:15.42MiccQwell, I don't mind paying someone.
01:15.53drmessanoI wanted a high availability Windows system, so I installed Linux
01:16.19drmessanoThat was kinda funny
01:16.22drmessanoAnyway
01:16.22MiccI've been running my asterisk 1.2 server without problems for years now.
01:16.37drmessanoStraight asterisk?
01:16.40drmessanoNo GUI?
01:16.45Miccyes.
01:16.52MiccI don't use any guis.
01:16.57jayteeI've been running my Novell Netware 286 server for over 18 1/2 years now
01:17.04drmessanoSo why did you ask about a "Distribution?"
01:17.06MiccBut I wrote our own sql config stuff.
01:17.23drmessanoI wasted like 3 or 4 good trixbox jokes
01:18.01Miccbecause I've been reading a lot of things like asteriskNow, and the asterisk appliance from digium.
01:18.22MiccI wanted to know if one of those or something else would work.
01:18.27jayteewhen I want appliances I go to Sears. When I want VOIP i build it myself
01:18.32MiccIt looks like the digium appliance does virtualization.
01:18.42drmessanoO.o
01:18.51drmessanoThe digium appliance.. does virtualization?
01:18.59drmessanoIts virtually an appliance.. WTF
01:19.16jayteeI think they mean it's virtually guaranteed to make you wish you'd bought something else.
01:19.16Qwellyeah...what?
01:19.18drmessanoWhat would it be emulating virtually?
01:19.23Miccyeah the switchvox thing.
01:19.28drmessanoIts an APPLIANCE
01:19.38drmessanoHow does it "do virtualization"?
01:19.55drmessanoDoes it fit inside another PBX?
01:20.01jayteemaybe that's why my toast never looks toasted. the virtual toaster is broken.
01:20.17Qwelljaytee: upgrade the toaster firmware
01:20.36drmessanojaytee: packing toast into your toaster and putting it in the oven is NOT VIRTUALIZATION
01:20.41Micclet me find that page I was reading about it.
01:21.01jayteeQwell, I may have to upgrade the virtual Mr Coffee because the virtual coffee hasn't been tasting even close to the real thing lately.
01:21.16drmessanoMicc: The fact you need to refer to a page of documentation rather than observing the lack of common sense here is quite disturbing
01:22.12drmessanoYour garage is not a hypervisor because you park your car in the garage.
01:22.36drmessanoThis is virtually silly
01:22.50trnzmetaanyone know of a good voip provider in sth africa
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01:22.50*** mode/#asterisk [+o russellb] by ChanServ
01:23.39drmessanorussellb: I hear the asterisk appliance supports virtualization.  Can you suggest an appliance I can use as a hypervisor for it?  Maybe a colecovision?
01:23.40carrarMorning Russle!
01:23.43carrarell
01:24.15MiccMaybe I read about someone else doing it.
01:24.31Micca home grown system. 200 calls per vm.
01:24.35drmessanoMicc
01:24.47drmessanoDo you understand what virtualization is?
01:24.58Miccyes.
01:25.06drmessanoHost OS running on the bare metal, OS's running on top?
01:25.28drmessanoHow does an APPLIANCE... a piece of hardware, operate in a virtual environment
01:26.03*** part/#asterisk russellb (n=russell@asterisk/digium-open-source-team-lead/russellb)
01:26.09MiccI don't know if that would fit the normal definition of appliance, but it sounds like a great idea to me.
01:26.16drmessano.....
01:26.46MiccAn appliance is just a machine when it comes down to it.
01:26.56drmessanoYes, its HARDWARE
01:27.04drmessanoSO how does hardware run on top of an OS virtually?
01:27.15drmessanoLike, how would a WRT54G run in a virtual machine on CentOS?
01:27.20MiccJust like it does on any hardware.
01:27.22drmessanoBlender?
01:27.39trnzmetahmmm...
01:27.40drmessanoSet it on top?
01:27.51drmessanoKrazyglue?
01:27.53MiccIf the vm was part of the hardware.
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01:27.56drmessanoStapler
01:28.20MiccAre you drunk or something?
01:28.20trnzmetago go palin
01:28.38MiccI feel like I'm talking to a 5th grader.
01:28.48drmessanoMicc: You are clearly the one that I should be asking that of.. the suggestion that you can run a piece of hardware on top of an OS is non-sensical
01:29.24trnzmetaI think he means applications
01:29.28MiccYou are not getting what I'm talking about, clearly.
01:29.39drmessanoI have tried metaphors, and I have tried common sense.. nothing is getting through
01:29.56drmessanoMicc: No, I think you clearly don't understand the terms
01:30.14trnzmetadid I already say... go go palin
01:30.15drmessanogo go ron paul!
01:30.19Miccappliances run an OS.
01:30.27Miccso they can run vm's as well.
01:30.51MiccIt wouldn't be your standard appliance at that point though.
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01:31.21drmessanoYou're gonna use an appliance, which is a stripped down unitasker, as a host for a virtual environment?
01:32.05MiccIts more of a prebuilt machine, not an appliance.
01:32.12MiccIts just a solution in a box.
01:32.45drmessanoUm ok
01:32.47drmessanoIf you say so
01:33.23MiccI'm not saying I would do it. I just saw it somewhere.
01:33.31trnzmetaok sorry I've lost something
01:33.51trnzmetabascially you want to run multiple asterisk on the single machine?
01:34.03MiccAll I'm getting at is that it sounds like I can run asterisk in a vm as long as I don't need to use a PRI card.
01:34.10MiccYes.
01:34.18drmessanotrnzmeta: On an asterisk appliance
01:34.35Miccdrmessano, no, not on an appliance. on anything.
01:34.43drmessano....
01:34.44drmessanoWTF
01:34.57trnzmetait's possible with virtualisation out there
01:35.00Miccdrmessano,I'm just saying I got the idea from some appliance type thing I saw somewhere.
01:35.07trnzmetadont' know what the debate is here
01:35.19trnzmetaI"m sure that's what the VSP are doing
01:35.34trnzmetaif you did this on a
01:35.56trnzmeta"appliance" you'd need to go embedded and play around wiht firmware
01:36.34drmessanoClearly I am the idiot here.. we went from "The asterisk appliance supports virtualization" to "I want to run an asterisk appliance as a virtual host" to "I want to run asterisk in a virtual environment"
01:37.17trnzmetayes drmessano you need to perfect your mental telepathy skills
01:37.20trnzmetaderrr
01:37.45drmessanoI tried to make sense of the convo, but I should have called off the search after dark
01:38.40trnzmetaguys: I have a situation where on my demo box in sth africa
01:39.00trnzmetathe connection to my provider is showing LAGED 783ms
01:39.36trnzmeta"sip show peers" mind you my provider is in Aust
01:39.50drmessano783ms is no big deal.. Just make sure everyone says "over" between sentences
01:40.21trnzmetahahaha, the problem is my demo box runs automated IVR out, so there is no way to test
01:40.41trnzmetafrom past experiences does 783ms lag stand a chance to even make a phone call
01:40.53trnzmetaI'm getting "unable to request sip channel" on my logs
01:41.27*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.168)
01:41.55trnzmetahowever it does register... but I then get "Peer is unreachable" in the logs
01:42.32drmessanoI wish I hadn't dropped out of electoral college
01:42.56drmessanotrnzmeta: Sounds like high latency and packet loss
01:43.17trnzmetathought as much... now I need to find a provider in sth africa!
01:43.30trnzmetaI have a provider in norway... but... not sure if that will work
01:44.19MiccAre there any SIP appliances that do load balancing like an F5 BigIP would do for a web server?
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01:52.25trnzmetawhat does F5 big ip do btw?
01:56.00MiccIt is a big router that will send http requests to one of a number of servers in a number of different fashions, for example: round robin.
01:56.01ManxPowerSER/OpenSER should be able to do load balancing as well as pretty much every SIP magic you can think of
01:56.17MiccAnd if any server goes down it routes all the other requests for other web servers.
01:56.19*** join/#asterisk pcrane (n=pcrane@120.89.80.54)
01:56.48MiccDoes SER work with asterisk?
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01:57.40Miccnevermind. I found my answer.
01:57.46Micchttp://www.voip-info.org/wiki/view/Asterisk+at+large
02:05.21trnzmetamicc: you mean high availability and clusterin?
02:06.04Miccyeah
02:06.25MiccI want to support 10,000 SIP phones.
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02:10.15Hadi-hello everyone
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02:12.56*** join/#asterisk BeeBuu (n=beebuu@219.132.190.182)
02:13.23BeeBuuhow can i know which agent is pick up in a queue?
02:15.19[TK]D-FenderBeeBuu: an AMI message is sent and you can see in "show queue [queuename]"
02:15.41jayteeah, he's back from martial arts
02:17.37BeeBuu[TK]D-Fender: when a user call in a queue, i want to know which agent pickup ,and use as a file name
02:18.25drmessanoBeeBuu: Call the number and say "Who is this"
02:18.47BeeBuudrmessano: funny....
02:19.01drmessanoI wasnt being funny
02:20.02[TK]D-FenderBeeBuu: You should be looking at how your agent is called.
02:20.11*** join/#asterisk gones (n=gones@219.134.228.30)
02:20.15BeeBuui tried Set(MONITOR_FILENAME=${CALLERID(NUM)}_${STRFTIME(${EPOCH},,%F-%H-%M-%S)}_${EXTEN}_${AgentChannel})
02:20.25Hadi-a little off topic but any cisco experts here
02:20.27BeeBuubut the ${AgentChannel} is null
02:20.30Hadi-cant seem to get caller ID name to work
02:20.40Hadi-with my asterisk box and AS5300 gw
02:20.41[TK]D-FenderBeeBuu: And what doc tells you that variable even exists?
02:20.58BeeBuuhttp://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List
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02:21.09BeeBuuAsterisk Detailed Variable List
02:21.20BeeBuuis it wrong?
02:22.58BeeBuu${AGENTNUMBER}
02:23.00jayteeI don't see an AgentChannel variable in that list on the WIKI
02:23.15[TK]D-FenderNeither do I
02:23.25BeeBuusorry,it's ${AGENTNUMBER}
02:23.30outtoluncthe term in 'interface'
02:23.33outtoluncer is
02:23.38jayteeand what is it in your actual code?
02:23.57BeeBuuSet(MONITOR_FILENAME=${CALLERID(NUM)}_${STRFTIME(${EPOCH},,%F-%H-%M-%S)}_${EXTEN}_${AGENTNUMBER})
02:24.13BeeBuui want mixmonitor
02:24.38jayteeI want to be rich and have a full head of hair
02:24.48BeeBuubut the filename above does'nt work...
02:26.48outtoluncchicken/egg
02:27.05jayteeBeeBuu, put a NoOp(${MONITOR_FILENAME}) after that line as the next priority just to see what it displays on the console.
02:27.43BeeBuujust ${AGENTNUMBER} is none~~
02:28.16BeeBuujaytee: what's the correct way to get that?
02:28.21jayteedunno
02:30.15*** join/#asterisk matt_ (n=matt@2001:770:168:1:20b:cdff:fe04:843a) [NETSPLIT VICTIM]
02:31.01[TK]D-FenderBeeBuu: what do you see in channelvariables.txt?
02:32.39BeeBuuso many,but i don't know which one for me
02:32.46codefreeze-lapInteresting: __ast_read: read() failed: Resource temporarily unavailable   -- in 1.4-- anybody else seeing this? the code is referring to to the alertpipe?
02:33.58drmessanoPalin in 0-12!
02:35.07[TK]D-FenderBeeBuu: do you see ${AGENTNUMBER}?
02:36.42BeeBuunothing
02:36.52lunaphytei'm just messing around with the web gui - make checkconfig indicates things look ok, but the web interface says "Nothing to see here. Move along."  how can i troubleshoot this?
02:37.16BeeBuu${AGENTNUMBER}* Agent number (username) set at login
02:37.40[TK]D-FenderBeeBuu: Perhaps only usable at the point where you login.
02:37.42BeeBuui don't know which is show the talking agent number...
02:37.56[TK]D-FenderBeeBuu: maybe what you are looking for does not exist at all.
02:38.23BeeBuuok,i know it now.
02:38.53BeeBuu[TK]D-Fender: did you know which is the correct one what i need?
02:38.54trnzmetaman why is palin fairing so low :(
02:39.12[TK]D-FenderBeeBuu: maybe what you are looking for does not exist at all. <----------
02:39.58BeeBuuO....it can't record the agent talking detail?
02:52.14drmessanoI voted for Ron Paul
02:54.47[TK]D-FenderBeeBuu: pastebin what you are trying.
02:58.40*** join/#asterisk DarkRift (n=dark@bas1-sthubert21-1096612081.dsl.bell.ca)
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03:09.30jayteeweird, the menus for my Xchat program disappeared.
03:10.23RB2jaytee, have you used Aastra phones?
03:10.48jayteeaha, menu bar was deselected. htfdth?
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03:11.01jayteeRB2, nope
03:11.52*** join/#asterisk [gnubie] (n=bintut@cm61.omega118.maxonline.com.sg)
03:12.22MiccBeeBuu, I think I understand what you want, but I don't know if asterisk provides that information, but it is available. A simple tweak should allow you to get the agent that answered from a variable.
03:13.27MiccBeeBuu, I believe theres also a cdr variable that tells what agent picked up the call.
03:13.38MiccAt least theres a config options for recording that in the cdr.
03:14.59[TK]D-FenderI want PROOF
03:15.19[TK]D-FenderRB2 if you have a questio, don't go fishing with it, just ask it out oud
03:15.22[TK]D-Fenderloud*
03:31.02*** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri)
03:35.35[TK]D-Fender*crickets*
03:36.17SwK[TK]D-Fender, everyones watching the election coverage
03:36.31[TK]D-FenderSwK: So am I... it called multi-tasking
03:36.40SwKhah
03:37.02[TK]D-FenderSwK: I tiled the MSNBC live video window, map, IRC, and a few others :)
03:37.33SwKhaha
03:37.34SwKnice
03:37.36*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
03:40.00*** join/#asterisk Gopher_77 (n=jim@akro-pool1-cs120.pool.dslohio.net)
03:40.57[TK]D-FenderSwK: OH.. and playing solitaire on my phone :)(
03:40.59Gopher_77can someone help me set up zaptel? zaptel says it can't open /dev/zap/ctl
03:41.49BeeBuuGopher_77: had you run  genzaptelconf?
03:42.04Gopher_77BeeBuu: not found
03:42.28BeeBuuwhich version is *?
03:42.52Gopher_77BeeBuu: ?
03:43.01Gopher_77BeeBuu: oh, the software
03:43.23*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
03:44.18Gopher_771.4.21.2
03:45.08BeeBuustep 1: install hardware, step 2: config the /etc/zaptel.conf and /etc/zapata.conf step 3:run zttcfg -vv & zttool check it is ok?
03:46.09Gopher_77BeeBuu: hardware installed, zaptel.conf configured from a tutorial I found on voip-info.org, zttcfg not found
03:47.54BeeBuuGopher_77: would you tell me how are you install the asterisk?
03:48.02Gopher_77BeeBuu: yum
03:48.13Gopher_77BeeBuu: fedora 8
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03:49.00BeeBuuhad you install the zaptel driver?
03:49.11Gopher_77BeeBuu: yes, with yum
03:49.41BeeBuuwhat's your card?
03:50.08Gopher_77BeeBuu: 2xfxo, 2xfxs, digium I think
03:50.38BeeBuuthe lights up?
03:50.59Gopher_77BeeBuu: what do you mean?
03:51.30jayteeyay, my image backup went well and my * box is again up and running and all tests were successful. It's time to head home and relax.
03:51.33BeeBuui'm a expert of fail~~~
03:51.37jayteebe back later
03:51.51Gopher_77BeeBuu: are the lights on the back supposed to light up when plugged in and not configured?
03:52.07Gopher_77BeeBuu: lspci is showing a device there
03:52.13BeeBuuGopher_77: if your driver is loaded.
03:52.30Gopher_77BeeBuu: hisax driver is loaded
03:52.44trnzmetaguys re: that "unable to request channel sip..." problem
03:52.54trnzmetayou're right, lag issue with provider
03:53.29BeeBuuGopher_77: check the prvite window
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03:56.24trnzmetayes!!! go ohio, go mccain
03:58.24drmessanoGod I hope you're kidding
03:59.29carrarMy neighbor has a Perot sign in his yard
03:59.39carrarPerot 92
03:59.57kerxMcCain seems to be winning
04:00.05drmessanoROFL
04:00.07kerx;)
04:00.08drmessanoNo he doesnt
04:00.10kerxI know
04:00.22kerxat least I made you laugh!
04:00.28kerxvery proud of himself
04:00.40kerxObama is president
04:00.42kerxfinal
04:00.44kerxbrb, TV
04:00.48drmessanolol
04:00.48kerxhe's on now @ chicago
04:00.49BeeBuukerx: really?
04:00.50drmessanono
04:00.54drmessanoThere is no final
04:00.59drmessanoit's projections
04:01.04trnzmetawhatever... if mccain wins ohio...  he's trailing 5%
04:01.12trnzmetathat's the bellweather state
04:01.17trnzmetago go mccain!
04:01.18kerxDude
04:01.21kerxIt's final
04:01.22ReDNeQITS OVER
04:01.24*** part/#asterisk Gopher_77 (n=jim@akro-pool1-cs120.pool.dslohio.net)
04:01.26kerxYep!
04:01.27ReDNeQJUST ANNOUNCED
04:01.30kerxNobody believes me
04:01.31kerxfor goodness sake
04:01.57drmessanoROFL
04:01.58trnzmetadude they haven't finished counting yet
04:01.59drmessanowow
04:02.02drmessanoSeriously
04:02.15drmessanoThey haven't closed the polls in like 8 or 9 states yet
04:02.27drmessanoThey havent announced jack
04:02.46drmessanoThis is the race for the media to be the first to call a winner right now
04:03.20trnzmetahahaha well... it's mccain by a nose
04:03.34*** part/#asterisk Howie69 (n=Owner@host-12-173-142-207.nctv.com)
04:03.47trnzmetaand palin by a breast
04:03.52trnzmetathey are calling it now
04:03.55trnzmetathey are calling it now
04:03.57trnzmetahmmm
04:04.00trnzmetacheck tv
04:05.43drmessanoHAW
04:05.46ReDNeQthe electoral college has been done
04:05.50drmessanoThey are calling it based on California
04:05.55ReDNeQOBAMA got 270 right now
04:05.58drmessanoand the polls JUST closed
04:06.03drmessanoNO HE HASNT
04:06.04ReDNeQCali/Ohio/Penns
04:06.16drmessanoCalifornia closed 7 minutes ago
04:06.24drmessano0% reporting
04:06.37drmessanoSheeple, please
04:11.01*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
04:11.05[TK]D-Fenderbaaaaaa baaaaaaaaa!
04:11.06trnzmeta292 prediction
04:11.16trnzmetatrial by media...
04:11.40[TK]D-Fendertrnzmeta: Sit this one out Nostradumbass :)
04:13.08trnzmetahahaha oh darn... obama will win
04:13.39trnzmetayou know... for sure some sicko will attemp to martydise obama
04:13.41*** join/#asterisk gones (n=gones@119.122.156.129)
04:14.02ReDNeQthat would be cool if at his big party someone launched a bag!
04:14.13ReDNeQthat would just top the medias appetite
04:14.36jayteecopies the chat including trnzmeta's last entry and forwards it to the Secret Service and FBI.
04:14.46trnzmetapfft I'm in australia
04:14.49trnzmetachase me if you can
04:14.55jayteelol
04:15.21trnzmetaI'm just pro mccain cause you guys are all lefty scum
04:15.22trnzmetamwhuauhaahu
04:15.36ReDNeQim for mccain
04:16.12jayteeCrazed Aussie arrested in plot to kill US President. "I'm innocent!!! I had nothing to do with this. That bastard jaytee in #asterisk set me up!"
04:16.32jayteegee, with a nick like ReDNeQ who'd have guessed.
04:19.05drmessanoI try not to listen to any of the rhetoric spewed forth by australia.. But I guess after they eliminate broadband in OZ and everyone will have to resort to using 19.2K baud connections over inmarsat
04:20.48trnzmetareboot time, my dialup is at it's 3 hr limit
04:20.57drmessanoexactly
04:21.18*** join/#asterisk jameswf (n=james@ip68-3-60-66.ph.ph.cox.net)
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04:26.42trnzmetathat's better, now I'm flying
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04:27.28*** part/#asterisk SplasPood (i=jwb@jwb.sh)
04:27.46*** join/#asterisk SplasPood (i=jwb@jwb.sh)
04:28.17garywsmithI just setup my first asterisk box.   Downloaded the CD Saturday.  It appears to be working fine.  Detects hardware, comes up in the browser.  The browser for IE7 seems screwy so I pulled it up in FF3 and I can create users but I can't delete them.  Read some faqs about getting latest copy of gui from svn but it isn't letting me.  Any ideas?
04:29.47riddleboxgarywsmith, did you use svn?
04:30.01garywsmithtried.  It gave me funky error;
04:30.14Kattyoh.
04:30.14riddleboxpastebin the error
04:30.15garywsmithas root:  svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui
04:30.15garywsmithsvn: REPORT request failed on '/svn/asterisk-gui/!svn/vcc/default'
04:30.15garywsmithsvn: REPORT of '/svn/asterisk-gui/!svn/vcc/default': 400 Bad Request (http://svn.digium.com)
04:30.22Kattywe're not going to hell in a handbasket.
04:30.34[TK]D-FenderKatty: not at the moment
04:30.47Katty:>
04:31.03riddleboxgarywsmith, http://asteriskNOW.org/install-related
04:31.27garywsmithok.  I'll try that
04:31.38Katty:>>>
04:32.14Kattydoes this mean no rioting? :>
04:32.41Gopher_77BeeBuu: are you there?
04:32.45garywsmithriddlebox: nope, gives me same error using that SVN link as well.
04:33.20riddleboxyou may want to ask in #asterisk-gui, and put bkruse in the sentence
04:33.32garywsmithwill do
04:36.07Katty[TK]D-Fender: i wonder what our receiptionist has to say about this situation.
04:36.13[TK]D-FenderKatty: Missouri isn't called yet...
04:36.28[TK]D-FenderKatty: Did you slap her for me earlier?
04:36.35Kattyno.
04:37.16[TK]D-FenderKatty: add it to the sting she'll be waling in with tomorrow :)
04:39.35Kattyand since it's all live.
04:39.42Kattywould it just be typical if someone got assassinated.
04:42.37[TK]D-FenderKatty: that would be the civil war you'd feared
04:42.46*** join/#asterisk Gopher_77 (n=jim@akro-pool1-cs120.pool.dslohio.net)
04:42.48drmessanoMass effectively legalized marijuana
04:43.03drmessanoThats interesting
04:43.06Gopher_77better to really legalize it
04:43.09[TK]D-FenderKatty: That is my current prayer.... that he survives it all
04:44.29drmessanoObama will be fine
04:44.50Gopher_77but what about us?
04:44.52drmessanoGives old racist southerners something to bitch about
04:45.00drmessanoQuiet Ohio
04:45.03trnzmetaeyes ReDNeQ...
04:45.10trnzmetabetter watch out for that one
04:45.24Gopher_77that one? hehe
04:45.38jameswfI can has my wealth now?
04:45.49Gopher_77me too
04:46.09Gopher_77i mean... can I have a piece of obama's wealth now?
04:46.56drmessanoYeah, yeah... Socialism socialism.. Taking tax breaks from the rich is called LEVELING THE FIELD.. Stupid right wing conspiracies
04:47.17jameswfmmmmmm government cheese makes good nachos
04:47.25Gopher_77actually no, socialism is where gov't owns parts of industry...
04:47.32Gopher_77look up marxism or communism ;)
04:47.58Gopher_77"From those as they have, to those as they need..."
04:47.58drmessanoLook up what the news has been calling Obama before trying to educate me, kthx
04:48.22drmessanoOh, sorry.. left off the ;)
04:48.29jameswfI am simply going to cling to my guns and god....
04:48.38Gopher_77me 2
04:48.48Gopher_77I like my morals and security
04:49.12jameswfwill go out and stock up the ammo b4 jan
04:49.34jameswfoh and bibles
04:49.38drmessanoObama is gonna take everyones guns, give them cooties, make them black
04:49.44drmessanoYep.. you caught him
04:49.50Gopher_77yes, code of laws helps too
04:50.07jameswfmy gun is already black
04:50.13jameswfboth of em
04:50.30Gopher_77well, only the "militia" is granted the right to bear arms in our constitution
04:50.57Gopher_77and in our constitution, the militia's officers and orders are assigned by the governors...
04:51.14jameswfwherever two or more are gatherd in the name of guns is a militia right?
04:51.47Gopher_77not as defined in our constitution, "the" militia
04:52.00jameswfbush will call martial law sometime early January
04:52.13Gopher_77and the bush empire will begin
04:52.33jameswfbush empire sounds like a porn
04:52.37drmessanoFrankly I am glad Obama won just so all the McCain loving biggots can STFU for a while.
04:52.38Gopher_77lol
04:52.40Gopher_77you're right
04:52.58Gopher_77hey, I never really liked mccain
04:52.58jameswfvoted Nader
04:53.11Kattywell it's not over till...
04:53.14Kattysomeone.. sings.
04:53.18Gopher_77i just liked his policies better than Obama... and he's the only candidate that really had a chance
04:53.25drmessanoYou wanna talk about bringing out the worst in people.. Welcome to the Republican party, circa 2008
04:53.28jameswfoh hey a fat lady
04:53.45trnzmetago go palin
04:53.53Gopher_772012!
04:54.01Gopher_77I can wear my shirt tomorrow!
04:54.04jameswfpalin's daughter sings
04:54.08jameswfheh
04:54.13drmessanoIn bed?
04:54.17Gopher_77palin's daughter is hot
04:54.20trnzmetawhilst... ahem?
04:54.28jameswfprego fettish?
04:54.29trnzmetanow that would be a cool shirt!
04:54.41drmessanoPalin's daughter uses Asterisk.. she loves the queue features
04:54.42trnzmetaerr crap... those videos came out?
04:54.45Gopher_77no, she's hot when she's not preggo
04:54.48Kattyi really wish i knew what our receiptionist was thinking.
04:54.56Gopher_77lol
04:55.00Kattyshe who thinks obama is the anti christ
04:55.05Gopher_77maybe I can get in the queue?
04:55.07Kattyher blog entry this morning was JUST so special.
04:55.10jameswfI videod my vote
04:56.07jameswfArizona has gained a senitor lost a governer fair trade
04:56.18Gopher_77ok so it's "from each according to his ability, to each according to his need..."
04:56.28drmessanoHere you go..
04:56.32drmessanoNot for the faint of heart...
04:56.33Kattyoh boy, almost time to see if someone gets assassinated!
04:56.42drmessanohttp://www.pastebin.ca/1245615
04:56.48drmessanoI got that e-mail today
04:56.54Gopher_77maybe this time it'll be a patriot on the grassy knoll
04:57.03Gopher_77and we'll have a super-gaffer for a president, nice
04:57.08drmessanoJust goes to show you that nothing in this country ever really changes
04:57.25jameswfkatty carefull that could be a theat
04:57.41Gopher_77oh yeah, remember that patriot act
04:57.42Kattyoh yes.
04:57.45Kattyvery threaty
04:57.54Kattyi must remember to watch my typing
04:58.00KattyDear Big Brother, please disregard above statements.
04:58.08jameswfwhat happens if one is assasinated before they are inaugurated?
04:58.08Gopher_77lol
04:58.15Gopher_77which one?
04:58.30Gopher_77probably vp takes over
04:58.49drmessanoVP would take over
04:58.58drmessanoHe's the president elect already
04:59.01jameswfI dont think there is a pre inaugeration hiearchy
04:59.15drmessanoBeyond VP, no
04:59.24jameswfso who iss VP palin?
04:59.28Gopher_77maybe all the racist old congressmen will keep him a lame duck for 4 years and there won't be any change
04:59.42garywsmithjameswf: depends.  If the delegates are cast, VP becomes president.  Otherwise delegates can pick anyone.
04:59.58Gopher_77so what if Palin was in it to take everything from the old dying man?
05:00.09drmessanoOh, give it two years.. we'll run the rest of those old coots out in 2010
05:00.45jameswfoh a gay class yay
05:00.59jameswfdisabled , not disabled
05:01.41jameswfoh my is that anarchy in the uk in the background
05:01.48Kattyyeah, that was cute.
05:01.57Kattyi do like the lesbian and gay bit tho
05:01.59Kattythat was a nice touch
05:02.35Kattyi also like the buddybuddying after all the bashing.
05:02.36drmessanoNotice that no one is booing when he mentions McCain and Palin?
05:02.48jameswfDamn no free tv's
05:02.55drmessanoBut McCain's supporters were booing like hell when McCain mentioned Obama
05:02.58drmessanoDamn racists
05:03.10jameswfnotice no biden on stage
05:03.12Gopher_77that's b/c they like mccain too
05:03.21Gopher_77lol yeah
05:03.30drmessanoNah it's called having class
05:03.38drmessanoSomething republicans know nothing about
05:03.42drmessanoClass, respect
05:03.47Kattyoh boy
05:03.50Kattynew puppy in the white house
05:03.54[TK]D-FenderI just heard!
05:03.55Kattyi wonder how that's gonna go
05:04.04Kattybetter not be MY puppy
05:04.15garywsmithhey, how do I setup outgoing calls on a new install?
05:04.15jameswfpuppy peing on American history
05:05.06drmessanopeer is a liberal
05:05.14jameswfgarywsmith:  cant you see we are a country in crisis and your worried about phone calls
05:05.18jameswf:) jk
05:05.48garywsmithI understand that.  Denial is the first step.  I'll get through it in four years...
05:06.04*** join/#asterisk Gopher_77 (n=jim@akro-pool1-cs120.pool.dslohio.net)
05:06.23Gopher_77so why would anyone pick someone to represent him to congress who gaffes like biden?
05:06.42[TK]D-FendergrayYou have a LOT of learning to do.  The GUI framework is a patchwork monster
05:06.45jameswfagain notice no biden on stage
05:06.53[TK]D-Fendergarywsmith: You have a LOT of learning to do.  The GUI framework is a patchwork monster
05:07.09garywsmith[TK]D-Fender: yes, I see that now...
05:07.09jameswfoh crap i hear echos
05:07.11[TK]D-Fenderjameswf: not YET...
05:07.53jameswfI just cant wait for my check to arrive
05:08.02jameswfoh crap he said 2 terms
05:08.32Gopher_77presumptuous
05:08.53Gopher_77this is what the dr ordered
05:09.46jameswfas soon as he gets out without a teleprompter as president he will be another bush as far as speaking..
05:09.58Gopher_77yep
05:10.05Gopher_77another 8 years of george bush :)
05:10.31Gopher_77I wonder if an ex-president can hold vp
05:10.41drmessanoHa.. the asian markets are up 2 - 6% since the announcement
05:10.48jameswfif he takes plays from years 2-6 of clinton he will get 8
05:11.06Gopher_77yeah, they've been investing in the democratic party for decades
05:11.24trnzmetadamn asians
05:11.45trnzmetaalways going against mccain
05:11.51drmessanoYeah, I dont see why they wouldnt.. Republicans screw up the economy so bad here, the other world markets tank with it.. I would invest democrat too.
05:12.02jameswfso countdown to next terror atack in the next 6 months
05:12.03Gopher_77and years 6-8 will lead to another mortgage meltdown
05:12.15Kattycome on guys, cheer up a bit
05:12.19Kattybe happy for 10 minutes (=
05:12.34Kattythere's enough sucky in the world to go around without reaching out and asking for more.
05:12.49jameswfoh look Jan 21 2009 the first "re-education" centers open here in Arizona
05:12.55Gopher_77it'll come at us whether we like it or not
05:13.02Gopher_77just some of us will be prepared when it happens ;)
05:13.06Kattydoesn't care
05:13.10Kattyis going to be happy for a few minutes.
05:13.19garywsmithHere is what I have.  A new install with a two port FXO card in it.  I have setup x-lite on my workstation, configured sip.conf, and I can see it registering when I start x-lite.  I can call into the general mailbox, I just get a file not found error when dailing out.
05:13.26*** join/#asterisk xacatecas (n=jkroon@dsl-240-156-31.telkomadsl.co.za)
05:13.27jameswfKatty huffing is bad mmmmmkay
05:13.40Gopher_77lol
05:14.11jameswfYES   WE CAN
05:14.13jameswfYES   WE CAN
05:14.14jameswfYES   WE CAN
05:14.22jameswfyay
05:14.28Gopher_77I thought that was katty pee
05:14.41xacatecashi all, a query related to BLF ... whenever * get's issued a reload the sip subscription info is lost, on our GXP20?0 phones at least there is no resubscribe - is there a way to directly after a reload send some SIP/HTTP request to the phone to request it to re-register?
05:14.45drmessanoI don't see what all the fuss is about the economy anyway.. McCain said the economy was strong
05:14.46[TK]D-Fenderjameswf: Good old rule of 3.  Worked for Ceaser, Nixon, and now Obama
05:14.49drmessanoI for one believe him
05:15.03[TK]D-Fenderjameswf: "read my lips"   "no new taxes"   "we the people"
05:15.05Gopher_77fundamentals of the economy
05:15.09jameswfbut not for four?
05:15.15Gopher_77but even fundamentals can be changed
05:15.23[TK]D-FenderGopher_77: .. that was Mccain
05:15.40Gopher_77[TK]D-Fender: yes
05:15.56trnzmetayou know what... I wonder if amercia/middle america has any idea how much the world disapproves of american policy in general
05:16.00jameswfwe are one we are borg assemilate
05:16.14Gopher_77we could all be made lazy like mexicans or iraqis, then our economy will be screwed
05:16.19jameswfnot god bless america god  damn america
05:16.21drmessanoHA
05:16.47drmessano"fundamentals of the economy" <--- Yeah, that didn't really mean "The economy" .. it was so taken out of context
05:16.53[TK]D-Fenderjameswf: America has always been pretty good at damning itself.  Its forked for so long already...
05:16.56Kattynice manly hug.
05:16.58drmessanoWhatever lol.. I'm so glad you right wingers get to eat it now
05:17.14xacatecastrnzmeta, even if they know, should they care?
05:17.15drmessanoThe months of doublespeak and BS are over
05:17.19[TK]D-FenderRP 2008!
05:17.33[TK]D-Fenderruns in circles.
05:17.47jameswfvoter fraud and 1 billion spent he would feel pretty stupid if he lost
05:17.49[TK]D-Fenderin the meantime I'll celebrate that the significantly lesser evil won ;)
05:18.05*** join/#asterisk neobsd (n=neobsd@190.81.184.1)
05:18.07neobsdhi
05:18.07Gopher_77maybe Obama will be caught in voter fraud with Acorn and get the boot
05:18.09Kattycan you imagine the next oprah show?
05:18.16Kattyshe's going to be giving away houses to boost the economy
05:18.22jameswfyay we will get canadian style health care
05:18.24neobsdsome help please ?
05:18.34neobsdi need configure my trunk with iax2
05:18.44neobsdi can call from A to B
05:18.49Gopher_77anybody have a tutorial for settng up zaptel drivers?
05:18.52neobsdbut i can't from B to A
05:18.57neobsdplease ...
05:19.03neobsdi have this messages
05:19.04jameswfoh look time to get healthcare in mexico
05:19.16Kattywell
05:19.19Gopher_77tequilla?
05:19.20Kattyno assassinations
05:19.22drmessanoMaybe we'll luck out and Obama will sell Ohio to Canada
05:19.23Kattyi guess i can go to bed now
05:19.24slingrhttp://i128.photobucket.com/albums/p185/JC125_photo/epiccombobreaker.jpg
05:19.27Gopher_77oh, that's what they have in mexico...
05:19.31neobsdsay     NOTICE[4512]: chan_iax2.c:7613 socket_process: Rejected connect attempt from 10.3.251.58, who was trying to reach '8640@'
05:19.32Kattynight all
05:19.35slingrnight katty
05:19.52jameswfdrill alaska dry and pass it back to russia
05:19.54Gopher_77maybe hire one of those illegal immigrants to do the job
05:19.56neobsdand    WARNING[4513]: chan_iax2.c:7823 socket_process: Call rejected by 10.3.251.58: No authority found
05:19.56neobsd<PROTECTED>
05:20.00neobsdsome idea ?
05:20.10Gopher_77yeah, palin's not going back there :)
05:20.22xacatecas*sigh* ... and so #asterisk too is not about * today ...
05:20.25xacatecashi all, a query related to BLF ... whenever * get's issued a reload the sip subscription info is lost, on our GXP20?0 phones at least there is no resubscribe - is there a way to directly after a reload send some SIP/HTTP request to the phone to request it to re-register?
05:20.42jameswfPalin the next white Oprah
05:20.46*** join/#asterisk subdolus (n=subby@subby.afraid.org)
05:20.46neobsdsome help please ?
05:20.55jblackDoncha know, she woulda picked out nice drapes.
05:21.06jameswfoh yah
05:21.11drmessanoTo match the carpet?
05:21.15Gopher_77yeah, but she'd poke holes in the carpet
05:21.41jblackcarpet? Bah. She woulda ripped that out and replaced it with carribu skins.
05:22.01Gopher_77because she loooooves animals
05:22.10jameswfWell with obama in palins daughter can go get rid of that pesky bby up to the 40th week
05:22.38Gopher_77didn't Obama vote for partial-birth abortions?
05:22.59jblackI know he voted for fisa _and_ the bailout.
05:23.10Gopher_77and liked it
05:23.37jblackBut yeah, I think he did, because there parts of the law he disagreed with.
05:24.22jblackI wonder how long he'll be pres before he gets assassinated.
05:24.48Gopher_77I wonder how many people are uttering the "A" word right about now
05:24.55drmessanoHe didnt vote for it
05:24.59jameswfjblack: he will get a pope mobile
05:25.08Gopher_77oh yeah, he votes present doesn't he?
05:25.19jameswfhttp://www.ontheissues.org/Social/Barack_Obama_Abortion.htm
05:25.26drmessanoHe OPPOSED the entire law because parts would LIMIT womens rights
05:25.36drmessanoSomething the republicans do already, even in absentia
05:26.05jameswfgod our governer is such a ehhh
05:26.10drmessanoBut I guess thats the same as voting FOR it, right..
05:26.11andrewyhe wanted a provision that would allow them if the mother's life was in danger
05:26.22andrewythe law that was passing was banning them with no such provision
05:27.07*** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri)
05:27.29Gopher_77jameswf: who's your governor?
05:27.39jameswfNapalitono
05:28.02[TK]D-FenderGesundheit ;)
05:28.09Gopher_77lkol
05:28.18jameswfshe will probably become the AG or homeland security head
05:28.35jameswfso rumors say
05:28.43Gopher_77maybe it'lll be a good thing to get another governor?
05:29.36jameswfas I said we get mc cain back and loose the governer fair trade
05:30.21jameswfJohn McCain a real Maverick a republican without money
05:30.57Gopher_77doesn't his wife have enough of that for the both of them?
05:31.35jameswfshe looked like a mustard bottle tonight
05:31.41Gopher_77lol
05:31.44Gopher_77i didn't notice
05:33.20drmessanoHow many houses does McCain have?  Oh yeah, so many he forgot
05:33.32drmessanoI can relate to him, as can everyone
05:34.16jameswfmccain doesnt have any
05:34.54jameswf'Change has come to America' <<< wow really? where the man isnt even in yet
05:34.57Gopher_77nah, he's just old and losing his memory
05:35.12drmessanoSorry, he's got condos
05:35.19drmessanoBut not sure how many
05:35.31Gopher_77maybe people will be upset with the lack of change by jan
05:35.47Gopher_77would be about the intelligence of america nowadays
05:35.53jameswfdrmessano:  Mccain has no money cindy goes away he is screwed
05:36.04Gopher_77yeah, he probably has investment condos his wife bought
05:36.11drmessano"McCain owns seven homes, with a total worth of $13 million"
05:36.22drmessanoObama has one $1.65 Million dollar home
05:36.29jameswfNo Budwieser owns 7
05:36.47Gopher_77that's owned by foreigners now
05:36.49drmessanoI see
05:36.51Gopher_77not cindy
05:37.24jameswfcindy = big money mccain= lucky bastard who met the right daughter
05:37.49drmessanoYeah.. Cause that makes a difference
05:38.19drmessanoHe's been married to her for 25 years, but it's not "his" money.. he's John the Pilot
05:38.33drmessanoI buy it.. Matter of fact, make mine a double
05:38.36Gopher_77cindy = big money mccain = man with free beer = lucky bastard who met the right daughter = the man we all want to be
05:40.02Gopher_77so... i have hardware installed and ztcfg says it can't open /dev/zap/ctl (doesn't exist). how do i get it to appear?
05:40.05jameswfdrmessano: you probably couldnt imagine the contracts dude probably had to sign to get in cindy's pants
05:40.20jameswfGopher_77: modprobe
05:40.34Gopher_77jameswf: modprobe zaptel?
05:40.46Gopher_77jameswf: module not found
05:40.51jameswfzaptel + modulles
05:41.09jameswfthere you go
05:41.19Gopher_77jameswf: I have zaptel installed, idk why it doesn't find the modules
05:41.31jameswfdepmod -a
05:41.48Gopher_77jameswf: nothing
05:43.30Gopher_77jameswf: I do locate zaptel and it only shows in /etc and /usr/share/doc
05:43.54Gopher_77jameswf: using fedora 8 and yum to install zaptel
05:45.19jameswfyou need zaptel-modules or kernel-modules-zaptel
05:45.57*** part/#asterisk garywsmith (n=garywsmi@c-67-164-10-244.hsd1.ca.comcast.net)
05:47.21Gopher_77jameswf: don't have any module packages
05:53.39*** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri)
05:56.07*** join/#asterisk obnauticus (n=lol@about/windows/regular/obnauticus)
05:57.19[TK]D-FenderWell this has been abig night...
05:57.31[TK]D-Fendertime to hit the sack and sleep a little better...
05:57.57[TK]D-Fenderlater all
06:06.07*** join/#asterisk SanityIO (n=SanityIO@77.242.106.191)
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06:09.59*** join/#asterisk tobias (n=tobias@user-0ce2hvs.cable.mindspring.com)
06:14.42*** part/#asterisk BigLuks (i=irssi@62.141.56.213)
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06:17.06prodyanhello all
06:19.19kerxhi
06:21.05JohnnyBeGoodWhat everyone thinks of new President?
06:21.13Gopher_77boo!
06:21.33Gopher_77the republican pres after him will have a cleanup job
06:21.34jetsHooray.
06:21.34JohnnyBeGoodI hope it'll move in good direction
06:21.38zr0=D
06:36.22*** join/#asterisk kamanashisroy (n=kamanash@202.56.7.182)
06:38.19Gopher_77well, looks like fedora refuses to supply a kmod for zaptel
06:39.18Gopher_77is that something that is distro-specific?
06:41.13prodyanhow do i call my asterisk server?
06:47.11*** join/#asterisk mintos (n=mvaliyav@nat/redhat-in/x-d4e18602f4a80d8f)
06:53.01prodyanbrb
06:55.24*** join/#asterisk prodyan (n=asterisk@www.alliance.com.ph)
06:55.29prodyanback
06:57.26prodyan[Nov  5 14:54:22] NOTICE[9788]: chan_sip.c:15055 handle_request_register: Registration from '"banks"<sip:banks@192.36.253.168>' failed for '192.36.253.134' - No matching peer found
06:57.43prodyancan anyone help me how to call?
06:57.51*** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com)
06:58.29tzafrir_laptopGopher_77, Fedora only has the userspace stuff of Zaptel
07:00.18prodyanguys how do i add an sip user to asterisk?
07:01.03Gopher_77tzafrir_laptop: yes
07:01.45Gopher_77tzafrir_laptop: could I get a package/source for it?
07:02.22tzafrir_laptoperr... maybe install from source :-(
07:02.58tzafrir_laptopI'm not sure how to install the kernel-space part of that
07:02.59Gopher_77yeah, it's sad, and the first time I would have to do it since vmware
07:03.16Gopher_77and only
07:10.06prodyananyone?
07:11.16prodyanoki nvm i just found all .confs in /etc
07:12.12*** join/#asterisk ming_zym (n=ming_zym@nat/yahoo/x-6f4336d5f8375ef9)
07:12.57*** part/#asterisk bertinskiz (n=bertinsk@rrcs-67-78-186-27.se.biz.rr.com)
07:14.49trnzmetaguys: I'm currently using asterisk 1.2, needing to upgrade
07:14.57trnzmetahow stabel is 1.6?
07:18.00*** join/#asterisk Ritzerisk (n=Ritztech@nv-65-40-156-46.sta.embarqhsd.net)
07:20.13trnzmetaoh still beta testing
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07:26.48write_eraseHi .... Is there a RNIS expert here ? When calling one DDI number  to my  2xT0 line, all phones ring twice (2 channels) .... Can someone have a look to my misdn.log ?
07:29.12write_erasehttp://www.aixmarseille.com/pub/misdn.log
07:30.57Ritzeriskim trying to see if someone could help me make sense of a macro of ssnoservice
07:33.16*** join/#asterisk af_ (n=getsmart@88-149-230-65.dynamic.ngi.it)
07:33.33Ritzeriskanyone know what this means
07:33.34Ritzeriskexten => s,n(notzap),Goto(from-pstn,${DID},1)
07:34.23Ritzeriski know its like a if notzap go to the from-pstn context whch that then takes me to s-service
07:40.56*** join/#asterisk error404notfound (n=shoaibi@58-65-160-128.nayatel.pk)
07:41.45creativxnotzap is the label name
07:41.53creativxwhich identifies the line and can be goto'ed to
07:42.12creativxthe goto command jumps into context from-pstn at extension ${DID} priority 1
07:48.29*** join/#asterisk chigital (n=chigital@services.mivitec.net)
07:49.34*** join/#asterisk ayrjola (n=ayrjola@support.ccxtech.fi)
07:50.06*** join/#asterisk ManxPower (n=manxpowe@223.sub-70-221-254.myvzw.com)
07:54.47*** join/#asterisk mvanbaak (i=vanbaak@asterisk/contributor-and-bug-marshal/mvanbaak)
07:58.14Ritzeriskhmm basically i go to a no server and all my contexts from the zaptel channels are going to from-zaptel
07:58.41Ritzeriski put if any DID/CID go to a specific location (ring group)
08:01.45Ritzeriskbut its like i get 2 calls then on the 3rd call since 701 and 702 is in use it starts the from-pstn then it says ss-noservice to me
08:02.50aiksa[LV]oslec works like a charm with redfone e1 adapter
08:02.54aiksa[LV]:)
08:03.44aiksa[LV]although redfone had hw echo canceler it still had a problems with echo in some specific cases. now with the adding of the oslec the echo is gone :)
08:05.34aiksa[LV]so it looks like ztd_ethmf is able to use OSLEC wo any additional patching.
08:05.47*** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr)
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08:20.28prodyanguys which config file do i change so that xlite can dial in to my asterisk?
08:21.53mort_gibsip.conf and extensions.conf
08:22.40prodyanahh thanks mort
08:23.03mort_gib~thebook
08:23.03jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
08:23.16mort_gibGood examples in the book :-)
08:23.49prodyan:D okey ill take a look at there also
08:29.01kerxi heard asterisk cookbook coming out soon
08:29.04kerxcan't wait :)
08:29.09*** join/#asterisk mikkel (n=mikkel@130.226.36.170)
08:29.50Zhadcan't wait to see everything written down three times so there's a compatible version in there for different releases of asterisk?
08:32.06kerxno, i would probably be more interested in the later stable release
08:32.17kerxbut that would be helpful
08:32.25kerxif Bruce decides to do such a thing
08:32.30kerxwould cost O'Reilly lots for print though
08:34.11drmessanoheh
08:34.31drmessanoI heard the asterisk cookbook was dead as a hammer.. Where did you get your info?
08:35.22Zhadinteresting, 1st Ed of The Future of Telephony is £5 more than the 2nd ed at amazon.
08:35.51drmessanoWell, it's out of print
08:41.01tzafrir_laptopZhad, it has less pages
08:41.17tzafrir_laptopso maybe it evens out on the delivery
08:41.46aiksa[LV]tzafrir_laptop: FYI (if you'll ever need it) ztd_ethmf plays very well with oslec
08:45.34*** join/#asterisk menschentier (n=wolf@pD9E06973.dip.t-dialin.net)
08:46.10menschentiergood morning
08:47.07*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
08:48.21aiksa[LV]morning.
08:48.32*** join/#asterisk PumPom (n=RDN@user-5446dbdd.lns5-c13.telh.dsl.pol.co.uk)
08:49.24aiksa[LV]I was wondering of another thing: lets say I had an asterisk box with mic and speakers connected to the soundcard and partly using this ast as the softphone
08:49.41PumPomGuys iv got a two way voice conversations being recorded on a VoIP logger, when you playback the recording, you can hear both sides but the quality of one side is very poor. I'm assuming the packets are being lost somewhere?
08:50.44aiksa[LV]what could i do with the echo here (caused by the audio going from the speakers into the mic)?
08:50.46phpboyhi
08:50.52phpboyi keep getting this error
08:50.55phpboy[Nov  5 10:50:08] NOTICE[25471] chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4
08:51.27menschentierjust to honor the wisdom of all american people - put their anthem in your moh as i did
08:51.29kaldemaraiksa[LV]: use a headset instead of speakers and a mic
08:51.35phpboyI've tried 3 diff cables and called the telco to check it out and they say it's fine.
08:51.45aiksa[LV]kaldemar: i am not as stupid as that :)
08:51.46*** join/#asterisk gr00t (n=the_html@82-38-252-250.cable.ubr01.shef.blueyonder.co.uk)
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08:52.08aiksa[LV]kaldemar: its exactly the case where I wanted not to use headset or handset
08:52.17*** join/#asterisk gr0mit (n=tim@dhcp3.zuk40.mot-tools.co.uk)
08:52.22menschentierhttp://www.star-spangled-banner.info/ royalty-free mp3s to download...
08:53.14aiksa[LV]kaldemar: i could of course "loop" this call through zaptel hardware first before sending it into PSTN, thus kicking in the ec
08:53.35gr00tMorning (to those around GMT-ish). Tearing my hair out trying to debug a "channel unable to hangup" message in IAX trunk outbound calls (sip -> * IAX -> Internet -> * IAX -> PSTN) - anyone have any ideas? running asterisk 1.6
08:53.50phpboyAnybody got an idea?
08:55.09tzafrir_laptopmenschentier, where on that site does it say that those mp3s are royalty-free?
08:55.15PumPomanyone can help me?
08:55.49*** join/#asterisk datacompboy (n=datacomp@213.187.250.81)
08:56.20menschentierAs far as we understand, none of the items on this site are restricted by copyright law. In which case, you are welcome to use any of them. If you happen to have any resources you'd like to share with others, feel free to  .
08:56.28menschentierthats what they say...the recordings are old
08:56.38menschentiershould be on the safe side
08:56.46aiksa[LV]"as fas as we know..." khem
08:57.02menschentieryah...but a recording from....1914...
08:57.15PumPomanyone any help?
08:57.19tzafrir_laptopmenschentier, if this is not stated explicitly, be careful with "what you understand"
08:57.22aiksa[LV]phpboy: what type of line? T1?
08:57.28phpboyE1
08:57.35menschentieri will use it just for this day
08:57.42menschentierim german anyhow....
08:57.46menschentier:)
08:58.12menschentierusually my asterisk is playing the bsd songs
08:58.19aiksa[LV]phpboy: whats the status of this span in zttool?
08:58.19tzafrir_laptopsome people tend to interpet the fact that a song was digitized as good enough reason to make it a new (though derived) work of art
08:58.28menschentieri know
08:59.43phpboyaiksa[LV]: alarm: OK / no alarms
09:01.09aiksa[LV]the calls go through ?
09:02.39menschentierbut - i also have a real topic, if anyone can help me...i use my nokia e71 as a wireless sip phone when i am at home. problem is, when i leave the house, asterisk does not recognise this and my voicemail system does not come on at once but rings the non-present phone - any chance to tell asterisk to check the registration state of this device more often?
09:02.46phpboyFrom what i can tell it's not droping calls, but let me test it explicitly
09:03.45phpboyaiksa[LV]: getting this too, chan_zap.c: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 4
09:04.55aiksa[LV]its the same one you pasted above
09:05.00phpboywoops
09:05.16phpboychan_zap.c: PRI got event: HDLC Abort (6) on Primary D-channel of span 4 <--- this too
09:05.23phpboyjust these two msgs :(
09:05.36aiksa[LV]phpboy: i have seen these on my e1 lines just right after e1 links is established
09:06.14aiksa[LV]not like regular appearances
09:07.57*** join/#asterisk UtopiahGHML (i=utopiah@gateway/shell/blinkenshell.org/x-499bc50f17992b3e)
09:08.01UtopiahGHMLhi #asterisk
09:08.38aiksa[LV]phpboy: as regards the cables for e1 i have not seen many possible options: just two cases
09:08.49aiksa[LV]but its not a cabling fault in your case
09:09.09aiksa[LV]if it was due to the cable you wouldnt have OK status for span.
09:09.43UtopiahGHMLI would like to do that http://seedea.free.fr/wiki/index.php?n=Seedea.Input#VoiceBasedInput but Im not sure how to do it, is it easy to do? am I considering it with the right angle? estimation for handling 100 users? 1000?
09:10.10UtopiahGHML(basically it's Speech Recognition relying on VoIP connected to "normal numbers")
09:11.05aiksa[LV]I am more interested in this message: chan_dahdi.c:8706 pri_fixup_principle: Call specified, but not found?
09:11.24aiksa[LV]smthg with signaling on the far end?
09:13.17*** join/#asterisk datacompboy (n=datacomp@213.187.250.81)
09:14.08aiksa[LV]what happens is that after  pri_fixup_principle: Call specified, but not
09:14.09phpboyaiksa[LV]: that's the thing, I thought it it might be the cable, hence changing it 3 times, and still no change.
09:14.40aiksa[LV]phpboy: thats not a cable. as i said it was cable the status for span would be red
09:17.18aiksa[LV]what does pri_fixup_principle does anyway?
09:17.54aiksa[LV]it looks to me that after this messages appears in the asterisk, EC gets disabled for all calls untill there is another new call
09:17.58aiksa[LV]could this be so?
09:18.03phpboyaiksa[LV]: I've got a call connected to my mobile at the mo
09:18.08phpboyno disconnect :/
09:18.29aiksa[LV]so it works?
09:18.41phpboyseems so, with no disconnect
09:18.56phpboyI'm keeping it connected to check if it bombs out
09:18.56aiksa[LV]but the errors still keep popping?
09:19.09aiksa[LV]sorry, not errors, warnings
09:20.14phpboyin /var/log/asterisk/messages, funny enough, zaptel itself doesn't complain in /var/log/messages
09:22.14datacompboyHi all!
09:22.55datacompboyI have asterisk 1.4.21, installed from aptitude. two TE210P cards, 3 E1 cables. Today's morning hangs up with 100% cpu eat
09:23.04datacompboyOnly kill -9 unhangs system
09:23.13datacompboyi was unable to even connect to it with asterisk -r
09:23.24tzafrir_laptopaiksa[LV], pri_fix_principle is: "make sure I have the right B channel"
09:23.54aiksa[LV]tzafrir_laptop: hmm.
09:24.07aiksa[LV]so no traffic on the designated b chan
09:24.12aiksa[LV]?
09:24.14tzafrir_laptopdatacompboy, asterisk is 100% cpu?
09:24.40tzafrir_laptopdatacompboy, is asterisk is 100% cpu but does not hang the system: are you out of disk space?
09:25.09datacompboytzafrir_laptop: yes, asterisk eats 100%, no incoming no output, system are unreliable -- i was able to log into only via DRAC, SSH not working
09:25.27tzafrir_laptopDRAC == ?
09:25.28UtopiahGHMLanybody here used Voiceglue?
09:25.48datacompboytzafrir_laptop: avail 56Gb of free space. DRAC = Dell Remote Access Controller. remote keyboard+monitor
09:25.51tzafrir_laptopdatacompboy, what did you see in the asterisk logs?
09:26.05aiksa[LV]tzafrir_laptop: so the way to proceed would be with "pri intense debug ..."
09:27.01datacompboytzafrir_laptop: i'm unsure. there lot, mo -- i'll pastebin intresting messages
09:28.06tzafrir_laptopdatacompboy, it could also be a loop in the dialplan
09:28.37datacompboytzafrir_laptop: dialplan simple: get on zaptel and forward to sip, and get from sip and forward to zaptel
09:28.45ana_michoHi all,Is there any new version of asterisk that supports VAD?
09:29.04tzafrir_laptopdatacompboy, in /etc/default/asterisk unrem AST_REALTIME=no
09:29.22tzafrir_laptopthis should prevent a 100% cpu loop of asterisk from hanging the system
09:29.30tzafrir_laptopand allow you to debug it
09:29.39*** join/#asterisk Aces1up (n=Now@ip70-173-52-152.lv.lv.cox.net)
09:30.33tzafrir_laptopgenerally in such a case if you get no log messages, try pressing 'H' in top to get the pid of the relevant thread and strace it
09:31.03datacompboytzafrir_laptop: http://pastebin.ca/1245744 -- here is log messages
09:31.37*** join/#asterisk IsUp (n=nocturne@unaffiliated/isup)
09:31.58datacompboytzafrir_laptop: how to disable realtime without restart? renice will do work?
09:33.35datacompboytzafrir_laptop: 17646: old priority 0, new priority 0 -- so, it not realtime alredy?
09:33.43tzafrir_laptopyou need chrt
09:36.27datacompboywhat need to set? left in RR but set prio to 0 ?
09:38.54datacompboyok, done - removed
09:39.25datacompboybut still question what to do? its real system, just old one have hardware failure, so transferred to new server. all soft installed from scratch
09:39.55datacompboylast time i got 100% cpu from asterisk -- it was broken appconference. that time i have no appconference -- only sip and zap with simple forward dialplan
09:40.20*** join/#asterisk neizd (n=neizd@host-n4-89-7.telpol.net.pl)
09:40.30neizdhello everyone
09:40.36neizdI need help with asterisk
09:41.22menschentieris here a dishwashing channel...?
09:41.31menschentieri would need some help at this...
09:41.32*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
09:41.40*** join/#asterisk The-Bat (n=The-Bat@203.199.114.33)
09:41.48IsUpBWAHAHA
09:41.58creativxtits?
09:42.10neizdI have server for voicemail it does record files to /var/spool/asterisk/default/{number}/INBOX/ but does not send mails - where should I check where problem is? Don't even know what for google search?
09:42.23*** join/#asterisk blogbasti (n=blogbast@calypso.planet-ic.de)
09:44.59neizdSo I can't find any help in here?
09:45.47datacompboyneizd: have you setup cron to send mails?
09:47.12tzafrir_laptopdatacompboy, there are no really crazy error messages there
09:47.30tzafrir_laptopNothing that indicates a frantic 100% cpu loop
09:47.36neizddatacompboy: will check it in a second
09:47.42*** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290)
09:48.33*** join/#asterisk KermitTheFragger (n=KermitTh@118-197.bbned.dsl.internl.net)
09:48.50datacompboytzafrir_laptop: yes, this why i get run irc :)
09:49.42neizddatacompboy: nope, there's nothing - tthanks I think that I'll manage to do it now
09:51.47neizddatacompboy: can you help me further and give me the command that will send it all? so I can place it in my crontab?
09:52.40datacompboyneizd: i'm not sure :) i just know that if you need to send something, something must that do. what asterisk? where did you configure where to send? etc
09:52.47IsUpi dont think you need crontab.
09:52.56IsUpwait a min neizd
09:53.09neizdIsUp: ok, will wait
09:53.39*** join/#asterisk propellerhead (n=yogurt2u@host38.190-136-115.telecom.net.ar)
09:53.42IsUpcan you paste your voicemail.conf to pastebin?
09:54.09[gnubie]gtg..
09:54.25neizddatacompboy: in voicemail.conf asterisk have a configuration line sendvoicemail=yes so that's why I haven't think that It'll need any cron job to do this ;-)
09:54.57IsUpneizd: can you paste your voicemail.conf please?
09:56.17datacompboyneizd: o! sorry -- last time i have setup voicemail, i have manually send them with cron :)) well, where do you send it? try grep DESTEMAIL /var/log/mail*; where DESTEMAIL is e-mail where you try send voicemail
09:57.13neizdIsUp: http://pastebin.com/m23b7f89f
09:59.40IsUpneizd: can you send mail over your server? have you ever tried?
09:59.47datacompboytzafrir_laptop: o -- again hangs
09:59.53IsUpmaybe theres something wrong with your 'sendmail'
10:00.06*** join/#asterisk steliosk (n=Stelios@ipa107.2.tellas.gr)
10:00.26datacompboytzafrir_laptop: hm. set chrt -p 1 `pidof asterisk` -- it still hangs system
10:00.47datacompboyhow to remove realtime at all? i have hangs system now -- but can slowly do something -- what to see ?
10:00.47tzafrir_laptopI hope you edited /etc/default/asterisk as well
10:01.06datacompboyyes, /etc/default edited, but asterisk was not restarted -- live system
10:01.24tzafrir_laptopnothing in the logs, right?
10:01.48datacompboytzafrir_laptop: mo, waiting for "ps -fax" output... just kill asterisk?
10:01.55neizdIsUp: yes I can
10:02.10tzafrir_laptopps fax won't show you threads
10:02.14neizdIsUp: sendmail works there perfectly
10:02.52tzafrir_laptoprun top, and press 'H'
10:03.06IsUpneizd, try uncommenting 'mailcmd=' in voicemail.conf
10:04.06neizdno mailcmd in there
10:04.59datacompboytzafrir_laptop: how to remove Realtime from running? chrt -o -p `which asterisk` ?
10:06.17tzafrir_laptopcat /var/run/asterisk/asterisk.pid
10:08.39datacompboywell, i can't do anything :(
10:08.44datacompboydoing power cycle
10:08.52datacompboyevent Ctrl+Alt+Del no effect
10:09.50tzafrir_laptopheavy swapping?
10:10.00tzafrir_laptopa process in D state?
10:10.57datacompboytzafrir_laptop: can't see :) fully stuck
10:11.09neizdIsUp: added mailcmd and still nothing :(
10:11.46tzafrir_laptophave you tried alt-sysrq commands?
10:12.02datacompboytzafrir_laptop: i can't send alt+sysrq from drac :( alredy tried
10:13.57datacompboyok, at least it now started without realtime
10:14.18datacompboyso on next hang i'll be able to see. what to do -- just gdb to hanging thread?
10:14.48*** join/#asterisk nfi|ermes (n=ErMeS@217.220.121.62)
10:15.20nfi|ermeshi all
10:15.22nfi|ermes[Nov  5 12:16:35] WARNING[6452]: app_dial.c:1242 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
10:15.26datacompboytzafrir_laptop: also, there cthreads on system, not pthreads -- so i see no threads
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10:19.45tzafrir_laptopnfi|ermes, and your Dial() line was?
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10:20.11tzafrir_laptopdatacompboy, before a full-fledged gdb, I'd try strace
10:20.38tzafrir_laptopbut from what you descript, I suspect something different hanging the system
10:20.38datacompboytzafrir_laptop: ok, installing strace now - while everything alive:)
10:20.56datacompboynope, 100% cpu eats asterisk, while in realtime.
10:21.01datacompboynow it not realtime
10:22.07neizdIsUp: Thanks for your help! Problem was with asterisk on the mail-gateway! ;-)
10:22.18neizddatacompboy: thanks for your help too :)
10:22.36datacompboyneizd: for me absolitely no for :D
10:23.20*** join/#asterisk unasi7 (n=unasi7@84-75-23-200.dclient.hispeed.ch)
10:24.07unasi7hi. is there a propperty on which quality/distance (p.E. 300ms) a client will be identified as offline/unreachable?
10:24.30unasi7because one client (WIFI) will always detected as UNREACHABLE because of high ping (>300ms).
10:25.04Carlos_PHX> 300 is effectively offline for a phone
10:25.33nfi|ermestzafrir_laptop, my dial line is  Executing [s@macro-dialout-trunk:14] Dial("SIP/39-08b0b838", "Zap/2/0554250774|25|Ttr") in new stack
10:25.55tzafrir_laptopnfi|ermes, what version of Asterisk is it?
10:26.55tzafrir_laptopand do you see channel 2 on 'zap show channels' ?
10:29.58nfi|ermesAsterisk 1.4.22
10:30.20nfi|ermescentralino*CLI> zap show channels
10:30.20nfi|ermes<PROTECTED>
10:30.20nfi|ermes<PROTECTED>
10:30.20nfi|ermesThe 'zap show channels' command is deprecated and will be removed in a future release. Please use 'dahdi show channels' instead.
10:30.20nfi|ermescentralino*CLI> dahdi show channels
10:30.22nfi|ermes<PROTECTED>
10:30.24nfi|ermes<PROTECTED>
10:32.50tzafrir_laptopstill, you don't have Zap/2
10:33.04tzafrir_laptopDo you see it in the output of lszaptel ?
10:33.44tzafrir_laptop(or: cat /proc/zaptel/* # alternatively)
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10:37.16joobieguys anyone played around with IFS numbers before? or know much about them? i'm curious how an IFS number would be dialled from various countries.. my understanding is you need to put the international code in then '800' for the country code followed by the number.. not certain though....
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10:57.17whymarkwhhi there anyone active
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10:59.42datacompboywhymarkwh: yes
11:08.11IsUpanyone using Slackwere around here?
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11:14.24dokeHello people
11:16.11dokedoes anybody here use Cisco IP phones? either 7965 or 7975? I'm having some troulbes with the xml file.... I don't know what would be the correct value to enable the remote headset hookswitch control
11:18.26dokeor enble g722
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11:21.10whymarkwhysorry if this looks like a repeat i got disconnected:
11:21.15whymarkwhyi have telko linking to * with pri then in turn * linking to Legasy pbx via pri. I can make calls from the legasy through * to telko but can not connect to from telko to legasy
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11:25.05stix_Does anyone know how to connect to a mysql-server with a non-default port with the asterisk mysql cmd?
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11:40.58flushyo
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11:58.32Assidhrmm if i have a remote phone  which keeps getting unregistered from the system, even tho i changed the expires to 240 , and added qualify=2000 .. is there anything else i can do?
11:58.45Assidshould i reduce increase the expires?
11:59.15Carlos_PHXYou should fix the network problem that's causing the phone to become unreachable.
11:59.23IsUp~nat
11:59.23jbotsomebody said nat was Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
11:59.33IsUpwhat was it
11:59.39IsUp:D
12:01.11Assiddunno what to address
12:01.38Assidthe server itself is on a static ip..terminating on it
12:02.27Carlos_PHXHow often does the phone get unregistered?
12:03.39Assidnot too sure.. im actually looking at it as we speak.. 1 thing tho.. 1 of the phones show up as port 5060 instead of 55000 or some arbitrary number.. which the other phones ont he same network show up as
12:04.12Carlos_PHXDid you do some port mapping on the phone network?
12:04.16Carlos_PHXWhat kind of router?
12:05.09Assidit registers.. and i think in under a minute or so .. it unqulifiesw
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12:05.30Assidnetgear - finding the details as we speak.. its a remote office
12:05.50Carlos_PHXIt does sound like a NAT issue.
12:06.01Assidyeah thought so as well
12:10.37Assidbut just wanted to re-confirm.. before theycome after me for giving wrong info
12:10.51dokepls does anybody use asterisk with Cisco ip phones?
12:11.04dokethe 7975g or 7971 - 7965 / 7961
12:12.19Carlos_PHXNot any more, got rid of them all.  Is your question on the Asterisk side or phone?
12:12.59boolean12Do you have a question about using the 79XX series with asterisk?
12:15.42boolean12Curious, has anyone had any problems paging large groups using the page application?
12:15.54boolean12I'm currently 1.4.22
12:21.08AssidWGR-614v6
12:21.19AssidCarlos_PHX: thats the router
12:27.57Assidwhats the general "recommended" registration expire  time?
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12:46.07whymarkwhif you want to use the telkos timing on pri should you set the timing in zaptel.conf to 1 or 0?
12:46.56Carlos_PHX1
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12:49.46gambler1Hello, is it possible to disconnect sip user?
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13:03.44whymarkwhwhat should the timing for span 1 be if i connect elastix via bri dual port to telko  : 1 or 0
13:04.02whymarkwhsorry that is pri not bri
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13:09.00ziram19hello
13:09.10ziram19can someone help me
13:09.14ziram19<PROTECTED>
13:09.16ziram19<PROTECTED>
13:09.18ziram19<PROTECTED>
13:09.19ziram19<PROTECTED>
13:09.21ziram19<PROTECTED>
13:09.22ziram19<PROTECTED>
13:09.58ziram19it happens with thomson st2030
13:10.34ziram19they call but not receive calls
13:11.08mort_gib~pastebin
13:11.09jbot[~pastebin] 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
13:12.32mort_gibIs the DND button on??
13:13.02mort_gibThat is what my Snom handsets say when DND is on (480 from xxx.xxx.xxx.xxx)
13:13.32ziram19>i have 5 sip (1 aastra51i, x-lite, 3 thomson st2030)
13:13.43ziram19the pb is only with thomson
13:13.58ziram19any ideas?
13:16.11datacompboyziram19: SIP/100 is written as peer, or friend ?
13:16.21ziram19a friend
13:18.08ziram19a am just try to do sip call on my LAN
13:18.29ziram19the thing that must works onfew seconds
13:18.50mort_gibThe one you pasted in here -sic- indicates a DND button on the handset....
13:19.07mort_gibDo you have SIP/100 - * - SIP/200
13:19.07mort_gib??
13:19.26ziram19100 is Aastra51
13:19.49ziram19and 200 is my new thomson 2030 that i plug yesterday
13:20.04ziram19and until now i am not sleep
13:20.18ziram19i become crazy
13:20.39*** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163)
13:22.22ziram19mort_gib i don't have so much
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13:27.08ziram19mort_gib
13:27.23ziram19you think of what
13:32.01mort_gibHow to get one of my clients logged into their business online bank...
13:32.15mort_gibSo you have two SIP phones, plugged into what??
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13:32.55*** mode/#asterisk [+o lmadsen] by ChanServ
13:33.56mort_gibziram19: You there??
13:34.06ziram19hes
13:34.42ziram19i mean simply added on network
13:35.01ziram19and on sip.conf
13:35.20ziram19sip show peers show them
13:36.12ziram19when 200 (thomson) calls (aastra51i) is works fine
13:36.55ziram19but aastra to thomson failed
13:37.08ziram19thosmson to thomson failed also
13:38.09ziram19i really need help
13:40.46ziram19i set a debug on one ip of thomson st2030
13:40.46ziram19<PROTECTED>
13:40.47ziram19tunas4*CLI>
13:40.48ziram19<--- SIP read from 10.63.1.183:5060 --->
13:40.49ziram19SIP/2.0 480 Temporarily Unavailable
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13:40.52ziram19Via: SIP/2.0/UDP 10.63.1.4:5060;branch=z9hG4bK18a2bc6f;rport
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13:40.53ziram19From: "100"<sip:200@10.63.1.4>;tag=as60554de5
13:40.53ziram19To: <sip:200@10.63.1.183:5060;user=phone>;tag=c0a80101-19ce2ca
13:40.53ziram19Call-ID: 27edbb3349e905741d5433a837da2e3b@10.63.1.4
13:40.53ziram19CSeq: 102 INVITE
13:40.53ziram19Content-Length: 0
13:40.53ziram19<------------->
13:40.53ziram19--- (7 headers 0 lines) ---
13:40.53ziram19<PROTECTED>
13:40.53ziram19Transmitting (no NAT) to 10.63.1.183:5060:
13:40.53*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
13:40.53ziram19ACK sip:200@10.63.1.183:5060;user=phone SIP/2.0
13:40.53ziram19Via: SIP/2.0/UDP 10.63.1.4:5060;branch=z9hG4bK18a2bc6f;rport
13:40.53ziram19From: "100" <sip:200@10.63.1.4>;tag=as60554de5
13:40.53ziram19To: <sip:200@10.63.1.183:5060;user=phone>;tag=c0a80101-19ce2ca
13:40.53ziram19Contact: <sip:200@10.63.1.4>
13:40.53ziram19Call-ID: 27edbb3349e905741d5433a837da2e3b@10.63.1.4
13:40.53ziram19CSeq: 102 ACK
13:40.53*** kick/#asterisk [ziram19!n=joe@216.191.106.163] by [TK]D-Fender ([TK]D-Fender)
13:40.53datacompboyziram19 USE pastebin.ca !
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13:49.00[TK]D-Fenderziram19: Do not spam in here again, us a pastebin <-
13:49.02[TK]D-Fender~pb
13:49.02jbot[~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
13:49.03[TK]D-Fender^^^^^
13:49.08[TK]D-Fenderuse*
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13:49.30ziram19ok
13:50.19*** join/#asterisk flush (n=SYN_ACK@ip216-239-67-37.vif.net)
13:50.52[TK]D-Fenderziram19: And a SIP 480 message usually means you have "do not distubr" enabled on the phone.  Go look at them.
13:51.17mort_gibTK: which I have already told him twice....
13:51.34mort_gibBy now I know THINGS about DND :-)
13:51.49*** join/#asterisk Pazzo (n=ugelt@reserved-225136.rol.raiffeisen.net)
13:52.41ziram19can you explain me more "do not distubr"?
13:52.55[TK]D-FenderJust goes to prove that you can lead a horse to water, but PETA won't let you hold its head under...
13:53.08mort_gibLOL
13:53.15[TK]D-Fenderziram19: "Do not disturb".  As in the phone is telling you to "go away"
13:53.56mort_gibWhich incidently  is much like being kicked of the channel for spamming
13:54.40ziram19and what can i do?
13:55.00flushwhats the matter, asteriskNOW doesnt support wifi? doesnt even have the "iwconfig" command
13:55.18[TK]D-Fenderziram19: Go look on the phone
13:55.31[TK]D-Fenderflush: Quite possibly not.
13:55.40[TK]D-Fenderflush: However this isn't the place to complain about it
13:55.46HeMananyone using SNOM 190 and 320s? We have a hard time to get the dhcp-server give both of them the right options to read configuration file
13:56.05HeManwe get either of them to work, not both
13:56.12flush[TK]D-Fender no i know, what should i do if i want to be able to access my asterisk box wirelessly
13:56.19flushinstall ubuntu, then asterisk?
13:56.27flushjust format asteriskNOW distro you think
13:56.31[TK]D-Fenderflush: Any distro you can manage
13:56.42mort_gibHeMan: like you can't get them to read their "identity" file?
13:57.03[TK]D-Fenderflush: If you can't handle rPath (assuming you have the OLD *NOW) and Conary, then go pick something else
13:57.59HeManmort_gib: the 190 need a option 67 with filename but the 320 don't like the same option
13:58.14mort_gibHeMan: Firmware??
13:58.39HeManmort_gib: 7.3.4 in the 320
13:58.50mort_gib7.3.7 is latest...
13:58.54HeManbooting a 190 to see what it has
13:58.59mort_gibTHey should be on same FW
13:59.14mort_gibAt least I would keep them on the same...
14:00.10HeManthe 190 is on 3.60
14:00.20mort_gibHmm, that is ancient
14:00.29mort_gibOr is that the SIP FW??
14:01.20HeManmort_gib: "Version-Code:snom190-SIP 3.60x"
14:02.03mort_gibHeMan: -And that is the latest version for that phone
14:02.21HeManmort_gib: yes, just found it on the snom pages
14:02.59*** join/#asterisk Ariel_Calzada (n=usuario@200.71.48.212)
14:03.14HeManwell, I think I can "hardcode" options to the 190's mac-adresses in our dhcp-server
14:03.22Kattymorning.
14:03.32HeManluckily we don't have that many of them, I think we have 3 or 4
14:03.46mort_gibIt's not a very recent phone is it??
14:04.50Kattyit's going to be a beautiful day!
14:04.51[TK]D-FenderKatty: morning
14:04.54Kattyand a happy day!
14:04.56Kattyhugs [TK]D-Fender
14:05.05Kattywe don't have rioting in the streets!
14:05.09[TK]D-FenderKatty: Oh, and slap that bitch for me, will ya? ;)
14:05.15HeManmort_gib: no it was one of our first phones
14:05.15mort_gibHeMan: http://wiki.snom.com/Features/Mass_Deployment/Setting_Files/Text#Specific_Setting_File
14:05.22Katty[TK]D-Fender: simmer down, dear.
14:05.36Katty[TK]D-Fender: where would we be without the 'stupid' population
14:05.44Katty[TK]D-Fender: we'd have to do jobs we really didn't want to.
14:05.45[TK]D-FenderKatty: I have... I jsut figured You'd refuse to kill her so I kicked it down a notch ;)
14:05.50anonymouz666hello Katty!
14:05.55*** part/#asterisk boch (n=fran@customer191-9.iplannetworks.net)
14:05.57Kattywell good morning mister anonymouz666!
14:06.00Kattyhugs anonymouz666
14:06.17HeManmort_gib: yes we use that but the 190 and the 320 needs different options from the dhcp-server
14:06.49[TK]D-FenderKatty>[TK]D-Fender: where would we be without the 'stupid' population <-- the same place that promised me my robot maid, flying cars, meal-in-a-pill, and the healthy demise of fossil fuels
14:07.01Kattygrins
14:07.57[TK]D-Fendergoes back...TO THE FUTURE!
14:07.58Kattythat reminds me of a newspaper i saw.
14:07.58KattyBarack to the Future
14:07.58Kattyoh the punnery.
14:07.58[TK]D-Fender*groan*
14:08.35*** join/#asterisk ast-thecode (n=asterisk@scalise.csata.tno.it)
14:08.54mort_gibHeMan: Yes, I'm sorry but I don't have an answer to why they do...
14:09.13jameswfI think now that Obama doesnt have to pander anymore he should have jeremiah wright do the opening prayer at his inaguration as a FU guys
14:10.06HeManmort_gib: np, just hoped that anyone had a clever Vendor-Class-ID-dhcp-hack or such
14:10.16mort_gibHeMan: Is this because one phone uses option 66 and the other option 67 ??
14:10.23[TK]D-FenderHeMan: And adding both simultaneously for all doesn't work?
14:10.24HeManmort_gib: yes
14:10.30HeMan[TK]D-Fender: no
14:10.38[TK]D-FenderHeMan: what happens?
14:11.18mort_gibHeMan: But, eh, why tftp??
14:11.39*** join/#asterisk rnovotny22 (n=rnovotny@71-220-121-209.mpls.qwest.net)
14:11.49HeMan[TK]D-Fender: if we have the 67 option ther the 320 won't find it's config and not having it causes the 190 to not find it's config
14:11.59HeManmort_gib: no, it's still http
14:12.37HeManmort_gib: but it uses those options for letting dhcp handing out the location of the file
14:13.31mort_gibHeMan: if you only have 2-3 phones then use MAC to give specific boot options for those handsets
14:14.04HeManmort_gib: yes thats the way it going to be
14:14.15HeManmort_gib: but I thought we had more 190's
14:14.48*** join/#asterisk bmg505 (n=leon@196-209-8-172-ndn-esr-2.dynamic.isadsl.co.za)
14:14.50mort_gibhttp://www.pastebin.ca/1245887
14:15.51*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
14:15.56HeManthanks but we run dnsmasq
14:16.40*** join/#asterisk cesar_CR (n=cesar@200.91.75.66)
14:16.44[TK]D-Fenderhow...quaint
14:16.48IsUpany ideas about provisioning Grandstream phones? i have 30-35 phones about in my office
14:17.03[TK]D-FenderIsUp: Have you tried fire?  Fire is good..
14:17.25*** join/#asterisk TOrrIeri (n=torrieri@nelug/crew/torrieri)
14:17.58IsUp?
14:18.13KattyIsUp: that generally means you're going to have a hell of a time.
14:18.20KattyIsUp: aka, DOOM
14:18.29Kattyjbot: doom?
14:18.30jbotACTION stalks katty, chanting doom on you, doom on you!
14:18.35IsUpoh
14:18.38Kattyso cliche
14:19.13Kattywonders what sort of puppy Obama's children will get in the white house.
14:19.28Kattyman would i love to have 20 minutes with THAT dog trainer.
14:19.32*** join/#asterisk allsop (n=allsop@64.91.214.126)
14:19.37IsUpmaybe i can try to write a small php script, they all have same passwords.
14:19.54IsUplogin -> set tftp server -> reboot
14:20.20Kattynot sure about grandstreams, but if they're like polycoms at all, the polycoms have cfg files and you can just symlink some stuff.
14:22.22*** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee)
14:23.18Kattyhugs jaytee
14:23.34*** join/#asterisk nikko (n=nikko@69.57.49.100)
14:23.46*** join/#asterisk PTorres (n=PTorres@200.68.87.146)
14:23.57anonymouz666Katty: some puppy similar to nasser
14:24.04PTorreshi everyone
14:24.24Kattyanonymouz666: mew?
14:26.39PTorreshi, I am having a different issue with isdn now... :(
14:26.39PTorres2 trunks randomly unlink : pri debug span 2 = http://pastebin.com/m75939ff4
14:28.18dokedoes anybody here use Cisco IP phones? either 7965 or 7975? I'm having some troulbes with the xml file.... I don't know what would be the correct value to enable the remote headset hookswitch control
14:28.41dokeor the value to enable G722 wideband
14:29.19*** join/#asterisk jer (n=jer@unaffiliated/jer)
14:29.20PTorresall zaptel.conf and zapata.conf already checked and re-checked , they are correct , any ideas ?
14:30.10jayteehugs Katty
14:30.33[TK]D-FenderPTorres: What ver of *, Zap/DAHDI, card, pastebin "cat /proc/interrupts" , zaptel.conf
14:31.01[TK]D-FenderPTorres: What OS....
14:31.44[TK]D-FenderPTorres: And what does your telco have to say?
14:32.06PTorres* 1.4.18  ,  zap 1.4.10 , centos 4.5 , digium 4 e1
14:32.51PTorresthe telco are going with another pbx to try , but if we hook the two e1's back to back they dont lose sync
14:33.30PTorrescat /proc/interrupts = http://pastebin.com/me0c639e
14:33.55[TK]D-FenderPTorres: 169:     553657   80818672   51871673   20550089   IO-APIC-level  uhci_hcd, wct4xxp <- GT IT ON ITS OWN.
14:34.05[TK]D-FenderPTorres: aND WHAT MODEL OF CARD?
14:34.25PTorres05:00.0 Communication controller: Digium, Inc. Wildcard TE405P Quad-Span togglable E1/T1/J1 card 5.0v (rev 02)
14:34.41[TK]D-FenderEww... ancient
14:34.59PTorresit was working fine for over 20 days... it just started to mess with us last monday
14:35.35[TK]D-FenderPTorres: Do make sure to set "priresetinterval=np" or whatnot...
14:35.53*** part/#asterisk allsop (n=allsop@64.91.214.126)
14:36.10PTorresnot familiar with that 'priresetinterval=np' where should I place it
14:36.12PTorres?
14:36.19PTorreszapata.conf right ?
14:36.38[TK]D-FenderPTorres: Yes
14:37.01pifon the subject of cisco phones, one of them boots with POE and two others (same 7.4 firmware) don't, any idea? (they work without POE though)
14:37.08pif7960's
14:37.58PTorresthanks, I´ll update when they let me restart *  :D
14:39.20HeManMy dnsmasq-conf to give the snom 190 it's own option, http://asterisk.pastebin.com/d1f985b83
14:40.29HeManI just assumed that the old ones has 00:04:13:22:* and the new ones has 00:04:13:24:*
14:44.38*** join/#asterisk xuser (n=JJ@unaffiliated/xuser)
14:45.07*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
14:50.27phpboyEish, I'm in serious trouble
14:50.41tzangerany recommendations for a 4-port FXO ATA (small business, connecting to *on-lan* asterisk install
14:50.54phpboySome calls are droping over E1, problem is... no patern, not logs (from what I can see)
14:51.03[TK]D-Fendertzanger: 4, no, 8 = SPA-8000
14:51.08*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
14:51.50*** part/#asterisk smurf (n=smurf@debian/developer/smurf)
14:51.50tzanger[TK]D-Fender: ok, will look into it
14:52.10*** join/#asterisk Tebi (n=user@support.ccxtech.fi)
14:52.25*** join/#asterisk stimpie (n=stimpie@213-73-211-48.cable.quicknet.nl)
14:52.29*** join/#asterisk flush (n=SYN_SENT@ip216-239-81-70.vif.net)
14:52.58phpboyWhere do you think I should start looking, considering I've checked /var/log/messages and /var/log/asterisk/messages etc
14:53.20Kattyhugs tzanger
14:53.29*** join/#asterisk sdaniels (n=chatzill@216.65.195.52)
14:54.32IsUpphpboy: whats your card?
14:56.32KattyQueen of Hearts.
14:56.55[TK]D-Fenderlooks for his Motorhead CD
14:57.04*** join/#asterisk smurfix (n=smurf@debian/developer/smurf)
14:57.16[TK]D-FenderTHE ACE OF SPADES! THE ACE OF SPADES! THE ACE OF SPADES!
14:57.20*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
14:57.24IsUpmoto psycho
14:57.26[TK]D-Fenderrocks out
14:57.34phpboyIsUp: Digium 1TE420F
14:57.40phpboyE1 enabled
14:58.56IsUpany output on console? and can you reproduce this problem?
14:59.28*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
14:59.46phpboyI cannot reproduce it and my console is going insane... so can't really tell there, 26 active channels at the mo and it's end of business day :(
15:00.02smurfHI, I want to link two machines via E1 (for testing). I think the crossover cable is correct (1+2 <=> 4+5), but both systems still show RED. Does anybody have a sample configuration for that?
15:00.49*** join/#asterisk xuser (n=JJ@unaffiliated/xuser)
15:00.52IsUp1 <-> 4, 2 <-> 5, 4 <-> 1, 5 <-> 2 this is the T1/E1 cross-over cable
15:00.55*** join/#asterisk wiscados (n=mint@81.25.184.155)
15:01.06smurf... or some idea how to debug that further?
15:01.23*** join/#asterisk wiscados (n=mint@81.25.184.155)
15:01.41IsUpfirst, change your card 1 clocking to MASTER, and change card 2 clocking to NORMAL
15:01.47IsUpand whats your card?
15:01.55*** join/#asterisk wiscados (n=mint@81.25.184.155)
15:02.12phpboyIsUp: do you think my problem could be as a result of a timeing issue?
15:02.40IsUpphpboy, try to check logs. and try to reproduce this problem. i dont think its about with timing.
15:02.56smurfIsUp: Wildcard TE110P T1/E1
15:02.57IsUpdid you ask your Telco?
15:03.11smurfIsUp: master vs. normal is already set
15:03.12phpboyIsUp: I can see absolutely nothing strange in the logs, my telco says everything is fine :(
15:03.56IsUpsmurf, edit your /etc/dahdi/system.conf (default path). and then restart dahdi and run dahdi_cfg -vvv
15:04.34IsUpsmurf, did you replace wiring?
15:05.15PTorres[TK]D-Fender: I cant find anything about priresetinterval=np , the closest is resetinterval=never but I guess that would give me problems every 1 hour ???
15:05.37*** join/#asterisk putnopvut (n=putnopvu@nat/digium/x-2ffa7b3e8f8fcffc)
15:05.37*** mode/#asterisk [+o putnopvut] by ChanServ
15:05.49[TK]D-FenderPTorres: Yeah, thats the option. Do it
15:06.37PTorresI will, they just want to wait until the telco people is there to start trying stuff :s
15:06.56*** join/#asterisk fskrotzki (n=fskrotzk@host198.textwise.com)
15:07.57*** join/#asterisk superpop02 (n=mozveren@se167-1-82-242-148-65.fbx.proxad.net)
15:08.00superpop02hello all
15:08.11*** join/#asterisk Toerkeium (n=Miranda@201.216.206.221)
15:08.15smurfIsUp: both sides say "SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) " -- system.conf: span=1,0,0,ccs,hdb3,crc4 (span=1,1,... on the other side)
15:08.20superpop02I plan to write a uacsta interface for asterisk
15:08.41*** join/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
15:08.41*** mode/#asterisk [+o russellb] by ChanServ
15:08.42superpop02do you know if anyone has started a project like it ?
15:08.49smurfI've built (and tested :-P ) the xover cable myself
15:08.51Toerkeiumguys, my ITSP uses g729, at the agent point (the telephone) support G729. I wonder if I will need licences for MoH?
15:08.57IsUp:D
15:09.07*** part/#asterisk russellb (n=russellb@asterisk/digium-open-source-team-lead/russellb)
15:09.10superpop02Toerkeium, no if you dont transcode
15:09.33[TK]D-Fendersuperpop02: I'll bet Google would...
15:09.42Toerkeiumsuperpop02: the music for MoH must be g729 encoded?
15:09.51[TK]D-FenderToerkeium: Clrearly YES
15:10.01[TK]D-FenderToerkeium: Otherwise that would imply transcoding
15:10.13Toerkeiumgreat, thank you
15:10.39superpop02Toerkeium, for me yes
15:11.12superpop02[TK]D-Fender, google search failed :)
15:11.27superpop02[TK]D-Fender, do you have any information about uacsta for asterisk ?
15:11.37[TK]D-Fendersuperpop02: then perhaps this is another case of "look for somthing that doesn't exist"
15:12.02superpop02But do you think it can be interesting for the project ?
15:12.03superpop02or not ?
15:12.39ToerkeiumI'm in the process of uninstall a FreePBX system and builing an asterisk from scratch, I guess I will have to call digium to give me again new licences (I own 10 right now)
15:12.44[TK]D-Fendersuperpop02: To you apparently... how many others?  Its the first I've heard of it, noone has tried before, so I think that give you an idea of the level of interest
15:12.56[TK]D-FenderToerkeium: No.
15:13.11[TK]D-FenderToerkeium: Your machine is licensed, not your OS.
15:13.25superpop02probably no interest
15:13.58Toerkeiumah.. great then, long time not dealing with asterisk
15:13.59[TK]D-Fendersuperpop02: Some questions just seem to answer themselves...
15:16.10*** join/#asterisk Knightfal (n=Knightfa@75.142.144.45)
15:16.28KnightfalHi Guys I am running asterisk 1.4.22 and have two te410 digium cards providing my * calls I also have 15 call queues and ~ 35 agents using the switch. Everything has seemed to be going good except now two time today there were calls in queue and they would not goto agents. I tried logging them on and off but to no avail.  I restarted asterisk with the init script and everything worked like normal after that but I dropped all my calls
15:16.29Toerkeiumdo you know the AtCom sip phones?
15:16.35Toerkeiumare they "good"?
15:17.12superpop02Knightfal, so what ? :) did you capture some traces on console ?
15:17.17tzangeratcomm... isn't that the guy who was building the quadspan with hardware g729?
15:17.29Toerkeiumno idea really
15:17.37ziram19http://pastebin.com/m2cf4fc6c
15:18.14[TK]D-FenderToerkeium: Shit-On-A-Stick
15:18.34Toerkeiumis that good or bad ? :P
15:19.00ziram19i paste a debug on my thomson that don't receive calls
15:19.20jayteeoh, dear lord! I just got an email for an Open Source PBX survey from Digium and the entire border of the email is in a Trixbox green color. Could Digium and Fonality be having some merger talks behind our backs?
15:19.28[TK]D-FenderToerkeium: BAD.  Go with Linksys or Polycom.  Choice depends on what your import cost looks like
15:19.29ToerkeiumI was looking for a phone if abour $100, I just found that ones
15:19.36Knightfalsuperpop02, it doesnt show anything on the cli
15:19.47superpop02dead lock ?
15:19.52Knightfalsuperpop02, Unless a call comes in
15:20.21Knightfalsuperpop02, Then it send it to the queue and it sits there
15:20.26[TK]D-FenderKnightfal: You know queue dumps while in this state might have been considered useful...
15:20.47jayteeI think I remember seeing Shit-On-A-Stick twelve packs on sale somewhere last week but I can't remember.
15:21.04*** join/#asterisk Deeewayne (n=Deeewayn@nat/digium/x-fa972e9e7e3881d2)
15:21.04*** mode/#asterisk [+o Deeewayne] by ChanServ
15:21.18Knightfal[TK]D-Fender Logs?
15:22.58KnightfalGuys please let me know what information I can provide that will help
15:23.32*** join/#asterisk joerg (n=joerg@p5488B4D2.dip0.t-ipconnect.de)
15:23.35joerghey
15:23.56joergI actually have got a question that has nothing to do with asterisk ;)
15:24.09joergis there a channel about voip/sip in general somewhere?
15:26.05*** part/#asterisk rnovotny22 (n=rnovotny@71-220-121-209.mpls.qwest.net)
15:26.19joergthe problem sound easy....but I already tried everything to solve it without success....
15:26.33joergsomehow sip packets don't really get through my nat router.
15:26.41joergnot even with a stun server.
15:27.40cesar_CRguys how can I play the periodic announce every 15 secs when all the agent are not busy and my timeout is 45 ?
15:29.02*** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net)
15:29.26[TK]D-FenderKnightfal: no.
15:29.31Knightfal[TK]D-Fender, Im not sure what you mean by queue dumps
15:29.34[TK]D-FenderKnightfal: live prrof as it happens
15:29.44[TK]D-FenderKnightfal: look at the QUEUE and its agents live.
15:30.36Knightfal[TK]D-Fender, I can see the agents logged into the queue but no one gets them
15:30.53[TK]D-FenderKnightfal: Until you show us we can't tell you anything.
15:31.16*** join/#asterisk KingOfDos (n=irssi@ht1-ix.kingofdos.net)
15:31.30Knightfal[TK]D-Fender, so you would like to see the CLI output when the issue happens again?
15:31.53[TK]D-FenderKnightfal: If you want any kind of help you need to show something useful...
15:32.16magronezis away: almoco
15:32.26Knightfal[TK]D-Fender, what would be considered useful to you
15:32.42[TK]D-FenderKnightfal: I just told you.  Don't make this a circular argument.
15:33.41Knightfal[TK]D-Fender, Im not arguing, Im jusy trying to figure out exactly what I need to do to get help.
15:34.05[TK]D-FenderKnightfal: I jsut told you to do a queue & agent dump as it happens live.
15:34.15[TK]D-FenderKnightfal: And then you ask me if thats what I want.
15:34.52Knightfal[TK]D-Fender, Is there info somewhere other thatn in the CLI that will help troubleshoot this issue
15:35.17*** part/#asterisk lmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
15:35.21[TK]D-FenderKnightfal: no, CLI is what is happening at that moment.
15:35.26Knightfal[TK]D-Fender, thats where im lost what exactly does that mean
15:35.58Knightfal<PROTECTED>
15:37.08[TK]D-FenderKnightfal: "show queues" "show queue [name]" "show agents"
15:37.22Knightfalok
15:37.32[TK]D-FenderKnightfal: things you should long since have known and used
15:37.38*** join/#asterisk jameswf (n=james@dsl093-157-131.phx1.dsl.speakeasy.net)
15:37.42[TK]D-FenderKnightfal: go read the full list of CLI commands
15:37.59*** join/#asterisk rcahilig (n=sysad@202.78.75.246)
15:38.02*** join/#asterisk ajohnson (n=ajohnson@63.147.46.186)
15:38.05KnightfalI mentioned that is saw the agents there earlierearlier
15:38.19Knightfalbut I will get the data as the issue happens
15:39.08*** join/#asterisk mtryfoss (n=mtryfoss@81.166.192.6)
15:39.55mtryfossdo asterisk support sending generic number (isdn), also known as AdditionalCallingPartyNumber ?
15:42.07jameswfObama is like Open source? http://www.youtube.com/watch?v=q-4afdMalVA
15:42.07*** join/#asterisk Ariel_Calzada (n=aricalso@190.146.166.254)
15:43.25jayteeI typed "show me the money" at the CLI and it returned: "there is no money and it is unlikely you will make any significant amount using this"
15:45.16[TK]D-Fenderjameswf: Nifty thought, but there is a chunk of his IT policy I don't like...
15:45.50[TK]D-Fendermtryfoss: "core show function CALLERID" <- thats what you've got.
15:46.04jameswf[TK]D-Fender: I was looking for the bug site but obama is not on SF
15:46.28*** join/#asterisk IsUp (n=nocturne@unaffiliated/isup)
15:46.32IsUp~gsmbug
15:46.32jbot[~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read :  http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39
15:46.39musse-Any one using asterisk-java that can explain if originateToExtentionAsync() launches one thread per callbakc
15:46.46jayteejameswf, I disagree for the following reasons. Obama is well organized and his campaign has kept meticulous records of donations and contributors and methods for fundraising (documentation). Obama heads in one direction at any given moment instead of several directions at once.
15:46.47musse-*callback
15:46.55cesar_CRhello guys in queues.conf how can I set the periodic announce to play every 15 secs when my timout is 45 or more ?
15:47.28jameswfjaytee: so he is sun microsystems or redhat?
15:47.59jayteejameswf, neither of those fit either
15:48.51[TK]D-Fendermusse-: There are probably about a half dozen users of it that have ever made it in here... all only having probelms, and noone having answers for them.
15:49.06[TK]D-Fendermusse-: Java is proving a relatively unpopular tool for this
15:49.34jameswfFCC Green-Lights White Space Access http://www.crn.com/networking/212000630 << this should be exciting
15:49.37jayteejameswf, I'm just pointing out that to compare him to open source is unfair. He's not disorganized. His documentation isn't messy, incomplete and scattered all over hell and creation and he isn't one of 27 people working on 3 different forks of the same damn thing.
15:49.58jameswfor is he
15:52.38jameswfgoes in to Jedi debate mode
15:53.38*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)
15:54.28musse-[TK]D-Fender: using asterisk-java you mean ? have used it since 0.2 and it has worked fine.. thought that i would rewrite it now though to use the new Live API..
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15:55.15*** part/#asterisk fred-tmft (n=fred-tea@c-98-243-174-131.hsd1.mi.comcast.net)
15:56.25aiksa[LV]~pb
15:56.25jbot[~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
15:57.39*** join/#asterisk theHub (n=theHub@69.177.93.21)
15:58.01aiksa[LV]anyone mind taking look why i have these "pri_fixup_principle: Call specified, but not found?" errors? http://www.pastebin.ca/1245949
15:58.11aiksa[LV]?
16:01.27*** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net)
16:02.03aiksa[LV]ordinary E1 link
16:02.17*** join/#asterisk shaw22dog (n=aaron@pacman.oaklandcorp.com)
16:02.22aiksa[LV]connected to * through redfone E1 bridge
16:04.04shaw22dogHello... Is there any opensource front ends for Asterisk, similiar to Switchvox and Druid. I'm looking for more of an enduser interface.
16:04.47shaw22dogA one-stop shop to have the operator panel, web voicemail, contacts, etc...
16:05.06Deeewayneaiksa[LV]: I haven't spent much time in that specific code, but it looks (to me) like the call was redirected to a busy destination, the call was torn down, and fixup failed because the call no longer existed
16:06.38*** join/#asterisk SirThomas (n=tomc@mail.kendeco.com)
16:07.33aiksa[LV]Deeewayne: ok, any workaround this?
16:07.57aiksa[LV]because - the strange thing is - as soon as i get this message - echo canceler would break down.
16:08.19aiksa[LV]and not resume the operations untill there is another Zap call
16:09.20aiksa[LV]I "watch" /proc/oslec/status and as soon as i would get this message it just falls back to "inactive mode"- waiting for a new call although there are other Zap calls present
16:09.49*** join/#asterisk shaw22dog (n=aaron@pacman.oaklandcorp.com)
16:09.54DeeewayneI'd have to reproduce the issue so I can learn exactly what is going on before I can provide you with a work around.  If I were you, I'd start looking at where the call is being redirected to and why the redirect destination is busy.  Is the location that the call is being redirected to under your control or is it somewhere out in the network ?
16:10.02ManxPoweraiksa[LV]: is there an OSLEC support mailing list or channel?
16:10.54[TK]D-FenderManxPower: https://lists.sourceforge.net/lists/listinfo/freetel-oslec
16:11.01[TK]D-Fender~oslec
16:11.01jbotmethinks oslec is Open Source Line Echo Canceller. See http://www.rowetel.com/ucasterisk/oslec.html .
16:11.17aiksa[LV]ManxPower: not sure about the channel
16:11.23aiksa[LV]but there is a mailing list
16:11.48aiksa[LV]and nevertheless I understand that "pri_fixup_principle: Call specified, but not found?" shouldnt happen
16:11.56ManxPoweraiksa[LV]: you might want to ask there.  I don't think many people here use it.
16:12.51ManxPoweraiksa[LV]: Unfortunately not that many people use OSLEC here, so I can't even tell you if that message is common or not with OSLEC.  I've never seen that message on an unpatched Asterisk.
16:13.10aiksa[LV]ManxPower - it happens even wo oslec patch
16:13.44aiksa[LV]it was just with the help of oslec that i discovered that it had negative side effects
16:14.28ManxPoweraiksa[LV]: I assume you have the latest version of Asterisk, libpri, and zaptel/dahdi for the major release you are using?
16:15.41*** join/#asterisk propellerhead (n=yogurt2u@host38.190-136-115.telecom.net.ar)
16:16.09aiksa[LV]Asterisk 1.4.22, zaptel-1.4.12.1 with bristuff patch applied + ztd_ethmf from redfone, libpri-1.4.3
16:16.21aiksa[LV]and oslec-0.2
16:17.28*** join/#asterisk tmiw_ (i=thinknot@cpe-66-75-245-86.san.res.rr.com)
16:17.31aiksa[LV]I have two suspections - 1. (most probable) I have done smth. wrong in zapata.conf, 2. - the remote party might send incorrect messages
16:18.28[TK]D-Fender~oslec
16:18.29jbot[~oslec] OSLEC is the Open Source Line Echo Canceler. It is an superior alternative to the native SWEC routines in Zaptel/DHADHI and debatably a small notch below that of Digium's HPEC and Sangoma's SoftEcho in effectiveness. Web site : http://www.rowetel.com/ucasterisk/oslec.html , Mailing list : https://lists.sourceforge.net/lists/listinfo/freetel-oslec
16:18.32[TK]D-FenderThere, much better
16:18.42aiksa[LV]:))
16:19.19ManxPoweraiksa[LV]: I strongly doubt it is a config issue.  Wrong switchtype is the only config option I can think of that could cause odd issues.
16:19.28*** join/#asterisk oej (n=olle@ns.webway.se)
16:19.44aiksa[LV]and the reason i asked it here - 1) it happened before applying oslec patch & 2) it looks  like it is not realted to oslec but to pri msg. handling
16:20.13ManxPowerbristuff massivly patches libpri
16:20.56ManxPoweras could ztd_ethmf might also patch libpri.
16:20.59aiksa[LV]I had suspicon of config issue because of: http://www.spinics.net/lists/asterisk/msg67621.html
16:21.03[TK]D-FenderLast I heard you couldn't sue BRIStuff & libpri together..
16:21.05[TK]D-Fenderuse*
16:21.08aiksa[LV]ztd_ethmf doesnt patch libpri
16:21.09ManxPoweryou're not getting HDLC errors are you?
16:21.20aiksa[LV]only makefile of zaptel
16:21.27aiksa[LV]adding additional module
16:21.34[TK]D-Fenderthen again, thats a while back
16:21.34aiksa[LV]ManxPower: no no HDLC errors
16:21.53aiksa[LV]i need bristuff for XorcomAstribank
16:22.02*** join/#asterisk luckyaba (n=lucky@ip70-177-7-204.sb.sd.cox.net)
16:22.25ManxPoweraiksa[LV]: I strongly doubt that.
16:22.40aiksa[LV]Astribank with BRI channels
16:22.56ManxPowerxorcom is supposed to include everything in it's drivers that are built into Asterisk
16:23.11aiksa[LV]bristuff ads xpd_bri module which is needed by xorcom astribank
16:23.25ManxPoweraiksa[LV]: Unfortunatly you have so many patches and changes to Asterisk, I doubt you'll ever find the problem.
16:23.31aiksa[LV]:PP
16:23.36aiksa[LV]fantastic
16:23.43ManxPowertzafrir_laptop: Are you around?
16:24.05aiksa[LV]tzafrir_laptop was the one giving advice on installign astribank
16:24.17aiksa[LV]i had some starnge issues with flashing of the firmware]
16:24.18tzafrir_laptopyup, following
16:24.25ManxPowertzafrir_laptop: He maintains the Xorcom drivers.
16:24.31aiksa[LV]sorry for typos.
16:24.36ManxPowertzafrir_laptop: does the Astribank need BRIstuff?
16:24.46tzafrir_laptopthe BRI one: yes
16:24.50*** join/#asterisk jpcansa (n=jpbenavi@201.201.66.155)
16:24.52ManxPowerAh, OK.
16:25.06ManxPowerWell, I'm out of ideas.  Best of luck aiksa[LV]
16:25.08jayteecan I redirect CLI output to a file?
16:25.17ManxPowerjaytee: happens by default.
16:25.22ManxPower/etc/asterisk/logger.conf
16:25.57jayteeManxPower, ever if I type a command like dialplan show?
16:26.30ManxPowerjaytee: You would have to try it an see.
16:26.45ManxPowerjaytee: it would take mere seconds to try it.
16:27.22aiksa[LV]tzafrir_laptop: does bristuff patch libpri?
16:27.29Deeewaynejaytee: asterisk -rx "dialplan show" > myfile
16:27.39*** join/#asterisk axisys (n=axisys@155.70.141.45)
16:28.09tzafrir_laptopjbot, solec is also OSLEC for DAHDI: http://docs.tzafrir.org.il/dahdi-linux/#_oslec
16:28.09jbotokay, tzafrir_laptop
16:28.15jayteeDeeewayne, thank you!
16:29.19jayteeManxPower, it doesn't when I just do the command and then check the logs with tail, just the normal console messages from * itself. Thanks for the suggestion though.
16:30.42ManxPowerjaytee: oh well.
16:30.45jpcansacan anyone recommend a software to monitor calls from all extensions in my asterisk?? like for billing?
16:31.14codefreeze-lapAnyone here ever use the Park() app successfully? I'm contemplating changing it...
16:31.41jayteeManxPower, I wasn't being very clear about what I was trying to do, my bad!
16:32.10[TK]D-Fendercodefreeze-lap: My guess would be just about everyone using Parking... which is a lot of people.
16:32.34[TK]D-Fenderjaytee: No, there isn't a clean way keep a logged scroll-back that I can think of.
16:32.55codefreeze-lap[TK]D-Fender: no, no... what I seek is someone who used the Park() app in their dialplan...
16:33.04ManxPower"screen" should be able to keep a scrollback log.
16:33.05tzafrir_laptopaiksa[LV], span 9 is an E1 link?
16:33.43[TK]D-Fenderjaytee: I suspect you could create a daemon that will initiate a "-r" to dumpt to file, and then a hook in logrotate to kill it, rotate, then resume, etc.
16:33.43ManxPowercodefreeze-lap: Asterisk parking uses the Park() app internally AFIK
16:33.44[TK]D-Fendercodefreeze-lap: thats what gets called when you call "700"....
16:33.54[TK]D-Fendercodefreeze-lap: Its just auto-generated.
16:33.54tzafrir_laptopaiksa[LV], yes, it does. And for once it gives it a nicer error printing function
16:34.06[TK]D-Fendercodefreeze-lap: Go look what happens when you use the parking context and try to park a call.
16:34.47[TK]D-Fendercodefreeze-lap: I blame this app for the devlopment of users.conf generating dialplan... horrid idea.
16:34.56jaytee[TK]D-Fender, not necessary. I didn't think of using the -rx flag with a redirect to file at the linux command line. Deeewayne pointed me in the right direction.
16:35.22ManxPowerchecks his garlic and wooden stake when he years "users.conf"
16:35.23[TK]D-Fenderjaytee: RX works for a single command... you made it sound like you wanted general output
16:35.24codefreeze-lapManxPower: [TK]D-Fender: Not exactly. There are several paths into the park functionality. Each is a little different. This one is different from the others, in that only one channel is involved...
16:35.29ManxPowerusers.conf can be blamed for many thinfs 8-)
16:36.04[TK]D-Fendercodefreeze-lap: Well the essence is that the call is handed off to that dialplan app, and rebridged when the pickup is requested.
16:36.19[TK]D-Fender~users.conf
16:36.19jbotusers.conf is a flaming pile of sh1t that takes the fine control of several perfectly usable * config files and reduces them to the lowest common denominator and makes your system behave like a "toaster grade" PBX system.
16:36.20[TK]D-Fender:D
16:36.47[TK]D-Fenderahh... an oldie, but a goodie!
16:36.49jaytee[TK]D-Fender, I admit, I was being a bit vague there. Mea Culpa. I worked till 11pm last nite. I'm not firing on all thrusters today.
16:37.05[TK]D-Fenderjaytee: Goose eject! Eject!!!
16:37.11[TK]D-Fendernoooooooooooooooooooooo
16:37.16[TK]D-Fender*b00m*
16:37.19jayteerofl
16:37.28*** join/#asterisk Firass-z0r (n=asadf@juicebox.vikcomm.wwu.edu)
16:37.50jayteeI liked Anthony better in "Gotcha". "Mon crayon es large"
16:38.20aiksa[LV]tzafrir_laptop: yes span 9 is e1 link
16:38.23aiksa[LV]ztd_ethmf
16:38.27[TK]D-Fenderjaytee: Double-negative connotation on that one :0
16:39.01aiksa[LV][TK]D-Fender: nice one!
16:39.09aiksa[LV]regarding users.conf
16:39.35[TK]D-Fenderaiksa[LV]: :D
16:39.53aiksa[LV]off for a smoke
16:40.03[TK]D-FenderSooo glad I turned off ChatZilla's formatting options (bold, italic, slimy decodes....)
16:40.25aiksa[LV]tzafrir_laptop: spans 1 to 8 are XorcomBri s
16:40.31ManxPowerThe first thing I do in chat clients is tell it to ignore formatting codes.
16:40.32[TK]D-FenderSmiley's I might turn back on, but it hurts poepl how "bullet-list" like A), etc
16:40.50aiksa[LV]visual smileys, yuck
16:41.03[TK]D-FenderManxPower: I leave MIRC colours on just because they aren't really stuff you can do by accident
16:41.08aiksa[LV]the worst thing to happen, since, well the worst thing
16:41.19[TK]D-Fenderaiksa[LV]: Go get some sliced bread...
16:43.50rcahiligHello, I have asterisk 1.4, I have a US DID number and I want to forward to my cellphone number, is it possible? Can't find any documentation in the internet.
16:44.24tzafrir_laptopaiksa[LV], at first glance this happens in the clean-up after the release
16:44.41tzafrir_laptopwhat actualy noticable issues do you see?
16:44.55[TK]D-Fenderrcahilig: This is your cellphone's job
16:45.17[TK]D-Fenderrcahilig: pretty much EVERY plan I've ever heard of includes this by deault
16:45.19[TK]D-Fenderdefault*
16:45.33[TK]D-Fenderrcahilig: Wait..
16:45.40[TK]D-Fenderrcahilig: How is your DID delivered?
16:46.02rcahiligdirect from my voip provider
16:46.13ManxPowerrcahilig: You are not describing forwarding, you are describing exactly what all Asterisk dialplans do.  i.e. accept an incoming call leg and generate an outgoing call leg.  that outgoing call leg can be a sip phone, an itsp, a cell phone or an alien mind control device.
16:47.30*** join/#asterisk pardove (n=chatzill@195.146.46.6)
16:48.40pardovemy pri channel says: SABME (set asynchronous balanced mode extended)
16:49.05ManxPowerpardove: that is not an error
16:49.22pardovebut my pri span can accept calls anymore
16:50.17pardovewhats the cause of this problem
16:50.20pardove?
16:50.55rcahilig[TK]D-Fender: a cellphone
16:50.57ManxPowerI don't know what the cause of the problem is.  I do know, however, that that message does not indicate and error.
16:51.35[TK]D-Fenderrcahilig: Then either you have * call out, or ask them to do it
16:51.53*** join/#asterisk l2trace99 (n=jr@75.112.133.235)
16:52.54ManxPowerexten => 5046665555,1,Dial(SIP/yourcellnumber@youritspsipconfentry)  where  5046665555 is your DID that you want to "forward"
16:53.09pardoveManxPower: when i run pri debug span 5 on cli i get lots of this message: Sending Set Asynchronous Balanced Mode Extended
16:53.13ManxPowerthis is not rocket science, this is basic Asterisk stuff.
16:53.16Kattyhttp://img147.imageshack.us/img147/5751/1225890455052st0.jpg )=
16:53.18*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
16:53.34ManxPowerpardove: You don't get it do you.  THAT IS NOT AN ERROR MESSAGE.  That is obviously a debug message.
16:53.47Kattythat's so sad.
16:53.50rcahilig[TK]D-Fender: thanks man
16:53.52*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:55.45pardoveManxPower: sorry! i got it. but my pri cant get calls and this is the only sign that i get
16:55.57PTorres[TK]D-Fender: setting the reset interval did not fix the problem... and the telco plugged their pbx and they didn´t had problems :(
16:56.21outtoluncdoesn't mind emails about events, but a survey (grr)
16:56.32[TK]D-FenderPTorres: You still have an IRQ issue to look at and other cards to test
16:56.45*** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net)
16:57.44mort_gibI have had a few incidents with Queues -Are there any "do never" around??
16:57.46PTorresyes, we have another card, I guess we can try that but I do not see IRQ misses in zttool
16:58.22ManxPowerpardove: Keep looking.
16:58.35pardoveManxPower: :(
16:58.56aiksa[LV]tzafrir_laptop: °it look like the echo cancelation for all of the other concurrent call stops
16:59.12[TK]D-Fendermort_gib: Sorry, could you be a little more vague please, I almost got that...
16:59.49mort_gibLOL, i could possibly be a bit more vague, if I could collect my thoughts
16:59.54aiksa[LV]tzafrir_laptop: i didnt joined these two issues before actually running "watch -n1" on /proc/oslec/status
17:00.03mort_gibNo, one of the phones just locked up a few times....
17:00.06*** join/#asterisk xacatecas (n=jkroon@dsl-240-156-31.telkomadsl.co.za)
17:00.38mort_gibOr the Queue itself had a problem, I didn't have to restart * though...
17:01.21aiksa[LV]the strange thing is - I have some 5 Zap chans up and active (used in conversation) then for one of the channels I get this "pri_fixup" error/warning and at the very same moment all of the info disapears from that status file
17:01.26aiksa[LV]and it states
17:01.33aiksa[LV]no echo canceller being monitored - make a new call
17:02.08aiksa[LV]then echo canceler kicks in with full force as soon as any new Zap channel is created
17:02.46shaw22dogDoes anyone have any suggestions for Web UI for Asterisk, I didn't know if there was something similiar to Druid or Switchbox. I don't want to use a precanned PBX, but I do like their one stop shop for Op Panel, Web VM, and contacts.
17:03.08aiksa[LV]so from my dumb observations I could assume, that somewhere in zaptel this error causes to drop ec on all chans
17:03.16shaw22dogNot looking for admin control, just basic user control.
17:03.25aiksa[LV]I might be off by a mile nevertheless
17:04.40ManxPowershaw22dog: you might have better luck asking on a channel where people actually use GUIs
17:04.49tzafrir_laptopaiksa[LV], so this pri_fixup is not at the end of a call?
17:05.03shaw22dogManxPower: Thanks.
17:05.32ManxPowershaw22dog: several of those channels are listed in the /topic.  Most clients display the /topic when you join a channe.
17:06.55aiksa[LV]tzafrir_laptop: it might be at the end of the call (it itself gives too little info. to understand which call it is related to, nevertheless).
17:07.09aiksa[LV]but the point is: imagine you had 5 calls through zap channel
17:07.11shaw22dogManxPower: Okay, I'll ask around. Thanks.
17:07.21aiksa[LV]5 different, not related to each other
17:07.24ManxPower~freepbx
17:07.25jbot[~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
17:07.26ManxPower~trixbox
17:07.27jbotsomebody said trixbox was a full linux distro that includes FreePBX and other 3rd party add-ons. It is all this extra stuff which makes trixbox VERY difficult to support, and is not supported in #asterisk. Try asking in #trixbox or on their forums & wiki at http://www.trixbox.org/.  We do not recommend using it.
17:07.58aiksa[LV]now - it takes only one pri_fixup message to disable EC on all other 4 channels
17:08.21ManxPowerhugs his external hardware EC
17:08.49tzafrir_laptopaiksa[LV], two things I suspect:
17:09.16aiksa[LV]mort_gib: - one of do nevers (i found this out the hard way) is that if your agents are on Local/ channels, then if you transfer the call and then later logout the agent the transfered call will be droped
17:09.19tzafrir_laptop1. bristuff (Asterisk) is a bit more aggressive in disabling the EC (if it suspects a data call)
17:09.36tzafrir_laptop2. some locking issue with oslec
17:09.46mort_gib<aiksa[LV]>Yes but I don't use agents
17:10.00aiksa[LV]well not agents by queue members
17:10.14aiksa[LV]sorry for wrong definition.
17:10.32mort_gibNo prob, -I use queue members so they never log off
17:10.42aiksa[LV]ManxPower: I also have HW ec on the redfone mudules, but in certain occasions i still have the echo
17:11.08aiksa[LV]oslec helped in those issues
17:11.12*** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290)
17:11.44aiksa[LV]tzafrir_laptop: so one of the possible solutions, could be to divide redfone and Astribank modules between two boxes
17:11.56*** join/#asterisk backblue (n=igor@82.102.1.42)
17:12.04*** join/#asterisk CunningPike (n=arodgers@vpn.dnv.org)
17:12.08aiksa[LV]and on the one wo Astribank doesnot use bristuff
17:12.14backbluehi, anyone knows anything about the "new uniq id" introduced in asterisk 1.6?
17:12.36tzafrir_laptopaiksa[LV], maybe
17:12.50aiksa[LV]tzafrir_laptop: any other way to try to get to the cause of this?
17:13.11ManxPowerbackblue: it was not mentioned in UPGRADE.txt or Changelog?
17:13.24tzafrir_laptopfirst-off, trace calls to disable the EC
17:13.35tzafrir_laptopstart at the asterisk level
17:13.58aiksa[LV]the usual: set verbose and debug
17:14.53aiksa[LV]and then I should try to force this
17:14.58tzafrir_laptopaiksa[LV], actually there are already debug prints there
17:15.06aiksa[LV]to find out how to replicate all of this
17:15.10Ritzeriskim trying to see if someone could help me make sense of why after 2 calls in use on the 3rd call i get a SS-noservice All my contexts are pointing to From-zaptel ive done a bunch of google research on this issue just cant seem to cap it though
17:15.50tzafrir_laptop(Enabled|disabled) echo cancellation on channel %d
17:15.57aiksa[LV]Ritzerisk: perhaps you have reached simultanious call limit?
17:16.11aiksa[LV]tzafrir_laptop: those messages doesnt show up
17:16.21aiksa[LV]in those cases
17:16.29aiksa[LV]i see them when transmitting faxes,
17:16.42tzafrir_laptopdo you have debug logging enabled? debug = 1 at least?
17:16.47aiksa[LV]not when EC disapears after pro_fixup
17:17.05ManxPoweraiksa[LV]: echo cancelers disable when they detect a fax tone.
17:17.12aiksa[LV]ManxPower: i know that
17:17.20aiksa[LV]and this is when i see that message in logs
17:17.31tzafrir_laptopThis is done in Zaptel and generates a kernel message
17:17.47aiksa[LV]however that message is not shown when EC disapears after pri_fixup
17:18.17tzafrir_laptopI also suspect locking issues with oslec related to the proc interface
17:18.30tzafrir_laptopCan you try avoiding any monitoring of the proc interface?
17:18.42aiksa[LV]tzafrir_laptop: just let me check if i had debug on when I took that pastebin log snapshot
17:19.07tzafrir_laptopcan you sense the presense of the EC from the quality of the call?
17:19.09aiksa[LV]tzafrir_laptop: when how I will be able to see if it is still active?
17:19.44aiksa[LV]tzafrir_laptop: i started to monitor it when users started to complain about echo
17:19.55aiksa[LV]so apperantly they did have the echo
17:20.14aiksa[LV]of course i cant rollback what was oslec status at the moment they told this.
17:20.22aiksa[LV]they normally report this after the call :P
17:20.32tzafrir_laptopno kernel messages from oslec?
17:20.40aiksa[LV]dmesg and syslog?
17:20.46aiksa[LV]just a sec.
17:20.48tzafrir_laptopit looks like the next place to add trace messages to
17:21.51*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
17:22.06aiksa[LV]no nothing in dmesg or syslog
17:22.14aiksa[LV]appart from syslog messages by zaptel
17:22.23aiksa[LV]which disables EC because of fax
17:24.05*** part/#asterisk shaw22dog (n=aaron@pacman.oaklandcorp.com)
17:25.22tzafrir_laptopat the same time? or earlier?
17:25.50aiksa[LV]at random times (when faxes are sent) and not the same times when oslec stoped
17:26.26aiksa[LV]at least smaller number of times than EC stopped.
17:26.50aiksa[LV]from what i gathered visually was that every time i had pri_fixup message, the ec stoped
17:26.50*** join/#asterisk freakazoid0223 (n=mattc@68.238.187.201)
17:26.51backblueManxPower: BRIDGEPVTCALLID
17:26.59backbluei think it's this...
17:27.53*** join/#asterisk xnixan (n=xnixan@unaffiliated/xnixan)
17:28.26ManxPowerbackblue: Huh?
17:28.56backblueManxPower: the thing is, i need to associate calls in CDR.
17:29.27backbluefor that, i need a uniq id, that it's really uniq! :p
17:29.48backblueas i read somewhere (i dont find now) 1.6 already as that.
17:30.17*** join/#asterisk waKKu (n=ugabuga@unaffiliated/wakku)
17:30.21waKKuafternoon folks ;)
17:30.33waKKudoes someone there using asterisk with ethernet channel bonding ? round-robin ?
17:30.42waKKuis it "usable" ?
17:30.49*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
17:30.55Corydon76-digwaKKu: yes, it's usable
17:30.57QwellwaKKu: That isn't an application level thing.
17:31.04waKKuQwell OT ?
17:31.05Qwellthe app doesn't know or care that you're doing that
17:31.24ManxPowerbackblue: all major changes are documented in UPGRADE.txt and all changes major and minor are documeted in Changelog
17:31.39magronezis back
17:31.51waKKuQwell well.. but, my doubts is over tcp packets ordering ... could it be a problem with voice packets ?
17:32.05ManxPowerbackblue: I don't know the answer to your question, but I know where you can fine the answer.
17:32.21ManxPowerwaKKu: um, asterisk does not use TCP for audio or SIP
17:32.22Ritzeriski dont know if its that aiksa .... only becuase i had existing a 4 fxo zaptel card and i replaced it with an 8 and i added more Iax2 exts but only 2 calls can go then i get the the number you dialed is not in service (ss-noservice)
17:32.39waKKuManxPower oh.. indeed ;D
17:32.43ManxPowerwaKKu: But as Qwell said, Asterisk knows nothing about that, it is an OS level thing not an application issue.
17:33.10ManxPowerRitzerisk: try putting the cli output of a failed call on pastebin.ca.
17:33.23ManxPowerRitzerisk: I hope you are not using a GUI.
17:33.54waKKuok.. but a round-robin bonding ethernet could increase throughput and decrease latency, could it  ?
17:34.12backblueManxPower: where?
17:34.34aiksa[LV]ok. off to home (away from console) for some hour or so.
17:34.46xacatecasis it possible to use multiple cdr modules in conjunction with one another?
17:34.57aiksa[LV]tzafrir_laptop: before I go - would it make sense to put this issue up on oslec mailing list?
17:35.06ManxPowerbackblue: IN THE ASTERISK SOURCE CODE.
17:35.14tzafrir_laptopaiksa[LV], I think so
17:35.26ManxPowerbackblue: That is where all the official documentation is located.
17:35.45tzafrir_laptopaiksa[LV], the first thing to do is to trace if the blame is kernel or userspace
17:35.57*** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender
17:36.01codefreeze-lapxacatecas: yes. You can use multiple cdr backends at once. Each CDR is posted to each backend.
17:36.52backblueManxPower: nice awnser.
17:36.52xacatecasok, i recall having attempt selecting which cdr backends to use, but couldn't quite get the hang of it, is it merely a matter of loading/unloading the ones you want/don't want?
17:36.58aiksa[LV]tzafrir_laptop: hmm if I have no meesages indicating that ec is disabled (only a suspection because of /proc/oslec) how could i do this?
17:37.17aiksa[LV]tzafrir_laptop: get dirty with oslec code and add some printouts?
17:37.43aiksa[LV]ManxPower: the old joke about moving from RTFM to RTFSC
17:37.53aiksa[LV]?
17:38.11ManxPowerI wish I knew why people don't even bother to look at the included docs.
17:38.53aiksa[LV]ManxPower: because it is the nature of the man to be lazy
17:39.05aiksa[LV]ok, thats it. leaving
17:39.50ManxPoweraiksa[LV]: But they don't even THINK about looking there.
17:40.04aiksa[LV]ManxPower: then its the lack of experince
17:40.35Ritzeriskkk had to remote in and laptop it haha .... heres the pastebin ....   http://www.pastebin.ca/1246016
17:40.37*** join/#asterisk chigital (n=chigital@tmo-100-46.customers.d1-online.com)
17:40.39tzafrir_laptopaiksa[LV], yes, a debug printk there could help
17:40.43ManxPoweraiksa[LV]: even experienced people don't even think to look in the Zaptel README to see what card goes with which driver,
17:40.43aiksa[LV]after nights of tracing the dependancies and various non compatible versions of libraries you get used to at least looking at INSTALL file
17:40.50tzafrir_laptopin the wrapper echo_can_free
17:41.16aiksa[LV]been a while since i did anything c
17:41.24aiksa[LV]but ok will try
17:41.45tzafrir_laptopManxPower, what makes you say that? cards that are included there do appear in the README
17:42.10tzafrir_laptophttp://docs.tzafrir.org.il/#_supported_hardware
17:43.08tzafrir_laptopAnd if there's anything to fix there, please let me know
17:43.08ManxPowertzafrir_laptop: Yes, the readme is where the official list of card/drivers are listed, but nobody reads is.
17:43.08aiksa[LV]tzafrir_laptop: btw - it looks like i had set debug at least to 1 by the moment i made that pastebin log
17:43.51tzafrir_laptopManxPower, I know nobody reads it initially. Nobody bothers reading the docs if things work
17:44.20tzafrir_laptopBut I made the table of contents work well so you can easily point someone to a specific section in the README
17:44.33tzafrir_laptopand then blame them with RTFR
17:44.57ManxPowerI've seen people spend DAYS on a problem that would have been solved if they read the Zaptel readme or the Asterisk UPGRADE.txt
17:44.59codefreeze-lapxacatecas: first, you have to make sure any libs you'll need are present; all the packages installed for the backends you want. Then you make menuselect; then you select your backends. Then you exit and save from the menuselect stuff; then you make, then you make install...
17:45.06tzafrir_laptopRTFM is useful when the M is a specific reference
17:45.13jayteedevelopers should start naming the files DONTREADME.txt or NOTHINGTOSEEHERE.txt. That'll almost guarantee people will read them.
17:45.22ManxPowerjaytee: excelent idea!
17:45.38tzafrir_laptopManxPower, what part of UPGRADE.txt applies to my system?
17:45.40codefreeze-lapxacatecas: Then you make sure all the config files for all the backends are present and properly set up.
17:45.50jayteeManxPower, just going on my gut instinct and tendency to rebel :-)
17:45.55ManxPowertzafrir_laptop: all of it.  It only lists the major changes.
17:46.02tzafrir_laptopcan you point me to a specific section in the UPGRADE document?
17:46.17ManxPowerI'm not suggesting people read the Changelog, which is what documents EVERY change.
17:46.33tzafrir_laptopNothing about queues is relevant to my system. Likewise stuff about mISDN and mgcp
17:47.38ManxPowertzafrir_laptop: Do you use Voicemail?
17:47.55ManxPoweror Dial?  or any of the other apps that UPGRADE.txt tells you about changes for?
17:47.56tzafrir_laptopyes
17:48.29unpaidbillaww yeah baby gimme that new ipod touch
17:48.37ManxPowertzafrir_laptop: Heck you should check UPGRADE.txt just to confirm that no major changes happened to mISDN or Queues
17:49.01xacatecascodefreeze-lap, sounds simple enough.  I've got a working one, but I want to trial a custom one, and I'd prefer running mine alongside the existing stuff for testing purposes.
17:49.14ManxPowerthere is a Queues section in the 1.4 UPGRADE.txt.
17:49.15tzafrir_laptopManxPower, you're telling me that people should. Aparantly people don't. But I aksed you a different question: how can you point me to a specific change from UPGRADES.txt ?
17:49.37codefreeze-lapxacatecas: quite reasonable and practical
17:49.37ManxPowertzafrir_laptop: I can read it for you and then tell you the specific lines.
17:49.57ManxPowertzafrir_laptop: and what I am wondering is what can be done to get people to read those docs?
17:50.00xacatecascool.  ok well, back to figuring out why this one phone doesn't like to hook up with my switch.
17:53.23tzafrir_laptopReduce noise. Make it easy to point to a specific part in them
17:53.38*** join/#asterisk StephenF (n=none@198.144.201.106)
17:53.46tzafrir_laptopMake it possible for you to point them to the parts that are relevant to them
17:54.40PTorres[TK]D-Fender: we took another working * box (with a different -2port board- ) there, plugged everything and still having the same problem... :(
17:54.54ManxPowertzafrir_laptop: and I replied by saying that I cannot point you to a specific section of the readme.  Can you point me to a specific section of the  manual for my car?
17:55.17[TK]D-FenderPTorres: Not sure what to advise at this point.
17:55.22PTorresok
17:56.02ManxPowerI'm not asking them to read War and Peace.  I'm asking them to read the one document whose sole purpose is to help a person upgrading their asterisk.
17:57.31awkhi anyone experieince this, call comes into reception, and transfered, I nly get the recording fromt he first leg of the conversation... is there a mod I can add to asterisk to allow prevention of this, like 1.6 cdr does?
17:57.31[TK]D-FenderOne thing that would really help is to ditch TEX.  FFS leave these as boring text files.
17:59.08ManxPowerawk: Wow, I almost understood that.
17:59.17tzafrir_laptopManxPower, here's an example: I searched yahoo (hey, let's not use g exlclusively) for asterisk "UPGRADE.txt"
17:59.41ManxPowertzafrir_laptop: yet again you are searching the web for info already on your harddrive.
17:59.52tzafrir_laptopThe first hit that seemed relevant and not too verbatim is: http://www.russellbryant.net/blog/2008/09/03/new-versions-of-asterisk-asterisk-addons-zaptel-and-dahdi/
18:00.02awkif i'm switching channels, this is tricky, I need some mod for asterisk to keep a track of unique id's and dum that into a cdr, something like SIP -> SIP -> ZAP or SIP -> SIP (tx) SIP
18:00.06awkManxPower there.
18:00.20*** join/#asterisk Ariel_Calzada (n=aricalso@200.71.48.212)
18:00.28ManxPowerawk: so really nothing whatsoever to do with a transfer and a recording
18:00.41tzafrir_laptopmy point is that where people write "users are strongly encoureged to read in UPGRADES.txt about foo" they badly miss the options for:
18:00.42awkofcourse, because I need to join the recording...
18:01.08awksip -> sip works, zap -> sip doesnt work.
18:01.13awkzap -> sip -> sip
18:01.18tzafrir_laptophttp://example.com/UPDATES.txt#_foo
18:01.22ManxPowerawk: BEST of luck with that problem.
18:01.40ManxPowerNot Found The requested URL /UPDATES.txt was not found on this server.
18:01.45*** join/#asterisk hfb (n=hfb@96.247.65.63)
18:02.07awkManxPower how does anyone else deal with it, ManxPower does your recordings that come in and have been transferd you have the whople length of the conversation? so I can reference the cdr have 1 entry an attach the recording to this?
18:02.07ManxPowertzafrir_laptop: I still don't understand why you think this is a bad idea.
18:02.09PTorres<PROTECTED>
18:02.25tzafrir_laptopManxPower, in fact, they had to create there a separate text file for zaptel->dahdi and not make it part of the standard UPDATES.txt just because you can't reference people directly to part of UPGRADES.txt
18:02.25ManxPowerawk: BEST of luck with that problem.
18:02.56tzafrir_laptopawk, "doesn't work" ==?
18:03.13tzafrir_laptopcan you try palying with the jitter buffer settings?
18:03.27awktzafrir_laptop me?
18:03.55awkwhat has jitter buffer got to do wit this
18:04.11awktzafrir_laptop its only issues with cards, if I use a quintum gateway I get the whole leg of the conversation.
18:04.21awkso I need to find a way to get it to work from zap -> sip -> sip
18:04.22tzafrir_laptopI recall it was responsible for some voodoo issues related to zap->voip
18:05.20awkhmm, intresting
18:05.25awkim prepared to tr anything :)
18:08.08awkcould this only change things with problem on high jitter problem environment....
18:08.08awk?
18:08.08ManxPowertzafrir_laptop: Line 169
18:08.09*** part/#asterisk nikko (n=nikko@69.57.49.100)
18:08.16tzafrir_laptoptr? I think sed is more potent
18:08.36ManxPowertzafrir_laptop: I have now referred you to a specific section of UPGRADE.txt that deals with changes to one of the applications you use.
18:12.43tzafrir_laptopManxPower, what version of Asterisk? I just checked, and line 169's content has changed on 1.4.1 , 1.4.12 and 1.4.21
18:12.52tzafrir_laptop(in the timeline of 1.4)
18:13.02*** join/#asterisk hmmhesays (n=hmmhesay@71-32-29-217.farg.qwest.net)
18:13.40ManxPowertzafrir_laptop: Exactly!  See how silly your need to refer to a specific section is.
18:14.06ManxPower(I was looking at 1.4.13, BTW.
18:14.37*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
18:14.40tzafrir_laptopIt's os silly that I do it all the time
18:15.08ManxPowertzafrir_laptop: I guess we could just remove all the UPGRADE.txt files since they are so useless anyway.
18:15.32ManxPowerLet the poor sods figure it out themselves rather than trying something that might help a few people.
18:20.18hardwiresigh
18:20.23hardwireI'm using a vegastream for my keyboard riser
18:24.41ziram19hi
18:24.43ziram19>http://pastebin.com/m2cf4fc6c
18:24.51ziram19can u have a look
18:25.24ziram19the is happens when mu phone aastra calls st2030
18:25.40ziram19when st2030 calls aastra it works
18:25.46ziram19strange no?
18:28.31ManxPowerziram19: turn off sip debug, add a noop to show the value of DIALSTATUS and HANGUPCAUSE and pastebin the cli output
18:29.10ziram19ok
18:31.11awktzafrir_laptop I found that ticket about jitter buffer but it was also set to add 't' to monitor but we use mixmonitor and no t option
18:31.42*** join/#asterisk lejun (n=lejunhu@66.178.134.235)
18:36.17ManxPowerawk: for reasons too complex to go into right now try adding the t option.  If that fixes it at least you have one more datapoint to use.
18:38.36awk<PROTECTED>
18:38.36awk<PROTECTED>
18:38.42awkif specified, it allow user to do *1 / *2 transfer...
18:38.50awkif you can do transfer, it set....
18:38.56awkso this doesn't help
18:39.41[TK]D-Fenderawk: and next time... PASTEBIN
18:39.56awk[TK]D-Fender its 2 lines
18:39.57awkcmon
18:40.43[TK]D-Fender"t" should have no impact on Monitor.  Either way the same stream is going through *
18:41.58awkhttp://bugs.digium.com/view.php?id=4528 not with bticket in 2005 :)
18:42.03awkt fixed his recording
18:42.10awkbut he is using monitor im using mixmonitor
18:42.25awkbticket/ticket
18:42.36xacatecasfrom a cdr module, is it possible to gain access to the channel variables?
18:42.46[TK]D-FenderResolution   no change required
18:42.54ziram19ManxPower http://pastebin.com/m653d8b80
18:43.15codefreeze-lapxacatecas: what, in C code?
18:43.19xacatecasyes.
18:43.20ManxPower[TK]D-Fender: I'm wondering of maybe one direction has been reinvited and the other not.
18:43.38awk[TK]D-Fender a->b do a call... it create 1234.wav, b transfert to c ... it create 4567.wav, on CDR, we see a call from A->C, but not A-B, B->C and A->C
18:43.49xacatecascodefreeze-lap, actually, i'd like it to be C++ but am willing to live without the STL.
18:43.55[TK]D-FenderManxPower: Monitor explicitly prevents reinvites from happening... just like "tTwW", etc
18:43.56awktthats basically my issue
18:44.55[TK]D-Fenderziram19: Fix your phone.  It is rejecting the call.  Either a targeted rejection, improper configuration, or DND.
18:44.57*** join/#asterisk beek (n=klinebl@65.211.106.242)
18:45.39ManxPowerziram19: standby I'm looking at it.
18:45.40[TK]D-Fenderawk: Resolution   no change required  <- seems to say it....
18:46.00awke using mixmonitor
18:46.02[TK]D-Fenderawk: And don't forget, there was no JB back then
18:46.03awknot monitor
18:46.19awk[TK]D-Fender I believ I need to look at this http://www.asterisk.org/node/48358
18:46.28awk[TK]D-Fender so what would you sugest in my case?
18:46.54*** join/#asterisk nikko (n=nikko@69.57.49.100)
18:47.56[TK]D-Fenderawk: Might have mixed associations there...
18:48.11[TK]D-Fenderawk: I'm off this one either way... probably for the best :)
18:49.16*** join/#asterisk chigital (n=chigital@tmo-100-69.customers.d1-online.com)
18:49.50awk[TK]D-Fender ;P i'll give my outcome when I test this... I can't wait for 1.6 to become stable.. 1.4 isn't even stable yet lol :)
18:50.57awkis CEL only on 1.6?
18:51.12Corydon76-digCEL is a branch
18:51.28awkahh
18:51.30*** join/#asterisk voxter (n=voxter@76.77.95.2)
18:52.19awkI know somebody who wrote a mod for asterisk for this problem, but not in the order I need it.. maybe I can get this and give to digium to certify
18:52.34awkcdrfix5 branche no more exist... commited on 1.6 :(
18:53.45*** join/#asterisk stoffell (n=stoffell@d51A4D575.access.telenet.be)
18:53.55seanbrightCEL
18:53.59seanbrightit's a different branch
18:54.47ManxPowerziram19: something really weird is happening on your phone.  Hangup Cause 19, according to the docs, only happens when the destination returns that code.  Maybe the phone has DND enabled or CF enabled or just plain screwed up.
18:54.49awkim starting to think the only way foward is to move to 1.6
18:54.55awkvery scary
18:55.24ziram19DND is dialplan?
18:55.31[TK]D-Fenderziram19: No, the PHONE.
18:55.38ManxPowerziram19: all those would have been set on the phone.
18:55.42[TK]D-Fenderziram19: Like I have told you several times before.
18:56.17ManxPower[TK]D-Fender: But I don't like your answer!
18:56.28ManxPowerSo it must be wrong!
18:56.29ziram19but i configure the same phone on my sip provider and it ringing when i call the number
18:56.30[TK]D-FenderManxPower: :)
18:56.53awkseanbright do you have any sugestions to throw at me or Corydon76-dig ? rsolving this issue on 1.4?
18:57.05awkor must i move to 1.6?
18:57.23seanbrighti have no suggestions and no idea what you are even talking about right now
18:57.31seanbrighti just saw that you were looking for CEL
18:58.16awkno, issue with call recording.. a->b do a call... it create 1234.wav, b transfert to c ... it create 4567.wav, on CDR, we see a callfrom A->C, but not A-B, B->C and A->C. s
18:58.36awkusing mixmonitor all from my research I see its a known issue on 1.4 and we must use CEL
18:59.03seanbrightok
18:59.06awkhttp://www.asterisk.org/node/48358  this is 18 months old and CEL branch no longer exsists
18:59.06seanbrightshrugs
18:59.32seanbrightawk: http://svn.digium.com/view/asterisk/team/group/CDRfix5/
18:59.34seanbrightit moved
18:59.38awk:D
19:00.00awk*BEAM*
19:00.10seanbrightbut based on the dates, it looks like the automerge may have stopped working
19:01.25awkok, thanks appreciate the help
19:03.27*** join/#asterisk mvanbaak (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
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19:16.56xacatecaswell, i'm off, thanks to all the bits and pieces of advice.
19:25.41lejunhey all, has anyone run into an issue where asterisk gets stuck, I'm not quite sure where or why it does.  When it does, calls continue to come in, but it doesn't try to connect an agent to them and the agents sit in 'not in use'.
19:26.17lejunIt won't reload after that, and any other tries to reload gets the 'last reload didn't finish'
19:26.29lejunseems the only way to get it back to normal is to kill the process and restart asterisk =\
19:28.04jameswfFAIL http://www.impeachmccain.com/
19:28.18stoffelllejun, have a loko at this: http://www.voip-info.org/wiki-Asterisk+debugging
19:28.22stoffelllook :d
19:28.27lejunk =)
19:28.27edoceoI have Aastra 480i phones, how can I write messages to their LCD displays?
19:29.43*** join/#asterisk n3hxs (n=HAMming@63.68.135.4)
19:39.54*** join/#asterisk mocker (i=ksexton@198.247.173.227)
19:40.26lejunah, a deadlock.. that's the term ;)  now to find out why.  thanks stoffell
19:41.30mockerCan someone take a look at this PRI debug and let me know if it's a telco problem or my problem? http://pastebin.ca/1246119
19:41.38mockerSome calls coming in are getting 'All circuits are busy'
19:41.51mockerAsterisk shows all channels clear.
19:43.43*** join/#asterisk oej (n=olle@ns.webway.se)
19:43.47[TK]D-Fendermocker: < Called Number (len= 7) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0)  '1201' ] <--- who are you calling?
19:44.05[TK]D-Fendermocker: Certainly doesn't look like a PSTN #
19:44.08mocker[TK]D-Fender: 1201 is my extension..
19:44.13mockerSame call goes through sometimes though.
19:44.16[TK]D-Fendermocker: Connected to another PBX?  Sure that #'s valid?
19:44.34[TK]D-Fendermocker: Perhaps the PBX thinks you're busy...
19:44.45mockerHappens to every extension.
19:44.53mockerI'll get a capture of a good call too.
19:45.05hmmhesaysNov  5 19:44:13 NOTICE[28703]: channel.c:1904 ast_read: Dropping incompatible voice frame on IAX2/voipjet1-1 of format ulaw since our native format has changed to g729
19:45.10hmmhesayswhat the hell is this
19:45.11gramulhaozinTELIAX SUCKS
19:45.28gramulhaozin[TK]D-Fender: hey man, what provider do you recommend ?
19:45.37beekgramulhaozin: I assume you're having the same problem -- they're system is offline.
19:45.40[TK]D-Fendergramulhaozin: What do you need?
19:45.47gramulhaozinneed a new provider
19:45.49gramulhaozinI'm up now
19:46.00gramulhaozinbut they turn off their switches without no prior notice
19:46.02hmmhesaysit accepts the call as g729 and gives me that shit
19:46.07[TK]D-Fendermocker: this is * calling OUT to another PBX right?
19:46.12gramulhaozinit sounds like they didn't paid the bill to internap and internap turned them down
19:46.30gramulhaozin[TK]D-Fender: need a provider for USA calls that is reliable
19:46.39*** part/#asterisk Ariel_Calzada (n=aricalso@200.71.48.212)
19:46.43[TK]D-Fendergramulhaozin: What kind of volume, etc?
19:46.44gramulhaozinI'm with teliax for more than a year but that's the third time that a problem like that happens
19:46.51gramulhaozin[TK]D-Fender: 10 lines
19:46.58gramulhaozin10 simultaneous calls
19:47.00gramulhaozinmax
19:47.06[TK]D-Fendergramulhaozin: In?  out?
19:47.11gramulhaozinin/out
19:47.13[TK]D-Fendergramulhaozin: les.net is pretty solid
19:47.28gramulhaozinhave you used them ?
19:47.34[TK]D-Fendergramulhaozin: Clients of mine do.
19:47.43gramulhaozingood
19:47.44[TK]D-Fendergramulhaozin: never heard a complaint
19:48.04gramulhaozinbecause I ported customers from att to Teliax and now I'm getting the blame
19:48.20mocker[TK]D-Fender: This is incoming call from PRI to extension
19:48.22mockerhttp://pastebin.ca/1246131
19:48.27mockerThat's a good call.
19:49.09[TK]D-Fendermocker: To clarify, these are all INBOUND attempts?
19:49.33mockerer...
19:49.39mockerThat may have been a bad second pastebin.
19:49.50mocker[TK]D-Fender: Yeah, inbound from cell phone to asterisk PRI
19:50.28gramulhaozingoing to migrate from teliax to LES.NET
19:50.35mockerNo, that was a good second paste, I'm just looking at the old one. :)
19:50.42gramulhaozinthe international rates are higher on LES but the USA is less expansive
19:50.48[TK]D-Fendermocker: pastebin yuor zapata.conf & dialplan
19:50.51gramulhaozin[TK]D-Fender:  do you recommend anyone for international ?
19:51.06[TK]D-Fendergramulhaozin: I have no specific advise for that.
19:51.21gramulhaozin[TK]D-Fender: do you know about a VOIP Provider Review wiki ?
19:51.33gramulhaozin[TK]D-Fender:  I think it's time for customers to know about those issues
19:51.48mocker[TK]D-Fender: Would be very hard, it's all realtime for the dialplan.
19:51.54gramulhaozinso people can think twice before signing up for an unreliable provider
19:52.20[TK]D-Fendermocker: ramp up core debug to max verbose too...
19:52.28mocker[TK]D-Fender: I have a call open w/ PRI provider.
19:52.34mockerI think they are getting the error as well.
19:52.52gramulhaozin[TK]D-Fender: What do you think about Broadvoice ?
19:52.59[TK]D-Fendergramulhaozin: Your issues could be considered rare to others.  Yes you are rfustrated, we get that, but its hard to quantify thngs like this.
19:53.20[TK]D-Fendergramulhaozin: There is no review site that I'm aware of, though there is a fair enough chance that one exists
19:53.29gramulhaozin[TK]D-Fender: I'm not really frustrated, but I bet there is 100 customers over teliax with the same problem
19:53.45[TK]D-Fendergramulhaozin: I can tell you that BV is near the lower end of the list.
19:54.07[TK]D-Fendergramulhaozin: They think too much in terms of "lines", and their proxies have been known to have issues
19:54.07gramulhaozin[TK]D-Fender: It would be nice to review those providers
19:56.38gramulhaozin[TK]D-Fender: yep, looks like Broadvoice is a bad option
19:57.07*** join/#asterisk wiscados (n=mint@81.25.184.155)
19:57.18*** join/#asterisk Bullterd (n=IceChat7@62.249.223.152)
19:57.20BullterdHey All
19:57.52BullterdI keep getting this error: [Nov  5 19:39:51] NOTICE[14339] chan_sip.c: Call from '4052' to extension 's' rejected because extension not found.
19:58.09BullterdI havent specified extension 's' anywhere :S
19:58.17BullterdI used the GUI to configure it all
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20:00.02gramulhaozin[TK]D-Fender: do you know any proxy from les ? Just wanted to check the latency from here to Canada (South Florida)
20:01.15[TK]D-Fendergramulhaozin: Googleable.. I don't ahve the answer handy
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20:01.57[TK]D-FenderBullterd: because when you did your REGISTER to your provider you did not tell them to send the incoming call to a numbered extension.  * therefor tells them "s"
20:02.00*** join/#asterisk StephenF (n=none@198.144.201.106)
20:02.00*** part/#asterisk fransman (n=frans@a80-127-14-241.adsl.xs4all.nl)
20:02.15[TK]D-FenderBullterd: and the GUI does a patchwork job of configuring *.  You have much to do  & learn by hand
20:02.38[TK]D-FenderBullterd: Also read the channel topic for the list of GUI support channels
20:02.39gramulhaozinhehe
20:02.51*** join/#asterisk StephenF (n=none@198.144.201.106)
20:02.52gramulhaozinDTMF problems with LES pbx :P
20:04.44gramulhaozin:P
20:04.55gramulhaozinlooks like getting a T1 would give me the control :P
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20:20.46*** join/#asterisk Arsenick- (n=rpurcell@modemcable026.33-70-69.static.videotron.ca)
20:22.29Arsenick-Hi all, is there a way to find how many call have been lost using CDR or something like that, because the caller just did something weird and they hangup ?
20:25.59*** join/#asterisk wiscados (n=mint@81.25.184.155)
20:26.12[TK]D-FenderArsenick-: You won't know why they hung up.... so that kills the thogh of it as a reliable trouble-shooting resource
20:27.39Arsenick-for now I'm using the filter to do this.. so call coming from zap, duration less then 30 sec etc.. it gives me good hint but as u said there's no way to know why the call ended..
20:28.49Arsenick-anyway thx for your anwser
20:28.59*** join/#asterisk wiscados (n=mint@81.25.184.155)
20:29.11[TK]D-FenderArsenick-: What kind of card?  Yougetting actual reports of dropped calls?
20:30.44Arsenick-it's digium TDM400P  but I don't get any report, I'm just getting call that are incomming and if they are short.. but it's not very accurate ;) anyway it's just for the beginning, I want to know if there's hole in my dialplan..
20:30.50Arsenick-menu etc..
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20:34.57PTorres[TK]D-Fender: our pbx with the nms (natural) board had the same problem... I think the telco will move them to r2 protocol, ( we asked for the change to isdn in the first place , it worked fine for 30ish days )
20:35.03[TK]D-FenderArsenick-: Ok, so you're fishing for trouble, not just "investigating" it ;)
20:36.21PTorresso... thanks :D
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20:48.43profxavierwhat is the default login info for a Polycom 330 ?
20:49.09beekprofxavier: You mean for the phone settings?
20:49.24beekprofxavier: if so, the password is 456
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20:50.42profxavierlogin is Polycom (case sensative)
20:50.45profxavierfound it, thanks
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21:00.04Arsenick-[TK]D-Fender, yeah exactly
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21:13.15saftsackis the t.38 support with the new patches for 1.6 as good as asterisk can act as a reliable t.38 passthrough gateway?
21:13.33*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
21:13.34jameswfI wonder how long it would take to sync 1.5TB
21:14.09[TK]D-Fenderjameswf: Let me pull out my acoustic coupler and find out!
21:14.25jameswfwaits...
21:15.21*** join/#asterisk johndo (n=hef@toucana.tavros.net)
21:15.42[TK]D-Fenderjameswf: 75 baud confirmed!  Let me get back to you when I'm done!
21:16.15johndowith asterisk 1.6.0.1 is it still only possible to use 1 trunk for incoming calls from a single service provider?
21:16.32[TK]D-Fenderjohndo: That makes no sense...
21:16.53johndocan I have multiple register => entries in sip.conf to the same server?
21:17.15[TK]D-Fenderjohndo: Yes, always could
21:17.17*** join/#asterisk MindTheGap_ (n=MindTheG@189.59.206.3)
21:17.26johndodoes it work correctly for incoming calls?
21:17.38johndonot according to http://bugs.digium.com/print_bug_page.php?bug_id=9678
21:17.51[TK]D-Fenderjohndo: Depends how they send youthe calls
21:18.28[TK]D-FenderResolution:   no change required  <--- clearly not a bug
21:18.58johndo(:   It a missing feature, not a bug I guess
21:19.28[TK]D-Fenderjohndo: How incoming calls will work depends on how the call comes in.
21:19.44[TK]D-Fenderjohndo: So tell you what... show us a specific scenario and we'll work with that
21:19.50*** join/#asterisk fordfrog (n=fordfrog@gentoo/developer/fordfrog)
21:20.07johndook
21:21.34johndoI am building a box that sits between some spa941 hardphones and a trixbox server
21:22.28johndothe box is supposed to be transparent to the phones and to trixbox
21:23.03StephenFwhy would you put asterisk between phones and trixbox?
21:23.43johndotrixbox doesn't offer call recordings
21:24.15giovaniyes it does
21:24.24johndofor calls coming into a queue only
21:24.26jameswfummmm trixbox is asterisk
21:24.27johndoas far as automated goes
21:24.33giovania) trixbox is just a wrapper around asterisk
21:24.40giovaniyou can modify asterisk configs to your hearts content
21:24.53giovanib) trixbox DOES have options in the gui for call recording
21:24.54johndonot on the trixbox pro cce edition
21:25.09jameswfdont use that
21:25.11johndoany changes I make to the gui overwrite the configs
21:25.18giovanijohndo: no ...
21:25.25giovaniyou're supposed to use the manual configs
21:25.29giovanithat are included in the auto-written ones
21:25.41johndowhere are the manual configs?
21:25.44giovanithis belongs in #trixbox
21:25.54[TK]D-Fendercheckout time... later all...
21:26.13johndowell, I wasn't worried about trixbox, I was worried about the asterisk box origanally
21:26.31giovanidon't do what you're planning
21:26.46johndogiovani: why not?
21:26.54giovaniputting a vanilla asterisk install in between phones and trixbox is the most conveluted and silly solution to recording calls
21:27.05johndogiovani: I completely agree
21:27.11giovaniso talk to trixbox
21:27.11johndogiovani: I did not choose the assignment
21:27.14giovaniabout your recording needs
21:27.18giovanithey will fix it
21:27.19johndogiovani: I did
21:27.24johndogiovani: they said it was not possible
21:27.24Qwells/^putting.*and //
21:27.44giovanianything done in asterisk can be done in trixbox, it's just a matter of difficulty
21:28.10Micchow many kbps is a ulaw call?
21:28.14Qwellgiovani: that is not the case.
21:28.17johndogiovani: every time a change is made on the trixbox cce website, they connect to the trixbox server via ssh and overwrite config files
21:28.50giovanijohndo: so they can make this change to your configs, you're a customer
21:28.55StephenFMicc, I think about 90 Kbps...
21:29.11MiccStephenF, that seems kinda high.
21:29.17StephenFhmm lemme check
21:29.30Qwell80 with overhead
21:29.33MiccI found it.
21:29.42MiccStephenF, it says 64kbps on this site
21:29.44StephenFyeah, there ya go. 90 is being safe
21:29.49Micchttp://hostseries.com/asterisk-understanding-codecs/
21:30.05Miccyeah maybe with all the overhead.
21:30.18johndoI am of the same mindset.  I consider the task convoluded and overly complicated.  the original design was actually worse.
21:30.40Qwelljohndo: so, why don't you just drop trixbox?
21:30.52MiccIs there a way to setup QoS between my customers sip phones and my asterisk box in a datacenter or is it unpredictable?
21:30.58johndoQwell: I can replicate the features of HUD
21:31.08johndo*I can;t replicate the features of HUD
21:31.54StephenFMicc: I think that would depend on your equipment. (Phones, routers, etc...)
21:32.14johndothe original design was spa941 -> trixbox -> t1 -> t1 recorder -> t1 -> cisco t1 to sip -> t1 data line to provider
21:32.21MiccStephenF, wouldn't it depend also on the equipment between the customer and my datacenter?
21:32.23StephenFMicc: I'm pretty sure Polycom phones tag their voice packets with a specific DSCP tag, which some routers can use to apply QoS
21:32.30StephenFMicc: right thats what i mean
21:32.54MiccStephenF, do you know if Aastra phones do that too?
21:33.04johndoQwell: If I could get hudlite to work I would drop it in a hearbeat, but I was unable to do that
21:33.10StephenFMicc, No sorry, im talking about equipment at each end, the stuff in between doesnt matter.
21:33.38MiccOk, good to know. So it is possible if we are in control of what equipment is at both ends.
21:33.44StephenFYes
21:33.58MiccBut there is still the possibility of a DOS attack.
21:34.17MiccOr would QoS handle a DoS attack as well?
21:35.30*** join/#asterisk sp00k3y (n=sp00k3y@wsip-98-190-136-194.ph.ph.cox.net)
21:35.32hardwireanybody set up heartbeat to monitor asterisk?
21:35.33StephenFDoS attack on your cleints network?
21:35.45hardwirenot sure how I want to measure quality changes'
21:35.55MiccNo, on my datacenter network. I wouldn't think my clients are targets.
21:36.03MiccAnd if they are, that is their problem.
21:36.10StephenFohh ok, yea I was wondering what kind of client this was...
21:36.22Miccjust small business customers.
21:36.24johndoon an unrelated note:
21:36.24sp00k3yhey all, whats the longest amount of time a voicemail message can be?
21:36.35johndodo any non-linksys phones do remote provisioning
21:36.37StephenFumm, I dont know how you would prevent DoS attacks at your data center
21:36.44StephenFjohndo yes, most brands do
21:37.04Miccsp00k3y, I think you can set it to whatever time you want.
21:37.11johndoStephenF: I did not find a way to do it for cisco or polycom, at least one that was not dhcp based
21:37.20Miccsp00k3y, as long as you have the disk space for it, I don't think there is a limit.
21:38.03sp00k3yMicc: i see
21:40.19Miccsp00k3y, look in voicemail.conf maxmessage
21:40.34sp00k3yyeah im looking there now lol thanks :)
21:40.43Miccnot to be confused with maxmsg
21:40.50StephenFjohndo, ok. Im not sure what you are asking then...
21:41.22johndoStephenF: linksys phoens can pull their configs from an xml file over http(s)
21:41.37johndomeaning I can configure them anywhere in the world.
21:41.58johndoI have stuff done for spa941 and spa942, I was wondering there were others I could add to my list
21:42.01StephenFjohndo: polycoms pull their xml configs from an ftp/tftp server (possibly http as well, never tried that)
21:43.11StephenFpretty sure cisco phones do a similar thing...
21:44.32johndoStephenF: there seems to be a newer style cisco phone that appear to be identical to the spa series
21:44.40johndoI have not gotten a hold of those
21:44.47johndothey are also about 3 times more expensive
21:45.41Ritzeriskpenguins :)
21:45.48johndo??
21:46.22jameswfholy crap california passed prop 8
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21:52.11sp00k3yugh, so i have a lcient who wants to use their PBX for dictation.  They need a minimum of 15 recording time most likley more than that.  They aslo want the recordings to be emailed to their office worker who will then download the sound file and load it into playback software to type it out on her pc.  What would be the bets way to tell them that their idea is FUBAR'd?
21:52.21sp00k3y15 min*
21:54.48*** join/#asterisk [TK]D-Fender (n=chatzill@64.235.218.194)
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21:57.43*** part/#asterisk PTorres (n=PTorres@200.68.87.146)
22:01.46ManxPowersp00k3y: when was the last time you looked thru the output of "core show applications"?  Pay special attention to the apps starting with a D
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22:02.27*** join/#asterisk chigital (n=chigital@tmo-100-169.customers.d1-online.com)
22:05.53sp00k3yright, i know about the built in dictation
22:06.23sp00k3ythis is the way they wanted it
22:06.33sp00k3ythey didnt want to have to name the files themselvs
22:06.43sp00k3ycuz they like ot be annoying
22:08.29sp00k3yim more concerned with the 15+ min WAV file they will be trying to email
22:08.29*** join/#asterisk bipolar (i=bflong@216-164-162-138.pa.subnet.cable.rcn.com)
22:08.39sp00k3yconsidering they record every call that comes into their PRI as it stands right now, and adding all that is going to eat up system resources, and they're too cheap to upgrade their hardware
22:09.16hardwirehmm.. I have an iax friend pointing to another IP on a subnet connected to the * box
22:09.21hardwireit's 1.2
22:09.28hardwirethe other IP is running 1.4
22:09.30*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.static.bezeqint.net)
22:09.45hardwireand it has an IP alias on top of another IP for that specific IP
22:09.48hardwireheh.. if that makes sense
22:10.21hardwirehmm.. pastebin time
22:10.42hardwireerr.. subway time.
22:10.54sp00k3ythat sounds good
22:11.05sp00k3yme too maybe i'll be less frustrated if i eat something
22:11.41*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
22:13.11*** join/#asterisk jer (n=jer@unaffiliated/jer)
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22:14.25jayteequittin time, be back later
22:23.53profxavierguys, in setting up a Polycom 330, I found that in sip.conf, i need to set nat=yes.  Why is that? On my Grandstreams (old phones -- moving to the Polycom) I do not need nat, but on softphones, we do.
22:25.18[TK]D-Fenderprofxavier: is the phone behind a remote NAT?
22:26.18*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
22:26.43profxaviernot sure, im not the senior admin
22:26.57profxavieris that a remote location, from the asterisk server ?
22:27.16[TK]D-Fenderprofxavier: yes
22:28.12profxaviernope, within the same LAN
22:29.39[TK]D-Fenderprofxavier: then there is no need for nat=yes
22:29.50profxavierok, ill recheck my settings
22:34.18*** join/#asterisk CGMChris (n=chris@mail.cgmyes.com)
22:35.45hardwirehmm.
22:35.57hardwiredoes the asterisk iax channel store it's own arp?
22:36.17CGMChrisI am trying to figure out how to configure my phones to indicate if line 1, line 2, line 3, etc. are in use (via the line indicators on my phones).  I am not using traditional lines, only SIP trunks.  Where do I set this up: in Asterisk or on the phones themselves>?
22:36.48CGMChrisTo clarify, i want all 4 phones to show when someone is using Line 1, or line 2, via their indicators
22:37.53*** join/#asterisk Pazzo (n=ugelt@sadsl-246059.rol.raiffeisen.net)
22:38.35[TK]D-FenderCGMChris: Will only work if each uses a separate peer.
22:38.37*** join/#asterisk etech3 (n=chatzill@68-243-103-134.area7.spcsdns.net)
22:38.55StephenFCGMChris: what is the reason for wanting that functionality?
22:39.00Kattyjbot: asterisk standard extensions
22:39.03*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
22:39.07CGMChrisStephenF: old habits die hard.
22:39.10Kattyjbot: useless!
22:39.10jbotACTION starts crying and hides from katty in the darkest corner of the room. :(
22:39.15Kattydigs out book
22:39.19[TK]D-FenderCGMChris: And it requires you to set up dialplan "hints" (go read about Presence on the WIKI") and for you to setup your phone to subscribe to them
22:39.44StephenFWhat is the habit? Press the available line to place an outgoing call?
22:39.50hardwireanybody using ip aliases and iax?
22:40.01*** join/#asterisk jasonwoot (n=jasonrot@bookit-dev.com)
22:40.01Kattywell joy. not sure what to look under
22:40.25CGMChrisStephen: The habbit is one person places the call on hold, and another person picks it up by pressing that line #
22:40.25Kattywhat would the I Entered A Number On The IVR that does not exist, so send me to /here/ be listed as?
22:40.30Kattysomething tells me i
22:40.31StephenFahh, I see
22:40.48StephenFCan't retrain them to use call parking?
22:40.58Kattylooks under IVRs
22:41.11CGMChris~wiki
22:41.21StephenFKatty, thats the "i" extension
22:41.44CGMChris[TK]D-Fender: en.wikipedia.org/wiki/Asterisk ?
22:41.44StephenFum, i for invalid I guess... is that what you mean?
22:42.00*** join/#asterisk jov4n (n=jovan@host119-99-dynamic.17-87-r.retail.telecomitalia.it)
22:42.12Kattyjbot: i
22:42.13jbotyou are probably the only girl in the channel, so be nice to her
22:42.24Kattyoh for goodness sake.
22:42.26StephenFlol
22:42.29jov4nHi
22:42.30kerx~thebook
22:42.31jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com
22:42.31Katty[TK]D-Fender: where are the standard extensions in the book?
22:42.35kerx~telecom101
22:42.42kerx~telco
22:42.45kerx~telecom
22:42.55kerxdarn
22:42.58kerx~help
22:42.59StephenFCGMChris: http://www.voip-info.org/
22:43.26CGMChrisStephenF: found it, thanks.
22:43.49Micchow do I run a shell script from and extensions.conf line?
22:44.00[TK]D-Fender~101
22:44.01jboti heard 101 is Telephony 101, which is a good read if you're unfamiliar with traditional TDM telephony.  You can download it at http://www.stromcarlson.com/docs/basics/NTtelephony101.pdf
22:44.11Kattyjbot: Invalid Entries and Timeouts are located on page 85 of the book. Specifically the i and t extensions.
22:44.12jbotKatty: okay
22:44.15[TK]D-FenderMicc: "core show application system"
22:44.27Kattyjbot: invalid
22:44.28jbotsugar quote ibot
22:44.44kerxTK: thanks :)
22:44.51Kattyjbot: invalid entries and timeouts
22:44.51jbotinvalid entries and timeouts are located on page 85 of the book. Specifically the i and t extensions.
22:44.52kerxDo you know if there is any better Telecom books?
22:44.54Miccthanks. TrySystem looks like it might do the trick.
22:45.06StephenFPage 85? my book its on page 130...
22:45.09kerxSomething much more advanced
22:45.16kerxI've already read that telecom 101 book now
22:45.18Kattyjbot: forget invalid entries and timeouts
22:45.18jboti forgot invalid entries and timeouts, Katty
22:45.35Kattyjbot: Invalid Entries and Timeouts are located on page 85 OR possible 130 of the book. Specifically the i and t extensions.
22:45.36jbotokay, Katty
22:45.39[TK]D-Fender~invalid entries and timeouts
22:45.40jbotinvalid entries and timeouts are located on page 85 OR possible 130 of the book. Specifically the i and t extensions.
22:45.41StephenFlol
22:45.41Kattygah!
22:45.45Kattyjbot: forget invalid entries and timeouts
22:45.45jbotKatty: i forgot invalid entries and timeouts
22:45.55[TK]D-FenderKatty: its not just the FIRST word
22:45.58Kattyjbot: Invalid Entries and Timeouts are located on page 85 OR possibly 130 of the book. Specifically the i and t extensions.
22:45.59jbotKatty: okay
22:46.01Katty[TK]D-Fender: yes. i know.
22:46.39Kattywe need something for asterisk standard extensions too
22:47.02kerxAny really good Voip book recommendations, or online ebooks?
22:48.11Kattykerx: how to thwart aliens with foil hats, by the discriminating lunatic.
22:48.20kerxlol
22:48.31gcbirzanThat should be a short book.
22:48.42Kattybasically a page with folding instructions
22:49.02gcbirzan"Don't. I for one welcome our alien overlords."
22:49.04Kattyand some fine print.
22:49.47Kattyhey! it's time to go home, have dinner, and go play pool!
22:49.53kerxhave fun!
22:50.20ZhadWhat's the difference between ExtenSpy and ChanSpy?
22:50.50ManxPowerZhad: Well, what does the application documentation say about it?
22:50.53lesouvageKatty: see http://www.voip-info.org/wiki/view/Asterisk+standard+extensions that is all there is to know.
22:51.01[TK]D-FenderZhad: read their instructions, I'm sure its as blatant as it says
22:51.01Zhad[Synopsis]
22:51.01ZhadListen to a channel, and optionally whisper into it
22:51.05Zhadon both of them.
22:51.12ManxPower"core show application X" in the Asterisk CLI
22:51.30Zhadthey both say the same thing
22:51.56Zhadwell, very nearly
22:52.05ManxPowerZhad: very nearly is not "the same"!
22:52.36*** join/#asterisk jtodd (n=jtodd@blob.fox-den.com)
22:52.42[TK]D-FenderManxPower: They are the exact same thing except for the differences!  WTF!?!?!
22:52.45ManxPowerI'll bet one requires a channel and one requires an extension and I've not even gone and looked the docs.
22:53.46ManxPowerI imagine a newbie might confuse a channel and an extension.  Unfortunately until you understand the differences you'll never be successful with Asterisk.
22:55.58Zhad<PROTECTED>
22:55.59Zhadaudio from an Asterisk channel.
22:56.07Zhadfrom core show application Extenspy.
22:56.19ManxPowerZhad: Ok, so that application expects and extension and context.
22:56.49Zhadiirc Chanpy doesn't wan't a channel as the argument either
22:56.53ManxPowerextensions are, of course, the part between the exten => and the first comma
22:56.54Zhad-'
22:57.05Zhadshould go to bed
22:57.14ZhadI know
22:57.25ManxPowerZhad: and chanspy expects a channel.
22:57.34*** join/#asterisk RypPn (i=TuMbL@rosscom.demon.co.uk)
22:57.50ManxPowerwell, technically it would want a channel instance (the instance is the part after the - in the channel name)
22:58.00Zhadand unless it's changed, Chanspy wants an extension as th4e argument and it doesn't work with extensions that have AGI scripts running or has Answer as the first line (although that was probably fixed a long time ago).
22:58.18ManxPowerso a channel would be Zap/1-1 or SIP/happyphone-ajf08934ns
22:58.23Zhadknows
22:58.28*** join/#asterisk nicoAMG (n=superunk@201.203.50.42)
22:58.36ManxPowerZhad: Chan spy does NOT want an extension.
22:58.42ZhadIt certainly used to
22:58.47ManxPowernever did
22:58.56ManxPowerthat is the whole reason extenspy was created.
22:59.24ManxPowerSo what channels are you trying to spy on that you can't seem to?
22:59.38ZhadI ahve no problem with chanspy
22:59.46ZhadIt gets used quite regularly
22:59.55ManxPowerand an example of that usage is?
23:00.02ZhadI just notcied someone talking about extenspy on the mailing list.
23:00.26ZhadUsually it's the boss at home listening in on a member of staff he wants an excuse to sack.
23:00.30*** join/#asterisk jer (n=jer@unaffiliated/jer)
23:00.32Zhadhe's a bit weird like that.
23:00.51Zhadgoes to bed.
23:01.10ManxPower[TK]D-Fender: another failure to enlighten. 8-(
23:02.21[TK]D-FenderManxPower: I have the solution for enlightenment....
23:02.24ZhadSorry, very tired. Was getting confused with Pickup (yeah I know, completely different App).
23:02.27[TK]D-Fender~fire
23:02.28jbotBender : Light a fire for a man and he's warm for a night.  Light a man on fire and he's warm for the rest of his life...
23:02.40[TK]D-FenderManxPower: Plenty of light now!
23:03.55*** join/#asterisk `Sean (i=Shaan@CPE0016d3fe20a0-CM001ade84fd0a.cpe.net.cable.rogers.com)
23:04.01*** join/#asterisk flush (n=SYN_SENT@ip216-239-67-136.vif.net)
23:04.19ManxPower[TK]D-Fender: I've mostly stopped even trying to correct misinformation on the mailinglists.
23:08.30[TK]D-FenderManxPower: You follow that?
23:08.33profxavierthanks Fender
23:08.37profxavierworked, no nat
23:09.31ManxPower[TK]D-Fender: Yup.  Less so than in the past.  It has a lot of GUI questions than I totally /ignore.
23:09.37etech3aastra sip phone popping and cracking on outside calls what should I check?
23:10.04ManxPoweretech3: Your outside lines.
23:10.13[TK]D-FenderetcShould check how it is you get "outside"
23:10.26etech3line are good
23:10.31[TK]D-FenderManxPower: Anything more interesting on the ML than here?
23:10.49[TK]D-Fenderetech3: Clearly how you get there isn't
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23:12.42harry_vcompiling 1.6 I am getting this error on make ael_main.c:90: error: `PATH_MAX' undeclared here (not in a function)
23:12.43harry_vplus 5 other related errors.
23:13.02etech3using a 4 port TDM card 2 CO and 2 stations
23:13.07harry_vGoogle does not seem to be comming up with some key words for this error
23:14.44[TK]D-Fenderetech3: if the lines are fine and the phones local, then your card is at fault
23:16.11`Seananyone ever heard of blaster dialer
23:16.21`Seanor some sort of dialer also being reffred to as blaster
23:17.01etech3I got a single line telephone plugged in to one of the stations and I dont notice this noise just on the aastra phones 2 of them
23:18.21harry_vSeems there is a possible issue in menuselect.h
23:19.52ManxPower[TK]D-Fender: It's sometimes funny how much some of the answers are just toally wrong
23:20.09[TK]D-Fenderetech3: So bridged zap is fine, otherwise static... again.. its the card.
23:20.22[TK]D-FenderManxPower: Sometimes not even funny...
23:20.51ManxPower[TK]D-Fender: Funny in the sort of way the Darwin Awards are funny
23:21.04[TK]D-FenderManxPower: I'm all for natural selection...
23:21.14[TK]D-Fendertosses another GUI user to the lions
23:24.58*** join/#asterisk mvanbaak_ (n=michiel@asterisk/contributor-and-bug-marshal/mvanbaak)
23:25.55etech32 different cards and 2 different systems and different phones codec or TX gain?
23:28.34[TK]D-Fenderetech3: if its static, then the card has issues.  its not just gains.
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23:28.53[TK]D-Fenderetech3: I've replaces several TD400's & TDM2400's because of this.
23:28.59[TK]D-FenderTDM400*
23:31.40etech3noise us when  the person speaks loudly
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23:32.04etech3ops and cracks not static
23:32.08etech3pops
23:33.28[TK]D-Fenderetech3: generally the same thing.  Poor audio around
23:35.35etech3SIP calls too
23:35.50etech3just on the aastra phones
23:36.53[TK]D-Fenderetech3: So between 2 Aastra's is bad?
23:37.36etech34
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23:38.18[TK]D-Fenderetech3: Then if phone to phone is bad then its the phone
23:38.38[TK]D-Fenderetech3: Its simple process of elimination
23:39.00etech3yea it's down the aastra config
23:39.41kerxwhat would the config for extconfig.conf be if i need it to connect the dialplan (extensions.conf) to mysql databse table?
23:39.54kerxhopefully it's possible :)
23:41.14carrarextensions => mysql,asterisk,extensions
23:41.29carrarassuming db asterisk
23:41.33carrarand table extensions
23:45.36kerxcarrar, where does it get the mysql connect info from?
23:45.54carrarfrom your include directory that you compiled the mysql driver
23:46.09kerxhrmm...
23:46.14kerxok, i got reading to do it seems
23:46.15carrarsorry
23:46.16carrarasterisk-addons
23:46.17kerxis there a howto for this?
23:46.26carrarsince mysql is not part if it by default
23:46.29[TK]D-Fender~book
23:46.30jboti guess book is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com or see ~buybook
23:46.31[TK]D-Fender^^^^^^^^^^^^^^^^
23:46.51carrarthere are a few if you google around
23:47.04carrarI use it with PostgreSQL
23:47.09kerxk, if tk posted the book, it must be in there and i missed it
23:47.12kerxthanks guys
23:51.25carraras it's not native
23:51.27carrarerr
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23:56.19hardwireweird. answer() -> echo() fails to establish two way audio but putting playback inbetween fixes it.

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