IRC log for #asterisk on 20110803

06:18.21*** join/#asterisk infobot (~infobot@rikers.org)
06:18.21*** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0-beta1 (2011/07/22), 1.8.5.0 (2011/07/11), dahdi-linux 2.4.1.2 (2011/04/11), dahdi-tools 2.4.1 (2011/03/03), libpri 1.4.12 (2011/07/06) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/
06:18.32WIMPyI guess that would make sense.
06:18.35ChannelZ(from what I understand that setting is really like a 'priority')
06:18.44cuscoright now they are all sequenced anywayz
06:18.49WIMPyBut I have no idea about the dahdi internals.
06:18.57ChannelZI wonder if the second card is trying to use sync from the first.  Me either.
06:19.00cuscoyes I have been reading it seemslike a priority but .. for the whole system or the rest of the card?
06:19.17cuscoI tried changing it many times
06:19.18cuscolol
06:19.24WIMPyChannelZ: It can't unless there's a timing cable.
06:19.37WIMPyBut it does sond like something.
06:19.40ChannelZHmm.
06:20.01ChannelZcusco: have you set up the first 4 spans as 1-4 and the second 4 as 1-4 as well?
06:20.07cuscook symptom maintains
06:20.07WIMPyIt IS a priority.
06:20.15cuscoChannelZ: I did that just now
06:20.21cuscoWIMPy: for the system or the card?
06:20.39ChannelZWIMPy: Yeah.. does it matter if the priority starts on "5" for instance?
06:20.46ChannelZI guess 0 really only has 'special' meaning?
06:22.30cuscohttp://paste.debian.net/124921/
06:22.43WIMPycusco: pardon?
06:22.50cuscothat 'ver-primarios' output is basically a dahdi_scan|grep alarm
06:23.18WIMPyChannelZ: I think it's just trying to use the i/f with the lowest configured priority (>0) if possible.
06:23.35WIMPyBut there have definitely been changes iregarding timing source selection in dahdi.
06:23.37ChannelZHmm.
06:23.54cuscook let me download older dahdi
06:23.57WIMPy0 is never used, yes.
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06:24.03schmidtsgood morning
06:24.13WIMPyMoin schmidts
06:24.23ChannelZAnd we know the 4 lines running into the card that isn't working work in the card that is.
06:24.37cuscodahdi-linux-complete-2.3.0.1+2.3.0/ - this is the one I previously had
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06:25.45WIMPyOr actually 0 is not only not used, but it provides timing.  I.e. for NT interfaces.
06:28.53cuscoI think that it being 'internally clocked' is the problem
06:28.59cuscoI'm not sure
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06:29.54ChannelZyes but the question is why
06:30.32ChannelZyou don't have your D channels screwed up do you?
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06:31.40cusco34.353023hu?
06:31.42cuscooops
06:31.44cuscohu?
06:31.51cuscoData channels screwed up?
06:32.03ChannelZyeah on the wrong channel or something
06:32.05cuscoHow would that matter if a span is not connected for instance
06:32.08cuscolol
06:32.11cuscono
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06:32.16cuscogenconf gets it right
06:32.20ChannelZI'm guessing no since all this worked
06:32.42ChannelZwell if a span is not connected you're not going to get sync either
06:32.58ChannelZAre you saying you're getting all these red alarms with nothing plugged in?
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06:33.30WIMPyWhere does dahdi tell you about 'internally clocked'?
06:33.44cuscoChannelZ: lol no, one of them has a cable plugged in
06:33.58cuscoWIMPy: dahdi_tool
06:34.12WIMPyI don't have dahdi_tool :-(
06:34.12cuscodahdi_scan shows syncsrc=0
06:34.31WIMPyAh
06:34.42WIMPyThat's what I see as well.
06:34.49cuscowhere?
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06:35.30cuscoworking pri shows synsrc=1
06:35.37ChannelZI don't know why, but that reminds me I need to get my MOH off a backup
06:35.43WIMPyLooks like dmesg is the only reliable information.
06:36.12WIMPyI get 0 on a working PRI, as well as an unconnected port.
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06:37.11cuscodmesg shows messages regarding changing timing sources
06:37.25cuscobecause when dahdi starts 1st card spans are yellow, and 2nd card are OK
06:37.28cuscobut only for 3 secs
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06:37.57WIMPyWhich indeed looks like it sees them as one.
06:38.22cuscohttp://paste.debian.net/124923/
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06:39.05ChannelZhmm
06:39.18cusco[  344.256561] wct4xxp 0000:04:01.0: All spans in alarm : No validspan to source RCLK from
06:39.24cuscothis is my problem
06:39.29WIMPyThere are no messages about sync source on the 2nd card.
06:39.42cuscoWIMPy: the one I just pasted
06:39.43ChannelZjust out of curiosity, how long have you had this setup with the 2 cards?
06:39.52cuscoChannelZ: for over a year
06:40.00WIMPycusco: Yes, there
06:40.14ChannelZand the second 4 always from the same (different) telco?
06:40.46cuscoChannelZ: yes, and some times more then 1 telco in it
06:40.58cuscoand then I realized I had timing problems and could only keep 1 telco per card
06:41.40cuscowhat can I do?
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06:41.48cuscocould it be the card?
06:41.50WIMPyWhich tells us that it would be neccessary to configure timing pools.
06:41.58cuscoseveral interrupt messages
06:42.10WIMPyMost probablyt no.
06:42.11cuscoWIMPy: how?
06:42.26WIMPycusco: Someone needs to invent something.
06:42.49ChannelZI'm wondering if the times it was working was only by accident
06:42.51cuscobut it had been working with no problems
06:43.02cuscowell I restarted it several times
06:43.05cuscoupgraded asterisk
06:43.09cuscoupgraded dahdi
06:43.10cuscoover time
06:43.11ChannelZbut I don't know enough about the hardcore internals of DAHDI to say
06:43.15cuscolibpri and spandsp
06:43.24cuscoonly this time
06:43.33WIMPyBut look at your dmesg. The SPAN x: xxxx sync source only appear for the first 4 spans. That dooes not look right.
06:43.36cuscoI finally moved debian to squeeze
06:43.46cusco:(
06:44.22cuscoif I coment out span1..4 conf, it will work
06:44.24cuscoI think
06:44.26cuscolet me try
06:44.29cuscorunning short on time
06:44.42ChannelZSeems like DAHDI as a whole only has one clock
06:45.14WIMPyThat would mean you can't use multiple cards without timing cable.
06:45.31ChannelZbut he actually needs the timing to be independent since it's from different sources.
06:45.48WIMPyThat would be dahdi--.
06:45.53WIMPyyes
06:45.56ChannelZUnless he can get N-1 of his telcos to sync to him instead
06:46.09WIMPyNo go.
06:46.12cuscogah now I can't be sure
06:46.16cuscoonly 1 cable there
06:46.20cuscoand I can't detect it
06:46.25cuscoits still red
06:46.39cuscobut I could swear that it happened yesterday
06:46.40cuscoor the day before
06:46.50WIMPyWhat?
06:47.14cusconot configuring spans 1..4
06:47.26cuscoand remaining (from 2nd card) work
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06:47.41WIMPyOk, that might make sense.
06:47.58cuscobut I can't be sure now
06:48.04cuscoWill leave it again for tomorrow
06:48.09cuscoservice is going to open in notime
06:48.54ChannelZI take it it's not saturated
06:48.57WIMPyMaybe the line was shut down because it's been in alarm?
06:49.06ChannelZWIMPy: I have to wonder
06:49.33cuscothe one with the cabel, perhaps
06:49.38ChannelZIt really seems like the other end is dead or misconfigured or otherwise not cooperating.  The card seems unfried
06:49.39cuscoI still have 2 cables disconected
06:49.50cuscothat should be on 1st span
06:49.50WIMPyHere they tend to disable lines after a certain amount of alarms/time.
06:49.52cuscothey are on
06:49.56cuscotelco is calling me everyday
06:49.57cuscolol
06:50.09cuscoI have to connect those 2
06:51.09WIMPywonders if the Digium cards would support multiple timing sources at all.
06:51.19WIMPyThe hardware that is.
06:52.19cuscohttp://paste.debian.net/124924/
06:52.23cuscoagain
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06:52.31Polysicshello
06:52.56cuscohi
06:53.02ChannelZIs "IRQ 16/wct4xxp: IRQF_DISABLED is not guaranteed on shared IRQs" normal?
06:53.10Polysicshow would one go about implementing a "credits" system so that a caller has a fixed amount of them, they get removed minute by minute, and finally the call gets dropped if they run out?
06:53.29Polysicspossibly with a "your credit is running out" message, but that is easy given the rest works
06:53.40Polysicson asterisk 1.8, i have Adhearsion available
06:53.46WIMPycusco: Again only 4 timing sources.
06:53.52ChannelZDial has time limits you can set.  If a credit is a minute, just feed it the number of credits they have
06:54.29ChannelZYou ust need to write the back-end to maintain the credit balance, subtracting minutes used after a call, letting them buy more, whatever
06:54.52PolysicsChannelZ, i know how to d oall that, but the problem is i need to drop calls when credits end
06:54.57Polysicsnot let them finish the call
06:55.11WIMPyAnd yes, the shared IRQ is definitely not ideal.
06:55.24cuscoPolysics: every time you place a new Dial() you specify the seconts time out flag
06:55.25ChannelZI told you, you do a lookup on their 'account' and feed Dial the number of minutes they have.
06:55.32cuscoseconds = credit
06:56.02ChannelZwhether it's 5 seconds or 5000, it's whatever their balance happens to be at the time they place the call.
06:56.10Polysicsthat could work. if the credit does not equal minutes, i can just work out the equivalent
06:56.20Polysicsabout the "your credit is running out" item?
06:56.29WIMPyThe interesting bit starts if you let them place more than one call at a time.
06:56.35ChannelZPolysics: core show application dial
06:56.37ChannelZall is revealed
06:56.40cuscodial too has a flag for when its timing out, I'm sure I read it
06:56.53cuscolike 15 secs before hangup it plays a warning
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06:57.13PolysicsWIMPy, no, they cannot place more than one call at a time
06:57.32WIMPyThat's fairly easy then.
06:57.44ChannelZMusicOnHold needs a flag so people can hit * or # or something to skip to the next song if what they're listening to is crap
06:57.58Polysicssince this thing has a user interface, that shows the credits going down, i guess i could do THAT by "pretend" decreasing the credits, then finalizing them when the call is over
06:58.11ChannelZAren't you listening?
06:58.18ChannelZLOOK AT THE L FLAG OF DIAL
06:58.19WIMPyChannelZ: Isn't that set by the mode?
06:58.25ChannelZIt'll annoy them if you want it to
06:58.34ChannelZhmm
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06:59.39ChannelZWIMPy: not sure I know what you mean.  There's no flags for that sort of thing in the app or the config that I am aware of
06:59.57ChannelZIt would of course make no sense if it was a live stream or whatever, but in files mode..
07:00.05WIMPyChannelZ: I think I've seen something.
07:00.34ChannelZI need to change mine, it's all illegal anyway
07:00.46ChannelZNobody gets put on hold that often though
07:03.11ChannelZneed to spend some quality time on SoundCloud trolling
07:04.19WIMPyI have no idea, what I was thinking of. But I have no idea where that option might exist, if noch for musiconhold.
07:04.26WIMPyif not
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07:05.14WIMPyAh. Got it.
07:05.26WIMPySet digit= in musiconhold.conf.
07:05.57WIMPyIt's for classes, but maybe that's useful.
07:06.00ChannelZhmmm I'd never seen that.
07:06.20ChannelZThat'd work more for like MOH "stations"
07:06.41ChannelZlike one class of classical, one of pop, that sort of thing.
07:06.45WIMPyYes.
07:06.48ChannelZNot just "this song sucks, play the next one" :)
07:07.12WIMPyBut maybe it works if you just skip to the same class that uses sort=random?
07:07.31WIMPySounds worth a try.
07:08.02ChannelZhmm worth a try
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07:09.28ChannelZInteresting.  The console tells me 'stopped MOH..' and then 'started MOH...' each time I press, but the song never actually stops or changes, it keeps playing through like I'm doing nothing
07:10.27WIMPyBad luck then.
07:10.50WIMPyBut that might be easy to change.
07:14.03ChannelZit's a little anecdotal, for another day.  Was just testing my MOH after restoring the files (HD died a few weeks ago) and I forgot what some of it even was.
07:14.26ChannelZanyway thanks for the kick, I wasn't even aware of the 'digit' option
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07:17.01ChannelZHmm.  I didn't know "Skullfuck" was a music classification.
07:18.52ChannelZAlright, I should be in bed.  Have fun y'all
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07:26.06Polysicshmmm
07:26.21Polysicsyour dial-with-credits idea is great, but i have a problem
07:26.28Polysicscredits are not per user, but per group
07:26.42Polysicsone or more users could be calling at the same time, all drawing credits from the same pool
07:26.56Polysicsso i can't just ask for a timeout, since the timeout is not known
07:28.46WIMPyThe interesting bit starts if you let them place more than one call at a time.
07:29.02WIMPyThat means doing some fancy stuff via AMI.
07:30.34irrootWimpy fancy stuff with AMI leads to the nuthouse :P
07:31.12WIMPyDon't be a wimp :-)
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07:41.35irroothehe im nuts
07:41.39irrootguess why
07:41.54PolysicsWIMPy, what could be a course of action? i will probably use Adhearsion for that part
07:42.03Polysicsthe whole app logic is in AHN already
07:42.34Polysicsbefore i said one call at a time but was only thinking of the single user - the group as a whole can and will do many calls
07:45.41Polysicsi basically would need a kind of timer
07:45.56Polysicsand i hoped to avoid having to build a separate daemon + RPC calls
07:54.23Polysicsany ideas, please? asterisk has nothing like a "call timer" thing? or a periodic event?
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08:08.46Dovidanyone from France here ?
08:11.17tuxx-trottoire
08:11.46tuxx-ehmmm
08:11.47tuxx-garage?
08:11.52tuxx-maybe emmmm
08:11.53tuxx-baquette
08:11.59tuxx-thats about all i know xD
08:12.08Dovidlol. i need some one in France
08:12.20tuxx-hey, just trying to help ;-)
08:14.11Polysicsthere is no way to have Asterisk emit a periodic event during a call?
08:15.23Polysicsis there a list of all the AMI events somewhere?
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08:16.53tuxx-Polysics: http://www.voip-info.org/wiki/view/Asterisk+manager+API
08:17.02tuxx-check the 'Manager Actions'
08:17.08Polysicsnot actions, events
08:17.12Polysicsbut thanks
08:17.22tuxx-ah
08:17.36tuxx-well, time for a sammich
08:17.36Polysicsi need to somehow get a periodic event out to remove credits fro ma shared pool
08:17.37Polysicssort of a calling card app
08:18.12irrootPolysics what we doo is use L option in dial within a AGI script based on database quereies
08:18.59Polysicsirroot, problem is, i cannot know the max duration in advance
08:19.06irrooti take the avail pool of credit divide by sim use and alocate that ammount as in use then "refund" the ammount left after the call
08:19.07Polysicsbecause the credit pool is shared between a group
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08:20.39irrootPolysics take the alloted ammount divide it by rate to work out the limit
08:22.04irrootthis is realtime the downside is you allocate a block of credit for each call that realisticly is not always used up
08:22.17Polysicsso if i hve 5 users in the group, i give total/5 to each maximum?
08:24.10irrootyip
08:24.21irrootor say they allowed 5 calls each
08:24.27irrootthen total/25
08:25.22irrooti use a "inuse" table where i insert a call and its length into a table and use that to determine num inuse
08:25.40irrootobviously with the credit allocated to them
08:26.15irrootselect sum(...),sum(credit) from inuse where XXXX = XXXX group by ....;
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08:38.26Polysicsand if a user needs a call to go longer than the allotted time, while having credit for it, and thus he in theory CAN have a longer call?
08:38.47Polysicssay i have 100 credits and 5 users, but only one is calling - will he be dropped at 20 anywya?
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09:21.29teloniuszhi guys. I'm playing with extensions.conf manually. I'd like to have an extension which just waits indefinitely while the user gets RING signal. How to do it with dialplan functions?
09:25.00gg0hi, (another one) how to set a different language just for a specific trunk?
09:25.02teloniuszoh, I see. Ringing() then Wait() will do the trick
09:27.10dymteloniusz: yupp
09:27.41dymgg0: well - in the trunk config
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09:29.50gg0I've tried to add language=xx everywhere. I managed to change it just globally in sip.conf
09:30.06Lantiziais there anyway I can view what password a device is attempting to use? (but failing)
09:42.42teloniuszgg0: Set(LANGUAGE()=xx) in the dialplan for specific trunk
09:43.30teloniuszgg0: or better Set(CHANNEL(language)=xx)
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10:24.57volker-is someone here experienced with sip-tls?
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12:35.23eduzimrsanyonw kwnows, this message appears at cli : "  == Connect attempt from '127.0.0.1' unable to authenticate"
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12:40.04leifmadsenmeans something is attempting to authenticate from localhost that is being rejected
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13:09.52eduzimrs@leifmadsen ok, but trying to auth in a sip peer or * manager?
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13:10.59prash10xhi
13:11.53prash10xpls help, i have to configure personal pbxmate
13:12.20prash10xit is not registering with sip server
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14:00.01E-bolaIsnt there anyway to list configured call/pickupgroups in the asterisk console?
14:00.11E-bolaOr any other way i can get an overview without going through all my sip.conf files
14:02.48*** join/#asterisk darkdrgn2k (~darkdrgn2@199.243.221.14)
14:02.51darkdrgn2kHey all,
14:03.29darkdrgn2kfrom time to time im getting calls come in with the remote perosn not bein able to hear the person speaking.  Any idea what could cause this
14:03.34*** join/#asterisk coppice (~chatzilla@116.92.29.157)
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14:07.22*** join/#asterisk joesuffceren (~chatzilla@ip68-104-167-226.ph.ph.cox.net)
14:07.25joesuffcerenAnyone know anything about Cisco 79X0 devices? I have a whole fleet of 7940s and am considering purchasing some 7940Gs. So far as I can tell, the only difference between the two is that the global (7940G) edition uses icons instead of words for the softkeys. I am wondering if I can just flash the 7940G with the 7940 firmware so that they show the words instead of icons.
14:08.26jayteedarkdrgn2k, make sure your rtp ports set in rtp.conf match what you have open for rtp on your firewall
14:08.46darkdrgn2kproblem seems to go away when we reboot our router..
14:08.57darkdrgn2kand its only at that one branch, that would point to a network issue
14:09.02darkdrgn2kbut i have no idea where to even look
14:10.16_Corey_joesuffceren: It has to do with the physical buttons on the phone, not the firmware
14:10.25_Corey_firmware is the same...
14:10.58_Corey_They come with stickers to place around the buttons if you really need the words... most people find the icons enough
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14:20.39joesuffceren_Corey_: so the screen above the soft buttons would still display text? I couldn't possibly care less about that. haha. Totally misunderstood. Thanks for the clarification!
14:20.56*** join/#asterisk Linuturk (~linuturk@unaffiliated/linuturk)
14:21.56_Corey_Yeah, it's just the four buttons around the (?) button we're talking about on the lower right.... :)  No problem
14:23.56p3nguinG means Global, as in pictures on the face of the phone instead of words.
14:24.15p3nguinBut then you add a sticker to the face overlaying the keys so you have words too.
14:33.32*** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel)
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14:40.28Qwellpsst
14:40.29Qwellhttp://store.digium.com/productview.php?product_code=810-00038
14:41.14jacc0@joesuffceren: I wouldn't buy any cisco or linksys phone right now
14:41.26dvdeveli have a question that i hope somebody can answer - asterisk 1.6.2.19, if i have a sip entry (say for exten 9705551234) and then i get a call from another asterisk box claiming to be from that number, the call is rejected.  can somebody explain why that is, and how to stop that behavior?
14:42.19jacc0I've reported some big security hole to cisco yesterday - there is no fix available
14:42.45*** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net)
14:43.05jacc0I'm not sure the cisco models you are referring to are also effected
14:43.17Qwelljacc0: oh?
14:43.29QwellIs it already publicly known?
14:43.39jacc0Uhm, well, ......
14:43.49jacc0you can find some info about it some where
14:44.15*** join/#asterisk DigitalFlux (~quassel@unaffiliated/digitalflux)
14:44.17jacc0but it is not yet publicly reported by cisco
14:44.18DigitalFluxHi Guys
14:44.29DigitalFluxI need some caller to input a phone number
14:44.35jacc0for now I will not share any technical details
14:44.42DigitalFluxand i should catch that in some var and Dial() it in my extensions ..
14:44.47Qwelljacc0: that's fine - was just curious if it was known
14:44.59DigitalFluxWhat would be an example for that so that i can put it in my extensions.conf ?
14:45.01QwellOnce it is released, I'd be interested in hearing about it.
14:45.04jayteeI've had Beans and Bytes coffee....it was pretty tasty stuff.
14:45.10Qwelljaytee: Go buy some :p
14:46.08jayteeQwell, I'd rather buy some green Kona beans and roast my own. Haven't roasted in almost a year.
14:46.22jacc0what I can say is Smithts (not sure what his name is exacly) that hangs around here normaly can confirm the big security hole
14:47.35jacc0cisco/linksys phones can be remotely triggerd to ; set up a call, clear call history, update firmwar (about everything you can do if you have the phone on your desk and know the passordw
14:47.47Qwellthis isn't the telnet thing is it?
14:48.06jacc0I will not answer to that
14:48.12Qwellso, yes :)
14:48.27Qwellif so, it's quite well-known already
14:48.43joesuffcerenjacc0: thanks for the heads up. I have 75 of them in production, though, so one more for a new user isn't going to make or break me. If someone can get telnet access to these phones, they're already far enough inside my netowrk that I'm screwed anyway. :-)
14:49.50DigitalFluxGuys, how can i capture the numbers from dtmf in a variable in Asterisk ?
14:50.00jacc0in some cases a malisous caller id is enough to do the trick
14:50.00DigitalFluxAny Asterisk app for that ?
14:50.27jacc0read()
14:50.33DigitalFluxchecking ..
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14:50.48DigitalFluxcool Thanks jacc0
14:51.16jacc0if the phone still has the XSS hole discribed in : uys, how can i capture the numbers from dtmf in a variable in Aster
14:51.26jacc0sorry:  http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCtr27226
14:51.46jacc0sorry again
14:52.10jacc0this document : http://www.owasp.org/images/6/6a/OWASPBeNeLux2010-State-VoipHacking.pdf
14:53.27jacc0then the new hole can be triggerd by sending a malicious caller id
14:54.01jacc0and yes; http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCtr27226 is the security report from cisco about it
14:54.03jacc0:p
14:54.05Naikrovekso you weren't going to talk about it but here's the document from last year describing it
14:54.09jacc0but it's not public
14:54.31Naikrovekit's on the internet; it's public
14:54.31jacc0combining the bug from last year with this one makes it worse
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14:55.35jacc0XSS can be used to attack linebase protocols
14:55.40jacc0*line-based
14:55.51jacc0as the new bug is in a line based protocol
14:56.00jacc0combing the 2 makes it worse
14:56.59jacc0you don't even need derect access to the phones ip
14:57.12jacc0all you need 2 do is set a malisous caller id
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14:57.40Naikrovekcombining* direct* malicious*
14:57.48Naikroveksorry
14:57.52jacc0I'm a non native
14:57.59Naikrovekfair enough
14:58.27jacc0@Naikrovek: http://tools.cisco.com/Support/BugToolKit/search/getBugDetails.do?method=fetchBugDetails&bugId=CSCtr27226 is on the internet. so you would say it's public?
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14:59.32Naikrovekwell let me log in and we'll see
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15:01.00Naikrovekmy account won't let me access it, but that doesn't mean it's not public
15:01.12Qwelljust means Cisco doesn't like you
15:01.20jacc0lol
15:01.26Naikroveki don't have the right type of login
15:01.35Qwelldoes
15:01.36Naikrovekwhat it means is that i've not whored myself out to them yet
15:01.39Naikrovekgood for you
15:01.40garymcHi Guys, for Asterisk ports, do I open UDP 5060 and UDP 10000-20000 in my router firewall to access my asterisk box remotley
15:01.44QwellI actually don't. ;(
15:01.52Qwellgarymc: yes..
15:01.59Qwellfor SIP
15:02.10garymcis that all I need to make calls?
15:02.18Qwellsure
15:02.20jacc0yes
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15:05.45jacc0last word from cisco about the security hole : "Our Dev team had fixed the enclosed issue and will be in the upcoming release."
15:06.28p3nguinhad fixed... as opposed to fixed.
15:06.39p3nguinThat means it used to be fixed.
15:08.30a1fais there a website that tells you where number terminates?
15:08.39a1fai lost my bookmark, i had one
15:08.50a1fait listed the company that owned the number, down to the hub
15:09.42Naikrovekthe answer to your question is "yes" but I don't know the answer to your next question, which is probably "what is the URL"
15:10.01darkdrgn2kMy guess is seeing how you HAD the website at one point it does exist?
15:10.19a1fayes please
15:10.26a1fathere are sites out there
15:10.30a1fakind of like domain whois
15:10.45_Corey_You looking for the NANPA database of NPA NXX info?
15:11.25a1faaybe
15:11.28a1famaybe
15:11.37a1fai think voip.ms had that feature
15:11.42a1fait will tell you who the number belongs to
15:11.42*** join/#asterisk ickmund (~ickmund@cli-5b7e85f2.bcn.adamo.es)
15:11.53darkdrgn2ki havent seen it on their site.. but what do i know
15:12.19*** part/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)
15:12.56a1fadoes anyone have a voip.ms account?
15:13.09darkdrgn2ki do
15:13.13_Corey_Well, this is where they probably get it from: http://www.nanpa.com/reports/reports_cocodes_assign.html
15:13.26_Corey_We have it loaded in MySQL and have an internal portal to run queries
15:13.41_Corey_so I can't point you to an easy site, but the data is there
15:14.05jayteehttp://www.area-codes.com/exchange/exchange.asp?npa=513&nxx=322  <-this is what I use for NPA-NXX lookups
15:14.19darkdrgn2k:( aww no canadian version
15:14.24a1fadarkdrgn2k: can you check 877-502-6442
15:14.30_Corey_Yeah, that looks like the same data
15:14.36a1faCheck Availibality on the porting
15:14.39darkdrgn2kthast an 800 number..
15:14.41a1fait gives you the owner of the number
15:14.42darkdrgn2kthye never terminate..
15:15.34a1fai wonder how you trace it back to the carrier
15:15.41darkdrgn2khttp://whocallsme.com/Phone-Number.aspx/8775026442
15:15.48a1fayes, i've seen that
15:16.03a1fahowever, i'd like to see who their carrier is
15:16.49darkdrgn2ki dont know if you can with 8xx numbers
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15:17.18*** join/#asterisk drmessano (~nonya@pdpc/supporter/active/drmessano)
15:17.32a1fayou should be able to
15:17.36a1fai am about to call verizon
15:17.52darkdrgn2khaha good luck
15:18.22p3nguin877..... is an 800 number?  Looks like 877 number to me.
15:18.33darkdrgn2ksoryr
15:18.33darkdrgn2k8xx
15:18.40a1fawell, they are harassing my wife on her cellphone
15:18.48a1fanext thing is android call firewall
15:18.57a1fabut i'd like to file a complaint with verizon
15:19.05p3nguinI have a VoIP.ms account, but I don't know how that's going to help you.
15:19.09a1fathey should not be trunking that number
15:19.15a1fap3nguin: if you go check number portability
15:19.20a1fait will tell you it cant port the number
15:19.30darkdrgn2kmight be a pw0wned asterisk box
15:19.30a1faand it will give you carrier who owns the number to call them
15:19.43a1fadarkdrgn2k: sure, but it needs to stop, dont you think?
15:20.21a1fait does no good to the community thats for sure
15:20.27_Corey_alfa: What do you mean by "they should not be trunking that number" ... I'm curious
15:20.34darkdrgn2ka1fa: for number portability you need to KNOW your own provider
15:20.44a1fa_Corey_: they should not be routing it on their network
15:20.45*** join/#asterisk webguynow (~webguynow@c-24-1-222-204.hsd1.il.comcast.net)
15:20.49p3nguinchortles
15:20.49a1fadarkdrgn2k: not through voip.ms
15:20.57a1fait will give you the carrier
15:21.00_Corey_alfa: why is that?
15:21.13darkdrgn2kthen voip.ms will not give you the varrier
15:21.18a1fa_Corey_: it's classified as "spam"
15:21.34_Corey_uh yeah, that doesn't matter
15:21.54a1fai know it does not
15:22.03a1fabut if its a scam, dont you think it should be disconnected
15:22.18a1faFCC ruled on this once already
15:22.25darkdrgn2kwhat are you the us government now? with DID siezeurs :-P hahaa
15:22.34a1faha ha ha
15:22.48a1fawhy is that foreign nationals, outside of US can terminate US DIDs?
15:23.07_Corey_I can present that number if I wanted to, as could anyone...
15:23.07a1fabusinesses too
15:23.36darkdrgn2kyeh cids arent written in stone
15:23.56a1faits just bad design, altoughether
15:24.03a1faaltogether*
15:25.43*** join/#asterisk brdude (~brdude@c-24-5-194-184.hsd1.ca.comcast.net)
15:26.02a1fap3nguin: where you able to check for me, via voip.ms?
15:26.15a1faand has anyone seen TK Defender lately?
15:26.53darkdrgn2ka1fa: where on voip.ms. i dont see it anywhere
15:27.03a1fayou need to go to number portability
15:27.07a1falogin to your portal
15:27.35a1fahttp://wiki.voip.ms/article/Porting_a_Number
15:28.06darkdrgn2kyes
15:28.07darkdrgn2ki did that
15:28.11darkdrgn2kbut it ask YOUR who the carrier is..
15:28.24a1fatry something random
15:28.26p3nguinI have not checked it, and TK is still around but not on this channel.
15:28.38a1fap3nguin: how come?
15:28.42darkdrgn2k"Service Provider Information" means YOU provide the service providfer info
15:28.52a1fatype in "Broadvoice"
15:28.54a1fahehe
15:29.06p3nguinWhen I try to login, it tells me I'm trying from a forbidden IP address.
15:29.41darkdrgn2kp3nguin, : wouldnt work, voip.ms doesnt tell you the serivce provider anyway..
15:29.49a1fait used to
15:29.50p3nguinI figured it wouldn't.
15:29.59darkdrgn2kusualy ports require you to PROOVE you own the #
15:30.03a1fai remember trying to port customers number, and it would not do it
15:30.10*** join/#asterisk defswork (~andy@mx2.3gcomms.co.uk)
15:30.17a1fadarkdrgn2k: aka thats why you click check availability
15:30.52p3nguinoutside of US can terminate US DIDs?   <--- what does this even mean?
15:30.59p3nguinDIDs don't terminate.
15:31.14a1fasip trunks
15:31.19p3nguinno such thing
15:31.27a1fa.. ok sherlock
15:31.33p3nguinYes, Watson?
15:31.40a1fadon't beat around the bush
15:31.49darkdrgn2ka1fa,: Your absolulty right.. but TOLL FREE numbers DONT HAVE check availabllity!
15:31.55a1fawhat's D-Fender's nick?
15:32.01a1fadarkdrgn2k: thanks for checking
15:32.02defsworkI was getting repeatable deadlocks in queue so I took all my queues out thinking that would solve it but have had another deadlock today.  Is 1.8.5 stable ? Am I the only one having these kind of problems ?
15:32.07p3nguinTermination is for OUTbound calls.  DIDs are INbound.
15:32.15jayteea1fa, go to the #freepbx channel if you want to chat with TK
15:32.16p3nguin[TK]D-Fender
15:32.23a1fap3nguin: hanks for setting me straight
15:32.39drmessanoorigination is inbound
15:32.42drmessanoerrr
15:33.16p3nguinWhen a call comes into your DID from the PSTN, that's origination (not termination).
15:33.34a1fap3nguin: thanks
15:34.20p3nguinIf toll-free numbers don't have the "check availability" thing, does that mean they don't port them?
15:34.27darkdrgn2kwhat if it originates from somethign other thebn a PSTN like skype :)
15:34.37darkdrgn2kp3nguin: no 800 numbers can ALWAYS be ported because of their nature
15:34.59darkdrgn2kp3nguin: in the PSTN world an 800 number MUST be attached to a normal DID
15:35.00p3nguinThey just can't check it because there is no "declined" choice?
15:35.17p3nguinWhat do you mean attached to a normal DID?
15:35.34darkdrgn2kyou cannot have a PAIR with only an 800 numbe ron it
15:35.40p3nguinI have a toll-free DID, and that's all that I have.  It seems normal to me.
15:35.49darkdrgn2kis it a voip line?
15:35.53p3nguinyes
15:35.57darkdrgn2kas i said
15:36.07darkdrgn2kvoip broke that requirement
15:36.16darkdrgn2kin the OLD PSTN world, you could not have a dial tone with only an 800 numbe ron it
15:36.17p3nguinI didn't see you say that.
15:36.40darkdrgn2ksorry i guess i thought it :(
15:37.17darkdrgn2kbut yeh you cant get a pair come into your office with only an 800 number on it.. the dmark is always a local line...
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15:37.20p3nguinSo if I go back to copper, I will have to have a "regular" number before I can port my toll-free over?
15:37.27darkdrgn2kthast why 800 numbers are so portable
15:37.27darkdrgn2kyep
15:37.47*** join/#asterisk mhaddog (~mhaddog@z65-50-116-231.ips.direcpath.com)
15:37.48p3nguinJust like if I have only DSL... there is still a phone number on the circuit even if I don't have a phone.
15:38.07elband it confuses the @!$!@$! out of support drones
15:38.16a1fa+1 elb
15:38.17a1fa+1
15:38.18elb"I show the number on your account as 111-222-3333"
15:38.32a1fanumbers are cheap
15:38.37darkdrgn2kworst part is you CID wont show the 800 numers either :)
15:38.38a1falets toss them around like salad
15:38.41elb"no, the number on my account is <some real number>, you can't call me at that number, it has no POTS line"
15:38.53elb"but I have to call you at the contact number"
15:38.55elb"@#$@#$@#$"
15:39.21p3nguinHow do you ever get them to move past that and get to the actual problem?
15:39.28a1fayou dont
15:39.33elbp3nguin: call back and get a different rep
15:39.49elbthe best part is when they ask you for the number, and you don't know it, because they've never given it to you
15:39.50defsworkis anyone else aware of deadlocking problems in 1.8.5 ?
15:40.12elbI gave all those problems up, though ... now I have VDSL (AT&T U-Verse) without television service, which is always the sticking point
15:40.14p3nguinIs that when you call back and get someone else to give you the number?
15:40.22elbthey swear they don't sell U-Verse without TV service when I call
15:40.24elband won't help me
15:40.30elbtelcos are AWESOME
15:40.34darkdrgn2kthe best part is whne they ask you for the phone number, then the accout number, which is the same, and they insist its not
15:40.44darkdrgn2kim like IM LOOKING AT THE BILL.. IT SAYS ACCOUNT NUMBER RIGHT HTERE!!!
15:41.29darkdrgn2ki have VDSL:( 25 megabit connection with a 75 gig cap.. WTF!
15:41.39darkdrgn2kLets do the math. thast what 4 ours of downloading
15:41.41p3nguinI don't have an AT&T bill handy, but I thought the phone number is the same as the account number.
15:41.43a1faATT UVERSE?
15:41.55a1falol 75 gig cap?
15:41.56darkdrgn2kp3nguin: it is with BELL..
15:41.58darkdrgn2kyep
15:42.04a1fai cant even fart with 75 gigs
15:42.12darkdrgn2kthey they are like We can upgrade you to the 50 megabit ...
15:42.14a1fai'd be running out daily ;)
15:42.16darkdrgn2kim like whats the cap on that
15:42.17darkdrgn2k100...
15:42.21a1faROLF
15:42.34darkdrgn2kim like " send that person back to school, he needs to learn fractions"
15:42.48darkdrgn2kthe person on the phone said i can buy INSTAUCACE at 5 bucks a pop x 3 to boos me up to 350..
15:42.53darkdrgn2kthatst 24 hours of downloading i think
15:42.53p3nguinI'm sure that's not the only area which needs improvement.
15:43.50a1fanah
15:43.59a1fayou'll be far fetched to use 75 gigs
15:44.06darkdrgn2kYeh
15:44.09a1faunless you are on torrents all day
15:44.12a1fa24/7
15:44.15darkdrgn2kumm
15:44.16darkdrgn2kdude
15:44.18a1faand even then..
15:44.23darkdrgn2ki upload 4 gigs ISOs to the datacenter..
15:44.29darkdrgn2kand stuff
15:44.35darkdrgn2kand its 75 gigs UP  + DOWN not UP OR DOWN
15:44.37a1fahave your work pay for the fees associated with overages
15:44.51darkdrgn2kso download the ISO from eopen... 4 gigs.. upload it to the datacenter .. 4 gigs..
15:45.07a1fawhy do you upload isos?
15:45.17darkdrgn2kESXi..
15:45.22a1fabut why
15:45.31darkdrgn2kcuase i have no GUIs at the datacenter
15:45.36a1fawget?
15:45.37darkdrgn2kand eopen doesnt work with lynx of wget
15:45.49a1faso put a jump box over there
15:45.56darkdrgn2kno room:(
15:46.02darkdrgn2kanywa moot point
15:46.03a1famake a virtual jump box :) yo
15:46.12darkdrgn2kdownload 2 bluerays.... and BOOM... there goes 60 gigs
15:47.02darkdrgn2kin december before the new rules cam into affect (last year it was 75 gigs. buck a gig after that to a max of 30 bucks)  i tried to see what how much i could pull in a month.. just a proof of concept.
15:47.12darkdrgn2ki ended up pulling like 1.7 tb.... and ran out of hd space
15:47.16darkdrgn2kthen i did the match
15:47.49darkdrgn2k1700 gigs = 75 gigs free + 250 gigs "insured"  leaves 1375 gigs x 1 buck a gig = $1,375
15:48.00darkdrgn2ki cant WAIT for a virus to infect one of theses poor suckers and rape their bandwith :)
15:48.10a1fa;)
15:48.13p3nguinI have a Windows XP ESXi machine for that sort of thing.  It's kind of a bother, but it works.  Since I don't use Windows on the desktop, and the console uses Windows... rdesktop to the XP vm and control all the other vms.
15:48.24a1fasounds like I'll be able to make a living :)
15:48.43darkdrgn2klol
15:49.11a1faat $79.99 for diagnostics, thats bread and butter.. free money
15:49.19a1fa$175 to fix simple things
15:49.42darkdrgn2klol
15:49.44darkdrgn2kstill
15:49.52darkdrgn2kwhy have vdsl when you have a 75 gig cap
15:50.00darkdrgn2kim terrified to do stuff at home now
15:50.56a1facancel
15:50.58a1fago buy cable
15:51.09darkdrgn2ki'd miss the 8megabit upload!
15:51.21a1fayou need to find alternative means dude
15:51.25elbcable is stupid expensive here
15:51.31elbof course, my vdsl is also slow
15:51.44a1fa$49.99 here with 12Mbit/3Mbit d/u
15:51.48elbbut ... I can't get cable for less than about $60/mo
15:51.56coppicemy kids can burn through 75G of youtube in no time at all
15:52.14darkdrgn2ki love it
15:52.23darkdrgn2kevery one is doing the "TO THE CLOUD" krap..
15:52.31darkdrgn2kand isps are sayinf PAY US FOR THE CLOUD
15:53.01elbthe cloud is a lie
15:53.23darkdrgn2kmeh i loke the cloud
15:53.29darkdrgn2kat least the ons hosting my files at home :)
15:53.34darkdrgn2kanother reason i LOVE vdsl
15:53.35coppicethe cloud is very honest - its a bit wet
15:55.40p3nguinMiss the 8 Mb upload?  Why?  I get damn near that much on cable.
15:56.38darkdrgn2kwe dont
15:56.44darkdrgn2kours is is like 4 max
15:57.56p3nguinWe also have 100 Mbit download speeds on cable.
15:58.06a1faso why did d-fender get shit canned?
15:58.39p3nguinI don't know about shit-canned, but he got a +q for saying a lot of unnice things.
15:59.14p3nguinEventually he just quit coming here reading what others said.
15:59.56darkdrgn2kyeh... but hes fun to watch some times :-P
15:59.59p3nguinI don't really think anything he said was uncalled for.  But I've been told I'm not nice at times as well.
16:00.00a1fasuprising, he was always nice?
16:00.12a1faand always helpful
16:00.20jayteep3nguin, you're always mean and nasty :-)
16:00.21beekHis banishment has been a major loss to this channel.
16:00.31coppicehe made the fatal mistake of being helpful and honest. only do a maximum of one at a time
16:00.48darkdrgn2kits funny it hoguth he'd leave freepbx first... he hats the crazuy dialplans
16:00.48a1fap3nguin: you just need to be more precise, and stop beating around the bush. if somebody is not correct, just say it out right, and correct them
16:00.48jayteep3nguin, while being extremely helpful at the same time
16:00.52p3nguinRegardless of his style, he did help a lot of people here.
16:00.53a1fano need to point fingers and laugh
16:01.12a1fabeek: i agree.. that's what suprised me
16:01.17darkdrgn2kumm guys
16:01.20darkdrgn2kyou know this is IRC right?
16:01.26a1faand?
16:01.26darkdrgn2kpointing an lafing is like a GIVEN!
16:01.39a1fanot on freenode, it has never been like that
16:01.47a1faa lot of things have changed in last few years
16:01.49darkdrgn2kLMAO where have you  been?
16:02.05a1fadarkdrgn2k: i've been on here since the network has been started
16:02.06darkdrgn2kok maybe not POINT AN LAF but smartassing is the LAW
16:02.14p3nguinI guess if you all want him back, you could always start a petition and give it to russelb.
16:02.24a1fathe attitude has changed dramatically in the last few years back
16:02.36darkdrgn2kthe internet has changed..
16:02.38beekp3nguin: russellb now works for Red Hat
16:02.46a1fafreenode used to be a place to get away from assholes on other networks, and actually get help
16:02.51a1falooks like trolls migrated
16:02.54p3nguinHe doesn't manage this channel anymore?
16:03.07beekI wouldn't think so.
16:03.12darkdrgn2ka1fa: i always get my ass wooped.. but lear sometihngi n the process...
16:03.13p3nguinI did not know that.
16:03.14QwellHe's still an op.
16:03.24*** join/#asterisk philippel_mac (~p_lindhei@c-67-160-11-168.hsd1.wa.comcast.net)
16:03.43p3nguinHe's the one that apparently made the decision about fender.
16:03.45beekMakes more sense for a Digium employee to be deciding what happens on this channel.
16:04.00beekp3nguin: yes.  I saw the exchange.  'fender sent me the links to the logs.
16:04.07a1fadarkdrgn2k: rewind that few years back
16:04.13a1fayou could actually learn something here
16:04.23Kobazoh, is that why fender is awol
16:04.30philippel_macquestion, anyone know if there is a counterpart to the IMPORT() function, or ImportVar application, e.g. the ability to set variables in another channel?
16:04.32a1fathere are few remaining channels left that are actually good
16:04.41a1faeverything else is a hit or miss
16:04.46philippel_macbasically something like MASTER_CHANNEL() but where the channel can be specified?
16:04.58p3nguina1fa: Society keeps deteriorating, so those of us who are intolerable get noticed more now.
16:04.59a1faever since LILO died, the network has been going down hill
16:05.01philippel_macin 1.8, or otherwise a patch or in trunk that could be back ported?
16:05.18a1falilo used to gline those trolls, pretty quick
16:05.21darkdrgn2ki been trollin since the days of multi collide bots and REAL net splits.... the ppl got more trollier (on avg) but the IQ has just increasesd. if you knwo what ot ask, what to take seriosuly and what NOT to do to feed the trolls its a great place
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16:05.35beekWell, 'fender may have been the rudest guy on the channel but if you did as he asked and provided the info that he asked he could have your problem solved very quickly.  He was extremely helpful to me.
16:06.04darkdrgn2kagreas with Beek
16:06.07a1fahe was never rude to me, and always helped
16:06.14a1fagoing back 5 years
16:06.20Kobazi always found fender amusing
16:06.23beekalfa: he just didn't suffer fools well.
16:06.24a1faor actually more than 6 years
16:06.26darkdrgn2ka1fa: he has no patients for NOOBs thast the problem
16:06.34a1fano body does
16:06.40p3nguinsome less than others.
16:06.45darkdrgn2ka1fa: yet every one of us had been one at one point
16:06.58QwellI was never a noob.
16:07.02QwellThat is a documented fact.
16:07.09a1fai was born with a 12 inch cock, but that's just me
16:07.10beekHe just followed the same rules on IRC as applied to mailing list:   DO YOUR HOMEWORK FIRST.
16:07.11darkdrgn2kbut as a noob i took his critisizm to heart and he helped me crack Nortel's SIP config for freepbx :)
16:07.11Qwellleifmadsen: wait, was it noob or newb?
16:07.19philippel_macQwell:  any idea on the above setting channel var in another channel?
16:07.35Qwellphilippel_mac: I don't.  Tilghman would be a good person to ask though.
16:07.37darkdrgn2kr/noob/nweb/
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16:09.50leifmadsenQwell: nub
16:09.57leifmadsenQwell: oh right, newb
16:15.11*** join/#asterisk BuenGenio (~BuenGenio@cm61-10-82-188.hkcable.com.hk)
16:16.39Kobazwho'se going to be the new russell?
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16:22.03serafieKobaz: right now three or four existing Digium people are divying up the work, including kpfleming and The_Boy_Wonder
16:22.53The_Boy_WonderKobaz: i'm looking at the timerfd issue now
16:23.09The_Boy_Wonderwhere you ever able to reproduce it consistently?
16:23.57The_Boy_Wonders/where/were
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16:40.25KobazThe_Boy_Wonder: yeah i can reproduce it pretty often with my unit testing
16:40.40KobazThe_Boy_Wonder: but i haven't put together a sample dialplan that does it... i have a lot of nuttyness going on
16:41.53The_Boy_WonderKobaz: if there is anyway you could narrow it down to something I can easily do here, that would be awesome for testing this.  I have a few ideas on what might be going on.  The problem is the turn around time required to tell if experimental code makes a difference or not
16:42.12The_Boy_Wonderif i can reproduce it here, i can get this done much quicker
16:42.47The_Boy_Wonderotherwise its a process of posting a patch, and waiting for feedback
16:43.20Kobazyeah
16:43.24Kobazi know it's brutal
16:43.40KobazI'm currently breaking everthing in my development, so it'll be a little bit
16:44.52*** join/#asterisk Defraz (~Defraz@63.226.95.152)
16:45.10The_Boy_WonderKobaz: alrighty
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17:02.18dvdevelwith asterisk 1.6.2.19, if i have a sip entry (say for exten 9705551234) and then i get a call from another asterisk box claiming to be from that number, the call is rejected, even if it's to another valid number in that box.  can somebody explain why that is, and how to stop that behavior?
17:02.31*** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital)
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17:08.07ChannelZSee the 'naming devices' section of the sample sip.conf
17:13.09dvdevelyes, that makes some sense.  still, why would it reject the call - it's a valid source of calls if it exists.
17:13.58dvdevelah, because the credentials don't jive
17:14.34dvdevelallow me to think on this.  thanks, ChannelZ
17:15.43ChannelZyes if it's matching the call to a peer when it shouldn't.  Using extensions/numbers as device names is probably not a good idea in your case
17:16.40dvdevelaye, i think i can work around it "easily enough"
17:18.34citywokdvdevel: it would be really annoying if random calls could be sent to all your 4 digit extension desk phones and drive all your people crazy
17:18.48citywokalso, how would asterisk know what context to send those calls to without a peer definition telling it?
17:19.16dvdevelagreed - i just wasn't "putting two and two together" that it wasn't matching the trunk entry but rather the extension
17:19.28ChannelZthe whole user/peer/friend thing and what happens is still clear as mud
17:19.31dvdeveli mean i've _only_ been using asterisk for about five years.
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17:23.24azv4I know this is OT, but I know there are some old phone system pros around here!  Any Panasonic Digital Hybrid phone pros remember if it is possible to connect to all phone's speakerphone in case of an emergency?
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17:49.49nnyis there a way to playback a tone in a meetme to only one side?
17:49.55nnyvs announce, etc
17:50.09*** join/#asterisk aidinb (~aidin@unaffiliated/aidinb)
17:50.45nnycan be pre the other person joining, I can use a hackish command to play something in that room before the join
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17:58.00eduzimrsanyonw kwnows, this message appears at cli : "  == Connect attempt from '127.0.0.1' unable to authenticate" its a sip or manager connection type?
17:58.01nnynm got it :D
17:59.18ChannelZeduzimrs: Looks like Manager
18:02.00*** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net)
18:02.45neurosysleifmadsen:  Like breaking from a queue, can a dialplan be made to break from music on hold to leave a VM if the caller is placed on hold and decides they no longer with to hold and leave a VM?
18:03.56leifmadsenneurosys: application map in features.conf
18:04.16neurosysleifmadsen:  Looking. Thanx :)
18:04.23*** join/#asterisk rjune (~rjune@75-150-213-153-Illinois.hfc.comcastbusiness.net)
18:05.55p3nguinI use a short queue timeout with a prompt to leave a voicemail if they want... or continue to hold, which drops the call back into the queue.
18:06.35rjune"called g0/#######" in the log indicates asterisk has picked up a line and dialed the number, correct?
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18:07.06p3nguinI guess your moh would have to play a message to the call on hold for the person to know he can press a key to leave a message.  Most people sitting on hold think they have only two choices: wait longer or hang up.
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18:22.49NephFLI have a server randomly rebooting with no errors showing in logs, dell t110, have run bios and raid updates...have switch power supply mb and cpu... have digium AEX800 to connect to incoming pots ...
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18:23.26NephFLrunning on CentOS 5.5 (freepbx distro)... and short of rebuilding...I'm at a loss
18:25.46rjuneNephFL, can you take it down temporarily?
18:26.25NephFLits live
18:26.34rjuneI understand, is it a 24/7 shop?
18:27.26NephFLno, 730 to 5 i think
18:27.36NephFLwhat do you have in mind?
18:29.44rjuneRAM test in specific
18:29.54rjunebad hardware will do what you're seeing
18:30.14rjuneRun memtest86 on it,
18:30.25rjuneInquisitor seems to be a decent hardware test suite in general
18:30.33rjuneJust be careful not to wipe the drive
18:34.47eduzimrsChannelZ should manager try to connect from localhost?
18:35.13eduzimrsChannelZ it never happend before
18:37.32neurosysp3nguin:  and how would you define the hold to go to a queue as opposed to the hold app?
18:38.36p3nguinYou don't.  That's not what queues are for.
18:39.25neurosysOk. I understand the queue break out...
18:39.52neurosysBut the customer wants "If they are place on hold after pickup, the ability to press # and leave a message and hangup".
18:40.05neurosysI think Ill just tell him not possible :P
18:40.30neurosysDang customers with their weird requests :P
18:44.33p3nguinI don't know if it's possible or not.  Dial()'s d option is similar, but it indicates that it is to be used while waiting on the call to be answered.  If the call is on hold, it has already been answered.
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19:08.20Gokee2Hello everyone, I have a Digium 410p PCI card with two FXS and two FXO ports.  After moving it to a new computer the fxs ports have stopped working.  However in asterisk the ports are all seen and appear to work, they are just dead when you plug a phone in.  Any idea's?
19:08.38*** join/#asterisk Beltechs (~Beltechs@cpe-76-175-74-169.socal.res.rr.com)
19:09.35JonathanRoseGokee2:  pastebin your chan_dahdi.conf and your dahdi/system.conf
19:09.52JonathanRoseThat'll help someone to take a look.
19:10.52JonathanRoseI'm pretty sure the card I'm using is similar.
19:10.57Beltechshello, Im running * 1.6, sip trunk, G729 codec, I'm having random calls with static. Including the initial ring produced by the pbx. Its most noticeable when dialing 800#'s any ideas would be appreciated.
19:11.13chazzamGokee2: is the power cord on the board connected?
19:12.01Gokee2chazzam, Hey, good question!  I had not though to ask that yet.
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19:16.31chazzamBeltechs: the SIP phone itself produces the ringing sound generally, not the PBX. if you get static there, then its probably the phone.
19:16.42ChannelZeduzimrs: well people or other apps connect to manager, and could have tried to do so to localhost.  You'd have to figure out what actually triggered it
19:17.01ChannelZPossibly just a guy doing 'netstat' on your box and seeing that port being listened to, and telnetting to it or something.
19:19.10a1faQwell
19:20.02Gokee2JonathanRose, Ok here is the chan_dahdi http://pastebin.com/dNasuQzF  dahdi-channels.conf http://pastebin.com/ngzQFeDZ and system.conf http://pastebin.com/zrXAKYNC  Let me know if you see anything wrong.  But I think chazzam may be right with his unplugged comment...  Got to contact the site again to find out though
19:21.12rjune"called g0/#######" in the log indicates asterisk has picked up a line and dialed the number, correct?
19:22.18*** join/#asterisk PoWeRKiLL (~powerkill@bzq-79-183-200-176.red.bezeqint.net)
19:22.26PoWeRKiLLhi
19:22.31PoWeRKiLLcoppice are you there ?
19:23.01chazzamGokee2: that system.conf only configure ports 1-3, while asterisk is configured for 1-4
19:23.41chazzamasterisk should fail to load chan_dahdi
19:24.00coppicePoWeRKiLL: possibly
19:24.16PoWeRKiLLcoppice I got a fax problem
19:24.21PoWeRKiLLI have a  WARNING T.30 Page did not end cleanly
19:24.44PoWeRKiLLI made a pcap of all the call
19:25.01coppiceT.38?
19:25.10PoWeRKiLLYes
19:25.40coppicecould be a broken implementation at the far end. does he page look OK?
19:25.58PoWeRKiLLThe start of the page yes
19:26.04PoWeRKiLLthen it's corrupted
19:26.38coppicesend me the pcap and I'll take a look
19:27.50Gokee2chazzam, Interesting, thanks!  Fixed http://pastebin.com/u2EKyT5P
19:28.46ChannelZrjune: mostly yes
19:29.20ChannelZrjune: it doesn't necessarily mean it succeeded AFAIK
19:30.48JonathanRoseIt could also just be configured backwards.  Here's mine, using Wildcard TDM410P, and it works:  http://pastebin.com/tnuhYe6K
19:30.58JonathanRoseGokee2: poke
19:31.22JonathanRoseI think the auto-generated script might have been backwards for me at first by the by.
19:32.11ChannelZIt just depends on the order the modules are installed
19:33.00ChannelZAnd remember an 'FXO port' uses FXS signalling and vice versa, so the config might always seems backwards from how you think of it
19:33.35JonathanRoseYeah, but I mean... if he's using FXS on FXS, it won't work, and everything will look right, but he wouldn't get dial tone.
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19:34.31JonathanRoseI'm no dahdi guru though.
19:35.15Gokee2JonathanRose, Its the correct order.  The first two ports work, and its worked with that config up until the card was moved.
19:36.13JonathanRosethat seems pretty reasonable then.
19:36.37Gokee2I am betting the power was just never plugged in when it was moved
19:37.02Gokee2Not sure though, should that kill the dialtone as well as ringing?
19:37.13JonathanRoseYes
19:37.26Gokee2Ok :)
19:37.54JonathanRoseI think the power is basically just used to amplify the outgoing signal.
19:38.02Gokee2Ah
19:38.46rjuneChannelZ, I just needed to know if that meant it tried
19:38.54rjunelooks like channel 1 has no dial tone
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19:47.15asterisk978Upgrading from 1.6.2 to 1.8.5 causes issues with my dialplan when using CDR(accountcode) and then dialling a local channel you can not access the accountcode. This worked in 1.6.2 but nolonger in 1.8.5. Is this a bug?
19:50.37jeffspeffdoes res_fax.so and res_digium_fax.so conflict?
19:51.22p3nguinThat reminds me I still haven't fixed my fax support.
19:51.49Qwelljeffspeff: The latter requires the former.
19:52.13asterisk978sorry should have said when using SET(CDR(accountcode)=23244)
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19:52.20jeffspeffhmm... thanks Qwell
19:52.55jeffspeffQwell, do you have a second to lend a hand with fax detection not working right?
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19:59.01asterisk978can any one help? I raised this as a bug, however it was closed
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20:01.50diegocnhello ppl... can asterisk be used behind a proxy like TOR?
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21:04.24p3nguinloader.c:382 load_dynamic_module: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: ast_fax_state_to_str
21:04.29p3nguinWhat's the fix for this?
21:07.00p3nguinThis is asterisk 1.4.39.2, res_fax-1.4_1.3.0-x86_32, and res_fax_digium-1.4_1.3.0-i686_32.
21:10.43p3nguinI guess I'll try res_fax_digium generic instead of i686.
21:13.59p3nguinThat didn't help.
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21:21.05p3nguinIt looks like using preload => res_fax.so helps res_fax_digium.so to load correctly.  I guess that indicates some sort of race condition during loading of modules.
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21:22.43chazzamJonathanRose: Gokee2 power is required to the board for FXS ports to work. period. They cant draw the required power from the pci bus
21:23.29chazzams/2/2:/
21:23.35chazzamI forgot that did that...
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21:44.41radicis there a way to place a call from the CLI und redirect it to a SIP-phone?
21:45.15p3nguinredirect, no.  Connect it to, yes.
21:45.24p3nguinCLI, originate
21:46.19p3nguinDo you want the call to first hit the SIP phone and then ring out to the other phone, or do you want it to ring out to the other phone and if someone answers then give it to the SIP phone?
21:46.39chazzamI just want cookies
21:46.41radicp3nguin: the second
21:47.31p3nguinDo you have extensions configured to call out to that number and also to call the SIP phone?
21:48.19p3nguinIf so, use something like this:  originate Local/3149691077@outbound_calls extension 3001@phones
21:48.54p3nguinIf not, use something like this:  originate SIP/valid-peer-here/3149691077 extension 3001@phones
21:49.23p3nguinIf the first part of that gets an answer, it will call extension 3001 in the phones context (your SIP phone's extension).
21:50.10p3nguinvalid-peer-here is, of course, your ITSP peer name as configured in sip.conf.
21:51.20radichmm
21:52.15p3nguinI know that "hmm."  What part are you having trouble with?
21:53.03chazzamI think he wants cookies now
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21:55.23radicp3nguin: I'm not at home an there is only a phone for incoming calls and I want that asterisk calls the number and If a recive the call it let ring for example the SIP-phone withe the extension 374
21:56.05p3nguinCan you repeat that again in plain English, please?
21:57.30p3nguinYou asked how to make a call from the CLI and connect it to a phone.  I provided you with that information.
22:02.20radicp3nguin: and 3149691077 is the number that should be called in your example?
22:03.26p3nguincorrect
22:04.22p3nguinIf you already have an extension to call out, I'd use the Local channel to call out.  See first example.
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22:06.13radicp3nguin: and If I take up the reciver (in the 2nd example) it calls 3001 in the extension phones?
22:07.21p3nguinBoth examples will call extension 3001 in the phones context if the outside number called answers.
22:08.12p3nguinLocal/3149691077@outbound_calls  utilizes a preconfigured extension in a context by the name of outbound_calls.  If your context is something else, change it.
22:08.27p3nguinIf you don't wnat to call 3149691077, change that.
22:08.49p3nguinIf your SIP phone's extension isn't 3001 in context phones, change that, too.
22:09.08p3nguinIt's an EXAMPLE.  You can surely figure out what bits need changed for your real usage.
22:09.25radicI'll move to the phone and tray it...
22:10.31p3nguinUnless you tell me what number you want to call, what context your outbound calls are going through, what SIP peer you call out of, the extension number of your SIP phone, and what context the SIP phone's extension is in, I can't write the exact literal copy/paste command for you.
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22:27.37radichmpf
22:27.55radicall phones here arn't working...
22:28.01radic+e
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22:28.46talntidAsterisk died with code 1.
22:29.06talntidlogs aren't giving any errors about it.... any ideas what usually causes that? dialplan error?
22:42.32radictalntid: what did you before asterisk died?
22:43.51talntidthis is a customer that called me
22:44.01talntidbut supposedly... http://www.withsupport.co.uk/node/83
22:44.24WIMPytalntid: I haven't seen dilpaln errors causing serious truble in operation. but they can cause crahses on reload.
22:44.39talntidi have returned both of those modified files back to factory....
22:44.57talntidbut asterisk still will not start, and doesn't really give me any errors to go by
22:45.20talntidi assume that because those files tried to modify the extensions_*.conf files... that my issue must be there..
22:45.37radichallo WIMPy
22:45.42talntidis there a way to regenreate those extensions_*.conf files, from command line? the web GUI will not load.
22:45.46WIMPyHi radic
22:46.23radicWIMPy: sag doch das du da bist :P
22:46.30WIMPytalntid: They are generated by the GUI, so probably not. But you need to ask that in #freepbx.
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22:56.40WIMPyMFBS! :-(
22:57.09WIMPyNow my dundi peers will only work when they are cashed.
22:57.44WIMPyThere had to be a catch to the loopback lswitch. It just worked far too well.
22:58.54WIMPyIs there an way to make a loopback wait for its target?
23:00.20jeffspeffI'm having troubles with fax detection... I have a fax extension set up on my inbound context, but when the fax comes through it just does an auto fallthrough and doesn't actually detect the fax. does anybody have any ideas?
23:00.59WIMPyNo. I't not about cacheing. It just works sometimes. F***
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23:26.48ChannelZjeffspeff: what version of *
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23:34.35jeffspeffChannelZ, 1.8.5
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23:41.37ChannelZhmm.  I guess I should test mine and see if it's still working.
23:42.00ChannelZThere were some problems with prior versions where internally it was trying to jump to a bizarre extension.
23:42.52ChannelZYou just have faxdetect=incoming set for your channel(s), and an extension called 'fax' in the context for those channels?
23:43.15jeffspeffChannelZ, hold on, and i'll do a pastebin of what i have real quick
23:46.18jeffspeffChannelZ, this is the relevant parts of extensions.conf ---> http://pastebin.com/PgzyZVGi        and then in sip.conf, I have "faxdetect=cng" set under [general], I have also tried "faxdetect=yes", and there wasn't a difference in results.
23:47.11Tim_Toadyjeffspeff: set wait at least to 3 seconds
23:47.19Tim_Toadytakes sometime to detect the tone
23:47.36jeffspeffTim_Toady, ok, let me set that and test it again... just a minute
23:47.47Tim_Toadyor maybe 4, its trial and error
23:48.00Tim_Toadybut for me it works at 3-4
23:48.35jeffspeffdoes it hurt to have it set to high?
23:49.49Tim_Toadyno, ur callers will just wait a bit more before reaching u:P
23:51.00jeffspeffTim_Toady, ChannelZ, I set the wait for 4 seconds, then sent a fax from our ancient fax machine that's not connected to this system, (asterisk should receive it as an incoming fax), but i got the same results
23:51.43jeffspeffTim_Toady, ChannelZ, FYI I'm using fax for asterisk by digium, and trying to configure it to use g.711
23:51.47ChannelZyou don't want Wait, you want WaitExten
23:52.06WIMPyIs there something like a minimum length for dundi requests? It seems to work if I dial more than one digit befor lifting the handset.
23:52.18ChannelZoh no, nevermind, fax detect I think happens regardless...
23:52.27WIMPyNot that anyone would do so...
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23:54.16WIMPyNo.
23:54.24ChannelZjeffspeff: going backwards, I don't think "faxdetect=cng" means anything, it should just be 'yes'
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23:54.45WIMPymakes a big mental not: Always use ! in switch patterns, never ..
23:56.01WIMPyLet's see what side effects this will have...
23:56.22jeffspeffChannelZ, I got that setting from the asterisk book, for version 1.8... you can set faxdetect in sip to be cng, t38, yes, no
23:56.30jeffspeffbut i'll try anything at this point
23:57.53ChannelZhmmm.. maybe the/my samples are out of date
23:58.29ChannelZIndeed mine are, my bad..
23:58.30Tim_Toadyjeffspeff: we talk about fax that arrives to asterisk by sip or some fxo port?
23:58.38jeffspeffsip
23:58.54Tim_Toadycodec set to g711?
23:59.18Tim_Toadyis the call indeed set up in g711?
23:59.25jeffspeffwhere do i set that at? i see in my fax license where it has an avaialbe g711 channel
23:59.53jeffspeffthe calls are all using ulaw
23:59.55ChannelZIE when you call in with your fax machine, it's not coming in as g729 or something from your provider?
23:59.58Tim_Toadyin sip.conf disallow all other codecs and only allow ulaw

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