irclog2html for #asterisk on 20050725

10:16.19*** join/#asterisk jbot (ibot@apt.bot.TimRiker.active.supporter.pdpc)
10:16.19*** topic/#asterisk is Asterisk: The Open Source PBX || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com || Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - Speakers wanted
10:18.02*** join/#asterisk dacleric (~dacleric@p5482B6D9.dip0.t-ipconnect.de)
10:19.46emrahAnyone here uses AreskiCC please? I just have a question about the AGI script. I'm having like an error when I launch the script.
10:21.16florzgres: try copying it to some other variable first, from the dialplan
10:22.56emrahAnyone use AreskiCC here?
10:23.25*** join/#asterisk amir (~amir@195.226.9.186)
10:23.39gresflorz, ok. thks
10:30.18nDuffemrah, maybe, maybe not -- but if you ask your question and wait around, you have a better chance of getting it answered than not asking until someone tells you they'll help.
10:32.25Kravenis there someone who can help me with bintec rcapi?
10:35.12clive-kraven what is that
10:37.32KravenCapi over Ethernet
10:40.56gresflorz, before execute my agi script, i do SetVariable(CALLDST=${EXTEN}). Then i recive in my script CALLDST! But i cat't still recive DIALSTATUS in my script after execute DIAL SIP/2760|30|tr...
10:41.19ManxPowerYou can't get automagically set variables in AGIs
10:41.48ManxPowerDo a SetVar(MY_EXTEN=${EXTEN}) before calling your AGI and access ${MY_EXTEN}.
10:42.37gresManxPower, I understand, thanks.
10:42.39ManxPowerAnd if you do a Dial from within your AGI I know of no way to access things that are set by Dial, like DISALSTATUS.
10:42.46ManxPowerThat's why I NEVER EVER use Dial from inside an AGI
10:43.20gres:)
10:44.22*** part/#asterisk fourcheeze (~rich@westbury.doilywood.org.uk)
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10:57.30tzafriremrah, maybe nobody, but ask your question anyway, for the record
10:59.38Jas_WilliamsKraven: is this what you are talking about http://lists.digium.com/pipermail/asterisk-users/2005-June/112893.html
11:01.49*** part/#asterisk d00gster (~doughant@toronto-HSE-ppp4334604.sympatico.ca)
11:02.06joergany idea how I can detect wether the phone is picked up when I do auto-dial-out with asterisk?
11:02.39joergI'm using sipgate.net for pstn connectivity
11:02.59ManxPowerjoerg: In what way?
11:03.08joerghttp://www.voip-info.org/tiki-print.php?page=Asterisk+auto-dial+out
11:03.11ManxPowerDIALSTATUS will give you that status of the Dial
11:03.30joergit's exactly what is written at "please note"
11:03.39ManxPowerThis doesn't apply to dialing out via ANALOG FXO ports, of course.
11:04.01ManxPowerjoerg: you mean this " If you are using POTS (Plain Old Telephone System) lines attached to a channel ban....."
11:04.07joergyes
11:04.17joergI dial out via a sip to pots gateway
11:04.17ManxPowerAre you using analog ports?
11:04.42ManxPoweran ITSP like sipgate should be using PRI ports and so it's not an issue.
11:08.32joergmhh...
11:08.37joergso what can I do? :)
11:08.45joergmaybe iptables blocks sth?
11:08.52ManxPowerjoerg: nothing.  You should be getting the correct DIALSTATUS back.
11:09.27ManxPowerUnless there's a problem with .call files, which can be the case.
11:09.49ManxPowerWhat IS DIALSTATUS set to when a call ends?
11:11.04joerghow do I check that?
11:11.33ManxPowerin your dialplan.
11:11.57ManxPowerYou need to test this WITHOUT using .call files first.
11:13.02pacant i have asterisk more verbose than vvv?
11:13.08ManxPowerpa: yes
11:13.25ManxPoweryou can have up to 255 v's and up to 255 d's
11:13.34paumh.. i used -vvvvvvvc but the error is the same
11:13.41joergManxPower: I have a dialplan for dialing out manually from my voip phone
11:13.46joergManxPower: that works very well
11:14.00ManxPowerjoerg: and what is DIALSTATUS when you do it that way?
11:15.09ManxPowerYou MAY have to use a Local/ channel in your .call file to make the call be processed via the Dialplan.  Maybe not.
11:16.14ManxPowerhttp://voip-info.org/tiki-index.php?page=Asterisk%20local%20channels
11:17.11*** join/#asterisk gonzo- (~gonzo@gif.lesnik.portaone.com)
11:25.21emrahPlease, anyone here uses AreskiCC
11:26.29joergManxPower: its the same problem
11:26.44joergManxPower: when I call my voip phone it works of course
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11:29.32puzzledmorning
11:31.50florzpuzzled: I guess any of the more recent ones should do, you just have to fix the zaptel include path version manually.
11:32.25puzzledflorz: thanks :)
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11:38.54darkskiezcan I do integer math easily?
11:40.38X-Robyes. 1+1 = 2
11:40.40X-Robthat's pretty easy
11:40.44darkskiezin a dialplan
11:41.03darkskiezMath(var=1+1), results in var=2.00000
11:47.17darkskiezso ?
11:47.39ManxPowerdarkskiez: try SetVar(VAR=$[1 + 1])
11:47.55darkskiezoh ta, cheers
11:47.59ManxPowernotice the spaces around +
11:48.22darkskiezand can local channels call an extension that matches a pattern, i'm getting errors that they cant be found
11:48.36ManxPowerit should be able to
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11:57.15darkskiezdont think it was liking alpha in the extension
11:59.01ManxPowerShould not be a problem.
11:59.10ManxPowerI use alpha extensions with Local/ and .call files.
11:59.24ManxPowerWell, at least non-number extensions.
11:59.25darkskiezi just created an infinite loop, oopps
11:59.34ManxPowerdarkskiez: That would be a problem. 8-)
12:01.10*** join/#asterisk fenlander (~neils@82.152.81.57)
12:01.33darkskiez953 active channel(s)
12:01.33darkskiez477 active call(s)
12:01.34darkskiezarse
12:07.48*** join/#asterisk lathos42 (~lathos42@65-42-27-66.dowdingindustries.com)
12:12.12*** join/#asterisk daork (~daork@don.bhnb.net.nz)
12:12.36daorkcan i force a digium FXO port to stay offhook?
12:16.24daorkwe've got 8 lines in a hunt group, and we only know where 7 of them physically are, so we're going to plug them all in, and get them offhook and then call the hunt group number
12:17.40ManxPowerdaork: Not really.
12:18.08ManxPowershort each of the analog lines to make them busy
12:18.37lathos42daork: Does the 8th line have its own phone number?  I know our Analog lines all do.
12:19.02ManxPowerYou can, of course, also just dialout from each of the lines and call somthing that supports callerid
12:19.08ManxPowerthen you can find each line.
12:20.07ManxPoweranalog POTS lines always have their own phone number
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12:25.44So3krisknows someone if the x100p uses 2 drivers i for the voip and 1 for the modem
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12:28.13QbYin what order does asterisk read extensions.conf? does it read extensions.conf and insert my includes into it in the order where they are 'included'?
12:28.27QbYor does it follow extensions.conf first, then look into the others?
12:28.28X-RobQbY yes
12:28.35X-Roborder of inclusion
12:28.41QbYk
12:29.29daorkManxPower: actually, these dont. sorta
12:29.43daorkthey all do, but the missing line's numgber is the one that hunts across the tohers
12:29.46daorkothers*
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12:32.17X-Robbah
12:32.22X-Robcvs is down. *grump*
12:34.19*** part/#asterisk daork (~daork@don.bhnb.net.nz)
12:35.38QbYhttp://pastebin.ca/18446
12:36.34QbYI am trying to get all of the calls to this number into a special automated attendant (so the caller doesn't have to go through the first)..  but all of the calls are being answered by the main attendant..  and its ignoring my extensions_custom --- the playback(abandon) was put in simply to test to see if its working..  http://pastebin.ca/18446
12:38.53X-RobAre you actually using AMP?
12:39.02QbYyes.. b
12:39.04X-Robor is it just the remnanants of an AMP system?
12:39.20QbYmore the remnants..  because i'm doing the custom stuff by hand
12:39.32QbYamp can't do (or i haven't found a way to) what i need done
12:39.47X-Robwell, if you were using AMP, you'd go to DID and set that number to go to the digitial rececptionist
12:40.09*** part/#asterisk jeffik (~Jeff@toronto-HSE-ppp3985436.sympatico.ca)
12:40.10QbYDID's won't do it for some reason..
12:40.15X-Robyes it will
12:40.23X-Robthat's what they're for.
12:41.07QbYson of a bitch
12:41.18QbYyesterday they wouldn't..  and some guy had me doing it this way
12:41.36X-Robask the guy doing the AMP documentation next time (eg, me 8)
12:41.43QbYhehe
12:41.48X-Roband you're not even _on_ #amportal
12:42.07QbYfigured it was more of an asterisk question
12:42.32ManxPowerWhen using any WebGUI no question is an Asterisk question.
12:43.18QbYyes sir
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12:54.33*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
12:55.04[TK]D-FenderJust read some news on SineApps.... anyone got a decent link for info on the new AEL scripting feature comin up?
12:57.45joergManxPower: hey, are you there? :)
13:04.15darkskiezYay, my cool recursive hunt macro script trick seems to work, huzzah
13:06.16*** join/#asterisk [TK]D-Fender (~joe@4.67.252.216.dsl1.colba.net)
13:08.32JunK-Utk: REAME.ael ?
13:12.48*** join/#asterisk LoRez (lorez@lorez.staff.freenode)
13:14.45tuxinator_linuxMwatch out, its a staffer ;-)
13:16.00[TK]D-FenderJunK-U : Where would I find that?  In a new download of CVS-HEAD?
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13:17.13JunK-Uya
13:18.40Darwin35what is asterisk? what does it do? what are the requirements to run it? where can I get it ? how much does it cost ? how much is support ?
13:19.36ManxPower[TK]D-Fender: Well you are not going to find AEL in 1.0.x
13:20.13pnvikingA pbx, connection phones together, linux (win/mac), www.asterisk.org, 0$, How much do you want to pay?
13:20.48[TK]D-FenderManxPower : Ok, where should I go for a heads-up on the new features since they don't seem to be in the Wiki so far?
13:21.04ManxPower[TK]D-Fender: README.upgrade?
13:21.15ManxPowerand of course Changelog if it's been updated.
13:21.23ManxPower[TK]D-Fender: the correct place is the asterisk-cvs mailinglist.
13:21.40ManxPowerAnyone using CVS-HEAD that is not on the asterisk-cvs mailing list is an idiot.
13:22.16*** join/#asterisk lehel (~Lehel@82.79.20.17)
13:22.24felipexanybody using chan_bluetooth or miax ?
13:23.30tzangermiax?  What's that?
13:23.33*** join/#asterisk PCadach (~paul@212.19.157.154)
13:23.39lehelhello
13:24.27|nixi'm the biggest idiot
13:24.27|nixhahah
13:25.56*** join/#asterisk Katty (~angela@68.112.15.110)
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13:26.29Hmmhesayshar har
13:28.57Kattymew
13:29.05Darwin35my embedded board should be here today
13:29.33puzzledtzanger: http://www.voip-info.org/wiki-Mobile+IAX
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13:30.00*** mode/#asterisk [+o anthm] by ChanServ
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13:30.41fitzelMoi
13:31.51Ahrimanesmmmust.. hhhave.. caffeine..?
13:32.19*** join/#asterisk _-Jon-_ (jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com)
13:32.23_-Jon-_good morning everyone
13:32.46HmmhesaysBRAINS
13:33.22CoaxDBRAINS!@#$
13:33.24Hmmhesayswell, coffee will do I guess
13:33.28Ahrimaneshehe
13:33.30_-Jon-_I'm wondering if anyone can assist me in linking 2 asterisk boxes together via iax.  I'm getting confused with the config file between type=peer, type=friend, type=user, etc
13:33.41iCEBrkrCoaxD: Yea, you gotz none.
13:33.44CoaxDthat was one fo the cheesiest movie series' i've ever seen
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13:33.51CoaxDIcebrkr: not this morning I don't, you're absolutely right
13:34.00iCEBrkrCoaxD: hehe
13:34.00Darwin35read  asterisk/configs/sip..conf.sample
13:34.17iCEBrkrIAX != SIP
13:34.18Darwin35it explains them
13:34.27Darwin35its the same in both
13:34.43_-Jon-_Alright I'll give that a read
13:34.49fitzelAnyone here with some practical experience for a softphone on a windows mobile PDA with wifi/wlan?
13:34.58CoaxDgawd, my data entry company that i work for - has a stupid server problem that isn't allowing me to download my batch headers.   Can't do my work without it.  I'm getting annoyed.
13:35.46iCEBrkr_-Jon-_: I used type=peer in my 'master' asterisk server in my iax.conf
13:35.48CoaxDooh, i had the batch anyway. i can do my work now. woo!
13:35.55Darwin35I hate mondays
13:36.13fitzelDarwin, they will pass, as every monday.
13:36.57Darwin35could not get comfy in bed
13:37.08Darwin35or on the couch or on the floor
13:37.10iCEBrkrDarwin35: Drink a few beers before bed :D
13:37.17*** join/#asterisk tengulre (~tengulre@219.145.57.79)
13:37.50Hmmhesays:) morning Katty
13:38.03fitzelis there an iax-client available for windows mobile? or only sip-stuff?
13:38.12KattyHmmhesays: i'm all groggy :<
13:38.20Darwin35iaxcomm
13:38.30HmmhesaysKatty: why is that?
13:38.40KattyHmmhesays: uhmm, cause my caffeine iv is not installed?
13:39.03Hmmhesaysahh I see, if it makes you feel any better I'm all sore
13:39.18Katty:<
13:39.28Kattyi'm a little sore in sunburnt fashion
13:39.43HmmhesaysI'm a bit sunburnt and sore from wakeboarding
13:39.50Kattyi was swimming
13:39.59Kattyand dunking $person under teh water
13:40.08HmmhesaysI did a bit of that too
13:40.14Hmmhesayslake or pool?
13:40.18Kattypool
13:40.24Kattyi don't swim well enough to be in a lake, i don't think
13:40.40Darwin35firefly
13:40.45Kattynot without another human floaty device
13:40.45Darwin35does iax also
13:41.03Hmmhesaysahh I see
13:41.06fitzeliaxcomm on a pda?
13:41.19Kattyiaxcomm isn't a j2me app
13:41.26Kattybut i hope to write one (=
13:41.58Kattyspeaking of such, my new beep beep phone comes in today :>
13:42.03Kattytwisted[asteria]: and then i shall bug you!
13:42.05Kattytwisted[asteria]: BEEP BEEP
13:42.31KattyDarthClue: you'll be happy to know i'll set my little 128 pixel wallpaper to the cluecon logo (=
13:42.46*** join/#asterisk astoria (~haydenth@66.235.201.217)
13:42.47KattyHmmhesays: besides lack of coverage? ;)
13:42.52astoriaGood Morning all.
13:42.55Hmmhesayshaha nope
13:43.07KattyHmmhesays: and the nextel phone support?
13:43.17Hmmhesaysnever talked to their support
13:43.18KattyHmmhesays: and their insane notion you should pay for upload software?
13:43.30KattyHmmhesays: whyfor grumbling?
13:43.43Hmmhesayscause I got no beep beep phone
13:43.49Kattyaww.
13:44.12Kattyis there nextel coverage out there?
13:44.19Kattyor are you waiting for your contract to end?
13:44.21fitzelI am looking to get the ipaq H6340, is it worth it? Can it be used to phone via WLAN?
13:44.32Hmmhesayshrm, they do have coverage up here now
13:44.39Hmmhesaysthey didn't last time I looked
13:44.45Kattyfitzel: you should probably visit the website the ipaq is posted on and read the specs :)
13:44.51KattyHmmhesays: :>
13:44.54astoriaEverytime I hear a nextel phone in a public place I want to vomit.
13:45.01Kattyastoria: you would
13:45.11Kattyastoria: lots of people don't turn the privacy setting on
13:45.12fitzelI know the specs and the website, but some first-hand experience is more interesting.
13:45.23Kattyastoria: and so it blairs like a walky talky
13:45.43Hmmhesaysheh, bad plans though geez
13:45.52fitzelit HAS a WLAN and there are some sip-clients available. But I am curious, how far the wlan can reach.
13:45.55Kattyfitzel: you might try going to a retailer and asking (=
13:46.11fitzelI am living in countryside
13:46.13Kattyk
13:46.16mutprobly a few hundred feet like any other generic wlan device
13:46.24astoriaI hear those damn things go off in class all the time.. It annoys me like no other.
13:46.28pacan i search somehow asterisk-users archive?
13:46.32mutlike 100ft
13:46.38astoriapa: google site:digium.com
13:46.59paoh, ok
13:47.15Kattyastoria: it's not just nextel phones
13:47.25Kattyastoria: there are all sorts of people who leave their phone on when it's not appropriate
13:47.33Kattyastoria: and with all other providers, there is a vibrate option (=
13:47.49astoriaKatty: i know, but nextel users are especially inconsiderate.
13:48.06astoriaKatty: or i just recognize their ring more.
13:48.19pahere my problem: http://lists.digium.com/pipermail/asterisk-users/2005-July/118413.html
13:49.29Kattyastoria: well, i'm quite considerate
13:49.44*** join/#asterisk oej (~oej@apollo.webway.se)
13:49.50Kattyastoria: i can't imagine why nextel people would be less considerate than other people
13:50.01Kattyastoria: they're all the unwashed public, afterall
13:50.11tuxinator_linuxMastoria: I was in court when one of the jurors phones did that walkie talkie thing, dead silent afterwords
13:50.52*** join/#asterisk zotz (~zotz@24.231.36.100)
13:50.53tuxinator_linuxMserious, whats worse, your phone ringing, or your phone talking really loud
13:51.06*** join/#asterisk lters_ (~lters@eg1.ekn.com)
13:51.14tuxinator_linuxMlathos42: what * in the middle ;-)
13:51.17Nuggethopefully never.
13:51.30Kattypif: ..?
13:51.52tuxinator_linuxMmay, it only takes a second to connect, how much faster do you need it?
13:52.00Kattylathos42: probably awhile yet, nextel is going to change their name to sprint i think
13:52.05Kattylathos42: and that hasn't happened yet
13:52.05astoriaIt's way worse when your sitting in class and you hear the nextel beeps followed by a, "HEY. YOU THERE??"
13:52.16Kattyastoria: now /that/ is impolite
13:52.28tuxinator_linuxMI concure
13:52.32Kattyastoria: i don't talk to someone if they don't answer their page
13:52.34astoriaI've seen people answer int he middle of lectures!
13:52.35tuxinator_linuxMconcure, he he
13:52.49*** join/#asterisk eKo1 (~bernd@metrored-gw.tropicohn.com)
13:52.55Hmmhesaysits in my sight, and the timing is right, for taking a bow, into the now
13:52.56lathos42astoria: I dont want a Nextel for exactly that reason.. I know alot of people that expect you to respond immediately if they direct connect you
13:52.57tuxinator_linuxMI think it is time for bed
13:52.58astoriaANd they'll just start talking...
13:53.25tuxinator_linuxMHey tiger, come over to my cube....
13:53.37Kattyeveryone has rude family members
13:53.42Kattymy family has several
13:53.48tzangerKatty: :-)
13:54.15Kattytzanger: mew (=
13:54.42tzangerhow are you Katty
13:54.59pahere this guy seems to have my same problem:
13:55.01pahttp://lists.digium.com/pipermail/asterisk-users/2004-June/051684.html
13:55.01lathos42Of course, then again, some of those same people call my phone over and over again until I answer or they get tired of hearing my voicemail greeting
13:55.01Kattytzanger: sunburnt and groggy
13:55.07pano answers :(
13:55.14tuxinator_linuxMlathos42: You too ;-)
13:55.20tzangerKatty: that's not good
13:55.28Kattytzanger: i'll get over it (=
13:56.15paumh.. isdn voice support into kernel?
13:56.19pai didnt noticed it..
13:56.23palet me check
13:56.44ManxPowerlathos42: nobody knows my cell phone number.  They MUST call my extension and dial 0 if they get my voicemail if they have something urgent.
13:57.02*** join/#asterisk willim_M (~icechat5@62.231.36.101)
13:57.13paOK, prolly my fault
13:57.19*** join/#asterisk skeffling (~Andrew_He@andrew.1ec.aaisp.net.uk)
13:57.20pathat should be the problem
13:57.21ManxPowerIf a call comes into my cell phone and I know it was a direct call I'll usually answer the call, scream "YOU HAVE THE WRONG NUMBER!" and then hangup.
13:57.27Darwin35I love my dial the weather
13:57.33willim_Mwhat ports do i need to open for iax
13:57.44Darwin35look in the iax.conf
13:57.47Darwin35it tells you
13:57.55willim_Mthanks
13:57.56astoriaread the wiki, there is a page about firewalls.
13:58.04lathos42ManxPower: None of my coworkers have my cell phone number, and I plan to keep it that way
13:58.07Kattywillim_M: i'll get you my port, hold on
13:58.20astoriaWhy are you afraid of co-workers calling your cell?
13:58.25astoriaI have a direct extension for my cellphone.
13:58.32astoriaI WANT them to call me.
13:58.33Darwin35people read the wikie and the conf files before asking stupid questions that are already covered
13:58.49Kattywillim_M: mines on 4569, and udp (=
13:58.55skefflingHello, we've just starting seeing loads of these in the asterisk CLI/log file....
13:58.57skefflingJul 25 14:45:34 ERROR[14205]: utils.c:509 tvfix: warning negative timestamp -198466.-574000
13:58.58skefflingand I've no idea what the problem is!
13:59.07astoriaKatty: you're too nice.
13:59.14Hmmhesayswoo hoo hoo
13:59.19willim_Mi already opened port 4569 and it didnt work
13:59.26Kattywillim_M: then it's not a port problem
13:59.27doolphanyone good with gnugk?
13:59.33Kattyastoria: and you are too annoying
13:59.38Kattyastoria: m'kay?
13:59.38astoriaKatty: ha ha, am i?
13:59.49*** join/#asterisk yaaar (~chatzilla@lifebook.tranquility.net)
13:59.54yaaarword
13:59.55Kattyastoria: well nothing here would work if Hmmhesays always told me to go read the wiki (=
14:00.11Hmmhesaysheh
14:00.15astoriaKatty: i was just kidding, but i know what you mean.
14:00.21Kattyk
14:00.59Kattyanthm: which reminds me i need to redo my dial plan to define the callerid for 6 and 2 lines
14:01.00lathos42Alot of my coworkers use their personal cell phones for business use, even though the company will happily buy you a cell phone if you need one
14:01.12Kattyanthm: so maybe later you can look at it and tell me what i've goofed up after i give it a try (=
14:01.20yaaarHmmhesays: hope you have your astrolabe
14:01.24Kattyyay for anthm making my dial plan pretty
14:01.29anthmok
14:01.37astorialathos42: i prefer to use my personal cell phone because the amount of minutes that work uses up is much less than my personal minute use.
14:02.04anthmdefine the callerid on who?
14:02.05astorialathos42: it's a hassle to deal with that kind of stuff if the office is paying for it.
14:02.18Hmmhesaysthis is like walking through a forest with a blindfold on
14:02.22joergis anyone reachable via sip?
14:02.27joergwould like to try out sth
14:02.47lathos42astoria: Yeah, my last job they just reimbursed me for part of my personal cell phone..  Which did have the advantage of not needing to carry two phones
14:02.47joergmy automated callback doesn't work
14:04.13Hmmhesayswhy not?
14:04.16lathos42Hmmhesays: Thanks for giving me flashbacks to when I tried to find information on the Shiva LANRover on Intel's site.. I just now stopped having nightmares about it
14:04.31Hmmhesayslathos42: No Problem
14:04.48lehelphrrr.. i cannot make an IAX call.. friday i could, today not anymore ???
14:06.14lehelRx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX     Subclass: INVAL
14:06.28joergHmmhesays: are u talking to me? :)
14:06.41Hmmhesaysyeah what is wrong with your callback deal?
14:07.15joergthere must be sth. wrong with sip
14:07.38joergit starts the musiconhold app when I answer the phone
14:07.44joergbut I don't hear anything
14:08.05Hmmhesaysare you using a callfile or the manager to originate the call
14:08.18joergcallfile
14:08.34*** join/#asterisk kimosabe (~nat@dsl-200-67-12-220.prod-empresarial.com.mx)
14:08.53joergthe callfile calls Local/200@wakeup
14:09.03joergand that calls a sip phone
14:09.31Hmmhesaysare you in a nat environment?
14:09.39joergnope
14:09.52Hmmhesayscan you call into a moh extension with that sip phone?
14:10.02joergthe confusing thing is, that it works, when I initiate a call manually with my sip hone to outside
14:10.44Hmmhesays<Hmmhesays> can you call into a moh extension with that sip phone?
14:11.06joergHmmhesays: let me explain you my setup :)
14:11.12Hmmhesaysno
14:11.15Hmmhesaysjust answer my question
14:11.17joergok
14:11.18joergyes
14:11.34joergthe sip phone is actually a phone on pstn
14:11.54Hmmhesayssame phone you are originating the call to?
14:11.59astoriaCan SpanDSP do ECM?
14:12.13joergyes
14:12.21astoriaWhere do I enable it?
14:12.27joergsipgate.net acts as gateway
14:12.29*** join/#asterisk brookshire (~matt@207.111.174.1)
14:12.58joergfrom the phone, I can call my voip number and hear my asterisk playing the greeting
14:13.04elduis there a way to force a codec per telco number within the same sip trunk ?
14:13.23joergbut the other way round it doesn't work
14:14.14Hmmhesaysargh, "click here for a non commercial license" *click* rinse repeat
14:14.33Hmmhesaysjoerg: that is a problem
14:14.40*** join/#asterisk Cadu20 (~Cadu20@200.102.53.174)
14:14.56lathos42astoria: From what i'm seeing, I dont think it does
14:15.03Cadu20How could I see the version/flavor of my G723 codecs?
14:15.07joergHmmhesays: placing calls from my voip phone via asterisk to the outisde world works.
14:15.18ManxPowerCadu20: you don't have G723.1 codecs
14:15.19astorialathos42: thats too bad. Maybe soon, when they finish the T.30 stuff.
14:15.22Cadu20Or how do I see more informations about a installed codec?
14:15.30joergHmmhesays: that is exactly the same thing the callback should do
14:15.40ManxPowerCadu20: "show codecs" and "show translations"
14:15.49Hmmhesaysexcept you are originating the call from asterisk itself
14:16.00*** join/#asterisk Goshen (~Goshen@67-40-107-29.slkc.qwest.net)
14:16.30Cadu20ManxPower, but show codecs only gives me G723.1.. the gw operator tells me they got g723.1r6... I want to know if I´m compatible.
14:16.35*** join/#asterisk jr352k ([U2FsdGVkX@pcp03618028pcs.univde01.de.comcast.net)
14:16.50Darwin35then you did not set your codecs in the conf files
14:16.55joergI have set verbose to 9
14:17.03joergit detects, when the line is answered
14:17.04ManxPowerCadu20: "show codecs" does not show the INSTALLED codecs, it just lists all codecs
14:17.13joergand starts the moh app
14:17.15Cadu20But i have installed G723.
14:17.23*** join/#asterisk syle2 (~blah@wnpgmb06dc1-167-98.dynamic.mts.net)
14:17.39Cadu20That no comercial license.
14:17.43ManxPowerCadu20: Don't tell us that.  It's illegal and you are breaking patent and copyright law, as well as Intel's own license.
14:17.54lathos42astoria:  I really like the idea of SpanDSP, but from what i've heard, I dont dare try to sell it to my company as part of our Asterisk system
14:18.06Cadu20Thats not what the folks told me.
14:18.16astorialathos42: I'm going live with it right now on a real-life implementation.. It's actually not bad. It gets a bad rep.
14:18.22Cadu20Tell you that I have installed a piece of software is ilegal too?
14:18.40*** join/#asterisk mogorman (~mogorman@207.111.174.1)
14:18.50syle23rd rule of business: don't get caught
14:18.51ManxPowerCadu20: I can tell you that you can fly.  That doesn't make it true.
14:18.52astorialathos42: I wish it had ECM, because that really comes in handy, but otherwise, it's okay.
14:19.09Cadu20Exactly
14:19.17ManxPowerCadu20: nobody here can help you with G723.1
14:19.19lathos42astoria:  What kind of success rate are you seeing with it?
14:19.22ManxPowerSince none of us use it.
14:19.26Cadu20So.. how do I get more information about installed codecs?
14:19.41ManxPowerCadu20: The source code is the only real docs for htat.
14:19.42ManxPowerthat
14:19.44astorialathos42: But the software is free, so I don't have a whole lot of room to complain. So far, about 80-90%. Way better than using a SIP adapter on a fax machine.
14:19.47syle2unfortunately ignorance is not an excuse in the law hehe
14:19.57ManxPowerHere is the licensing priceing info for G723.1 direct from the patent holder's web site: http://www.dspg.com/technology/LicensePricing.html
14:20.15astorialathos42: sometimes, it just mangles a page here and there, but thats because of noise on the line and the lack of ECM.
14:20.17Cadu20But they have download for the code? I didn´t found it.
14:20.31astorialathos42: once they get ECM working okay, I would say it's ready for prime time.
14:20.59Cadu20All right, i pay.
14:21.04Cadu20This is NOT the problem.
14:21.12Cadu20I JUST wanted to know, how the f* a
14:21.21Cadu20II diplay more information about a codec?
14:21.29lathos42astoria: That's cool..  I'd like to give everyone their own Fax DID, but I doubt the company will front the money for a T1 Fax Card and another server to run Hylafax on
14:21.51astorialathos42: plus you gotta pay for a bunch of pots lines too, spandsp lets me run it through a PRI.
14:22.14mishehupri pri pri
14:22.18syle2when it says you pay per channel, this mean if you have a PRI line you pay for 23 lines
14:22.30Cadu20? No one? Codec information?
14:22.37astoriasyle2: yeah, a PRI is 23 (technically 24) channels.
14:22.38*** join/#asterisk teapot (~tandrews@mail.grok.org.za)
14:22.49mishehuCadu20: http://www.voip-info.org ?
14:22.50syle2yeah but last one is used for data
14:22.51newlUnless you're talking about an E1 PRI ;)
14:22.56ManxPowerCadu20: Um, the patent holders will refuse to license G723.1 to you unless you comit to a LARGE purchase.
14:23.07astoriasyle2: the last one is used for signalling. Yeah, I'm talking about T1 PRI
14:23.12lathos42astoria: I'm hoping to continue my sales pitch for Asterisk on our Company president now that she's back from Vacation
14:23.15syle2how many in a E1 not familiar with europe, prob something like 27 hehe
14:23.29Cadu20ManxPower, if you don´t want to help, ok. I take that. But please, DONT DISTORT WHAT IM SAYING.
14:23.45astorialathos42: if you want to see how spandsp works on a fax, drop me a msg, and i'll give you my fax number and email you a copy of what it generates.
14:23.46newl30 plus data and sync.
14:23.48Cadu20By the way, thanks a lot.
14:23.50Cadu20My god...
14:24.15syle2anyone running SER?
14:24.23mishehuManxPower: what's so great about g723.1 that the patent holders are so cocky?
14:24.35ManxPowerCadu20: Use G729 if you MUST use a patented codec.  The G729 patent holders licensed it to Digium and Digium sells per channel licenses for G729
14:24.41ManxPowermishehu: nothing 8-)
14:24.44fearnorpatent holders are cocky, period
14:24.50lathos42astoria: It might be interesting to see how it handles a fax from our crappy Laserjet 3100 :)
14:25.04mishehuI donno, the g729 holders are somewhat reasonable.
14:25.08So3krisgewoon die kernel qosq
14:25.13pifbic!
14:25.19*** join/#asterisk grimse (~grimse@p5481D4AE.dip.t-dialin.net)
14:25.36mishehuIrq 233: no one cared!   <--- that's the  kernel panic of the day...
14:25.39*** join/#asterisk wunderkin (kev@12-215-218-160.client.mchsi.com)
14:25.40eKo1Why not use GSM or iLBC or some other non-patented codec.
14:25.46Cadu20ManxPower, i GOT g729
14:25.46syle2i run only hp printers now, never a problem since, owww i hated brother, 5 colors drying up is expensive
14:25.57Cadu20ManxPower, as I told, paying for it is NOT the problem
14:26.29ManxPowerCadu20: See http://lists.digium.com/pipermail/asterisk-users/2004-September/064099.html
14:26.39ManxPowerCadu20: There is no legal way to use G723.1 with Asterisk.
14:26.47syle2does noone run SER at all?
14:26.57fearnori run ser
14:27.02syle2k
14:27.03syle2PM
14:27.30*** join/#asterisk Mike (~mike@201.135.48.172)
14:27.35teapothohum
14:27.54*** join/#asterisk BoNaDs (~theplot@ool-44c53748.dyn.optonline.net)
14:28.13Mikeanyone having trouble with recent cvs and voicemail?
14:28.23teapotIs the bristuff ever going to make it into CVS ?
14:28.31BoNaDsto checkout cvs head, do i checkout -r vCVS-HEAD?
14:28.31*** join/#asterisk coppice (~chatzilla@30.195.17.210.dyn.pacific.net.hk)
14:28.53Mikehey steve
14:29.02ManxPowerThis kind of sucks.  Digium won't replace an old TDM400P (the one without the power connector) if it's older than 2 years.
14:29.23ManxPowerteapot: BRIStuff will never be in Asterisk
14:29.31BoNaDswhat is the power connector for?
14:29.36teapothuh ?
14:29.40BoNaDsalways wondered why it had one
14:29.43ManxPowerteapot: BRIStuff will never be in Asterisk
14:29.45teapotwhy is that ManxPower ?
14:30.01ManxPowerBoNaDs: because many motherboards cannot supply enough voltage over the PCI bus to make the card work
14:30.02BoNaDsdoesnt the board power from the pci bus?
14:30.02Cadu20Ok, thank folks.
14:30.17BoNaDsah so if i power the board while on the bus it wont hurt anything?
14:30.20ManxPowerteapot: because BRIstuff contains stuff from people that will not disclaim the code to Digium.
14:30.32teapotpah!
14:30.35ManxPowerBoNaDs: Well it wont work if you have FXS modules, that's all.
14:30.48BoNaDsbut i wont fry the board if i plug the power into it?
14:31.10ManxPowerBoNaDs: Huh?  If you don't plug power into the board the FXS modules won't work.
14:31.33BoNaDsright which i have none
14:31.35BoNaDsonly fxo
14:31.52fearnormanx: have you played with bristuff?
14:31.52ManxPoweron boards without a power connector the FXS ports may or maynot work depending on your motherboard.  Mostly they will work for a while then fail.
14:31.58ManxPowerBoNaDs: then it should not matter.
14:31.59BoNaDsbut i am having some wierd shit which i think would make sense to be relared to being underpowered
14:32.04ManxPowerfearnor: since I don't have a PRI....no.
14:32.12BoNaDsgonna give it a shot
14:32.15BoNaDsbrb
14:32.27ManxPowerBoNaDs: just be sure to do it when the system is powered off.
14:32.45fearnori'm thinking that doing BRI to the isdn phones seems to be a very cheap solution for non-ghetto phones
14:32.46ManxPowerfearnor: um...since I don't have a BRI...no.
14:32.59*** join/#asterisk mkrufky (~mk@68.160.103.77)
14:33.08Kattyi sure could use a new recipe
14:33.16Kattyi don't suppose anyone would like to volunteer their favorite
14:33.19*** join/#asterisk emp (~emp@70.57.239.37)
14:33.44ManxPowerKatty: procmail?
14:34.10KattyManxPower: you have a recipe for procmail? :P
14:34.25*** join/#asterisk kswail (~kyndar@modemcable244.73-81-70.mc.videotron.ca)
14:34.26KattyManxPower: i was thinking pasta, actually (=
14:34.52Ariel_morning everyone
14:35.01Ariel_hello Katty
14:35.10KattyAriel_: moo'rning (=
14:35.19BoNaDsbeing that i am cvs-dumbass, how do i get CVS-HEAD rather than v1-0 as shown on the website instructions
14:35.27ManxPowerKatty: My best source for pasta is Walmart 8-)
14:35.28BoNaDsvCVS-HEAD?
14:35.36KattyManxPower: i see.
14:35.52KattyAriel_: recipe?
14:35.59ManxPowerBoNaDs: don't specify -r v1-0 on your CVS command line
14:36.03BoNaDsah
14:36.13BoNaDsdanke
14:36.16mishehuI much prefer maildrop over procmail
14:36.17*** join/#asterisk Malthus (~admin@port0043-aas-adsl.cwjamaica.com)
14:36.23*** part/#asterisk Malthus (~admin@port0043-aas-adsl.cwjamaica.com)
14:37.01Kattymaildrop cookies!
14:37.22Kattythere should /so/ be an asterisk cook book in engineering layout
14:43.31*** join/#asterisk Blackvel (~blackvel@dsl-084-057-124-066.arcor-ip.net)
14:47.03syle2crap whats LCR stand for again
14:47.21skefflingLeast cost Routing
14:47.48*** join/#asterisk thieums (~darkmind@bea75-1-82-234-122-35.fbx.proxad.net)
14:48.00coppiceInductor, capacitor, resistor
14:48.24thieumsdo you know where I can find a running oh323 modules for 1.0.9 ?
14:48.48*** join/#asterisk darkskiez (~darkskiez@194.247.78.146)
14:48.54*** join/#asterisk Defraz (~t0tal@67.130.216.13)
14:48.55Hmmhesaysh323 ouch
14:49.03thieumsyes I know
14:49.06thieumsbut i need it
14:49.14Hmmhesaysall of the versions of oh323 or on the inaccessnetworks site
14:49.19Hmmhesays*are on even
14:49.26thieumsfu**ing customers
14:49.37*** join/#asterisk greg_work (~greg@d221-73-237.commercial.cgocable.net)
14:49.47darkskiezManxPower: you were wondering about the recursive script I was writing : http://voip-info.org/tiki-index.php?page=Asterisk+power+hunt
14:49.52Hmmhesaysyou might try ooh323 with cvs-head
14:49.59Hmmhesaysit seems to work alright
14:50.00*** join/#asterisk ChkDigit (~mike@static65-87-228-18.regina.accesscomm.ca)
14:50.06izoanybody has 7940 or 7960
14:50.07thieumsok I'll try it
14:50.08Hmmhesaysnot dependant on openh323 and pwlib
14:50.08ManxPowerdarkskiez: I've written recursive dialplan macros before 8-)
14:50.10thieumsthanks
14:50.45darkskiezManxPower: oh? perhaps recursive is the wrong word for this, multithreaded? :)
14:51.06ManxPowerdarkskiez: My macro calls itself when needed.
14:51.26darkskiezManxPower: whats it do?
14:51.48*** join/#asterisk SwK[Work] (~SwK@border0hsv.asterisksgi.com)
14:51.50SwK[Work]j #redbull
14:51.59anthmhmm, nobody every mentions chan_woomera =D
14:52.08darkskiezwoom what?
14:52.11*** join/#asterisk file[laptop] (~file[lapt@mctnnbsah25-142166093154.nb.aliant.net)
14:52.17SwK[Work]*yawn8
14:52.17*** join/#asterisk pussfeller (~todd@216.223.173.189)
14:52.41coppicei saw someone mention chan_woomera just a few lines ago
14:53.22astoriaWhat is chan_woomera?
14:53.28ManxPowercoppice: I have the same issues with chan_woomera as I do with chan_oh323.  I.e. I feel that the basic design is flawed.
14:53.45ManxPowerastoria: Yet Another Asterisk H323 Driver.
14:53.49astoriacoppice: You said the other day that you have thousands of users using SpanDSP? Doesn't the lack of ECM cause some problems?
14:54.01astoriaManxPower: thanks, i'm a h323 virgin thankfully.
14:54.08anthmooh neet, can I hear how the design is flawed ?
14:54.08*** part/#asterisk QbY (~QbY@adsl-068-209-210-253.sip.cha.bellsouth.net)
14:54.23syle2how do PRI's work for calling numbers, lets say you want to forward a DID to a sip server...do you do exten => 2223334444,1, blah as well as exten => 3334444 , for local calls as well as 12223334444 for long distance calls or does the PRI recognise all calls as one format no matter how it is dialed?
14:54.45ManxPoweranthm: by handleing all the RTP stuff inside the channel instead of letting Asterisk's RTP stack handle it.
14:55.09izoanthm: chan woomera support VAD by any chance ?
14:55.13astoriasyle2: i'm not sure what you're saying exactly. ALl your calls are on zap channels.
14:55.18anthmhmm you ever hear that bit about assume
14:55.48syle2yes but how do you forward just a specific DID?
14:55.54anthmchan_woomera is only 20k compiled as a matter of fact so I bet there is no rtp stack
14:56.07syle2as far as i know the DID can be on any of the 23 channels
14:56.07astoriasyle2: you look at the dnis data ( the called id )
14:56.11ManxPowersyle2: DIDs come into Asterisk as extensions
14:56.17astoriasyle2: your provider will forward the dialed numbers to you
14:56.22coppiceastoria: why should the lack of ECM be a problem. Most FAX machines lack ECM
14:56.23ManxPoweranthm: at 20k it could not even handle H323 by itself.
14:56.31anthmding ding
14:56.59syle2ok then read back what i wrote above
14:57.03anthmchan_woomera is a thin client to an external process where you runn the voip engine in it's own process on any platform you choose
14:57.14anthman another box if you wish
14:57.18izohey coppice i started using your spandsp last week really cool stuff !!!
14:57.31ManxPowersyle2: It all depends on how the telco delivers the dialed number.  3 digits?  4 digits?  10 digits?
14:57.35astoriacoppice: ECM is a nice thing to have, most people I've dealt with have ECM lines.
14:57.43anthmand asterisk only blisfully sees a natice slin audio channel and has no worries about integration
14:57.46astoriacoppice: You have a new spandsp? yay!
14:57.47ManxPoweranthm: And chan_woomera communicated with the rest of Asterisk using SLIN.
14:58.09ManxPowerStill seems rather limited.  More like chan_woomera is really a channel to access external woomera gateways.
14:58.29yaaarhey guys, i'm trying to setup cdr_mysql right now, and when i reload asterisk i get an error saying 'could not connect to mysql database asterisk on localhost.' ...but i can connect and view the tables in that database just fine with the username and password from cdr_mysql.conf.
14:58.38yaaaranybody know what i might have screwed up?
14:58.44syle2manxpower: is it possible they could use more than 1 format?
14:58.55ManxPowerI can see the advantages of pretty much ripping all audio processing out of Asterisk, but I still think its a flawed design
14:58.55syle2for longdistance, local etc
14:58.57*** join/#asterisk [TK]D-Fender (~joe@216.252.67.4)
14:59.09anthmwell since woomera will soon support sip iax and h323 you will soon get 3 protocols to work on 1 tiny simplistic channel driver
14:59.10izoyaar: you either use local or TCP/IP socket
14:59.16coppiceECM shouldn't be that important these days. even without it most FAXes are clean. it was pretty important before the mass deployment of digital exchanges, but just as they got ECM speced that mass deployment ook place :-)
14:59.20ManxPowersyle2: that would be unusual in my experience, but I've only dealt with a telco that ASKS us what format we want the dialed number to arrive in.
14:59.26anthmand btw, I also wrote the skeleton of the objective sys one
14:59.39fearnorecm doesn't help against frame slips, and without frame slips, there's no problem ;)
14:59.49anthmso i am not pluging anything I just want ppl to test it
14:59.51yaaarizo: not sure i follow....
14:59.56astoriacoppice: cool, i've got some confidence in spandsp now. I just went live with a deployment this morning!
15:00.18ManxPowerI still need to "backport" spanDSP to 1.0.x
15:00.21*** join/#asterisk illek (~mike@ip68-227-104-152.ok.ok.cox.net)
15:00.37ManxPowerlast I saw it was a few changes to CLID handleing and that was it.
15:00.51coppiceManxPower: a man too mean to even *know* about non-free codecs :-)
15:00.52ManxPoweranthm: so what would be the point of using Asterisk then?
15:00.53izoyaaar: you can connect to mysql on two ways via local socket /var/run/mysqld/mysqld.sock or port 3306 they both hmight have different permissions
15:01.11ManxPowercoppice: Yeah!  Viva la Free Codecs!
15:01.12coppiceManxPower: why would spandsp need backporting?
15:01.28ManxPowercoppice: CVS-HEAD's callerid structures are slightly different than 1.0.x's
15:01.29*** join/#asterisk denisgalvao (~Denis@200.146.0.254)
15:01.34anthmfor all the asteriskness
15:01.37denisgalvaoHi all.
15:01.50izoManxPower: i belive copppice has two versions on his FTP
15:01.54denisgalvaoI need some help on Asterisk + Unicall....
15:01.58izoManxPower: for both 1.0.x and 1.1x
15:02.02anthmproviding native interface to voip protocols is not asterisk's strongest offering
15:02.11ManxPowercoppice: I of course mean "SpanDSP+rxfax+txfax" when I say "SpanDSP".
15:02.16yaaarizo: it gives me the same error either way.
15:02.21fitzelAnyone here with some practical experience for a softphone on a PDA with wifi/wlan in every-day use?
15:02.40izoyaaar : well then you have something wrong with your mysql setup
15:03.02ManxPowerizo: *nod*  The old "not compatable with Brother or Cannon fax machines" version for 1.0.x and the "works with prectically all fax machines" for CVS-HEAD. 8-)  Unless he released a 1.0.x version with his updates in the past couple of months.
15:03.25izoyaaar : rty connecting frmo command line like mysql -u user -h localhost -p database
15:03.33anthmwith the example of h323, most channel drivers based on openh323 have issues cos the threading model is not compatable so by letting the h323 run in it's own process the way it wants it makes compatability easier
15:03.34izoManxPower : really ?
15:03.38ManxPowerHmm...I'll bet I can use the new SpanDSP with the old rxfax/txfax....
15:03.47yaaarizo: yeah like i said connecting with mysql -u asteriskuser -p works just fine
15:03.49*** join/#asterisk montag___ (~montag@host187-252.pool8175.interbusiness.it)
15:03.49ManxPowerizo: what version are you running with 1.0.9?
15:04.01izoyaaar : notice -h parameter for localhost
15:04.15izoManxPower yep
15:04.24*** join/#asterisk joerg (~joerg@cl-666.ham-01.de.sixxs.net)
15:04.46montag___hi, when a call a sip extension with a .call file asterisk don't wait that remote extensions answer, but forward immediately the call...only with sip....with IAX or ZAP all it's working....any tip ?
15:04.59coppiceManxPower: what's wrong with the latest rxfax and txfax?
15:05.12*** join/#asterisk kshumard (~kenny_@207.111.174.1)
15:05.19yaaarizo: yeah, still works fine with 'mysql -u asteriskuser -h localhost -p asteriskcdrdb'
15:05.20ManxPowercoppice: give me 10 mins and I'll tell you.  What's the URL of a working location to get the stuff?
15:05.59ManxPowercoppice: I've not bothered to report it since I pretty much get shot down anytime I report any problem with 1.0.x
15:06.03ManxPowerto ANYONE.
15:06.45izoyaaar : how do you specify mysql database in cdr_mysql.conf ?
15:06.57anthmthe whole notion of 1.0.x to begin with is a problem
15:07.02coppiceI only use 1.0.x. support for 1.1.x in my software is a bit weak. they kept breaking things, so I decided to wait until the dust settles. 1.2.5 sounds a likely time for that :-)
15:07.08yaaarizo: dbname=asteriskcdrdb
15:07.09ManxPowerSee what I mean?
15:07.27izoyaaar : enter whoe thing into pastebin.com
15:07.34yaaark
15:07.51ManxPowercoppice: you restructured your site!
15:07.59coppiceno
15:08.17ManxPowercoppice: last time I looked there was NO 1.0.x directory for rxfax/txfax
15:08.47denisgalvaocoppice: Could you help on an Unicall related problem?
15:08.50*** join/#asterisk _DAW (daw@67.128.57.2)
15:08.56yaaarizo: http://pastebin.com/320288
15:09.01*** join/#asterisk tzafrir (~tzafrir@local.xorcom.com)
15:09.14coppicedrwxr-sr-x  2 root root    4096 Jun 12 21:29 asterisk-1.0.x
15:09.16coppicedrwxr-sr-x  2 root root    4096 Jun 12 21:32 asterisk-1.1.x
15:09.18coppice-rw-r--r--  1 root root    1445 May 11 22:31 README
15:09.19coppice-rw-rw-r--  1 root root 1288267 May 11 22:25 spandsp-0.0.2pre18.tar.gz
15:09.20ManxPowercoppice: But CVS-HEAD is perfect!  It works!  It's more stable than 1.0.x!
15:09.34yaaarizo: hehe
15:09.38yaaarizo: never mind
15:09.52coppiceso why are you interested in 1.0.x?
15:09.52astoriacoppice: where can I obtain these spandsp+rxfax+txfax changes you've made?
15:09.52izoyeah
15:10.01ManxPowerftp://ftp.soft-switch.org/pub/spandsp/spandsp-0.0.2pre18/
15:10.04astoriaThanks!
15:10.11yaaarizo: only when i typed it into the pastebin did i go 'gee, wonder if those single quotes really belong there'
15:10.23izoyaaar :-)
15:10.44ManxPowercoppice: I run 1.0.x
15:10.56ManxPowerI'm repeating the drivel everyone else seems to be saying about CVS-HEAD
15:11.02coppiceI think the 1.0.x and 1.1.x stuff for fax is OK. There are some issues with the 1.1.x stuff for unicall
15:11.16*** join/#asterisk |dennis| (~dennis@200.32.215.82)
15:11.22anthmasterisk is way too unstable as a whole to be developing sects towards imaginary version neumbers
15:11.41denisgalvaocoppice: Yes. Im runnig 1.1.x rxfax and txfax without prpoblem, but Unicall...
15:11.47bkw_ManxPower, drivel?
15:12.15bkw_why 1.0.x days are numbered in the double to single digits
15:12.19ManxPowerThe award says "Against all odds, still a friend of 1.0.x"
15:12.20izoManxPower : whats the award/prize  meadl or something or just handshake ?
15:13.03ManxPowerizo: a small sculpture made from old Digium USB FXS adapters.
15:13.17bkw_CVS-Stable is not the answer!
15:13.40ManxPowerEveryone knows what MY idea of "the answer" is.
15:13.43anthmit's really rather silly to argue about the code from the outside perspective =D
15:14.16Kattyless arguing, more hugging.
15:14.19anthmespecially since it's the same code to start with
15:14.28MikeJ[Laptop]sigh...
15:14.32coppicewell if asterisk of any sort is the answer, it was a bloody strange question
15:14.41ManxPowercoppice: LOL!
15:14.44MikeJ[Laptop]bitch less, work more...
15:14.47MikeJ[Laptop]:P
15:15.01ManxPowercoppice: keep the award, it's already a collectors item.
15:15.18bkw_ManxPower, you the award fairy?
15:15.30anthmthe number of bytes typed bitching about stuff probably shadows the asterisk code base on a daily basis
15:15.41izo:-)
15:15.41bkw_hahahahahahahahah
15:16.12coppicewell, there are far more people moaning than actually capable of doing anything useful
15:16.26izothat is actually true
15:16.34bkw_ya think?
15:16.43*** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc)
15:17.03izoand there is plenty ppl doing stuff for commercial use not bothering to participate
15:17.04anthmimagine if everyone typed thier rants into articles and migrated them into docs
15:17.54MikeJ[Laptop]heh.. anthm, yeah right.. like thats going to happen
15:18.03*** join/#asterisk ArkyLady (ArkyLady@adsl-67-64-6-10.dsl.ltrkar.swbell.net)
15:18.12Kattyanthm: i'd explode.
15:18.42Hmmhesaysheh, sounds like fun
15:18.53coppice* needs VAD pretty badly. I wish I had the time to do the DSP bit, then I'm sure Steve Kann would get the rest done.
15:18.55MikeJ[Laptop]bitch less, work more... that means you ManxPower, you need a cvshead test box...
15:19.16anthmthe truth is, most people who get to stable follow the same path of "I wish I could ____"
15:19.27*** join/#asterisk switch (~switch@61.206.115.5)
15:19.36Kattytwisted[asteria]: beep beep?
15:19.39ManxPowerMikeJ[Laptop]: I need many things.
15:19.46twisted[asteria]Katty, meh... let me wake up
15:19.50anthmand that path has already been carved by the ppl who went and added that same functionality
15:19.53Kattytwisted[asteria]: oh, i wasn't going to
15:19.57twisted[asteria]Katty, oh ;P
15:19.58Kattytwisted[asteria]: was just saying hi (=
15:19.59anthmhence CVS head
15:20.01twisted[asteria]Katty, hi :)
15:20.06ManxPowerMaybe I'll set up two CVs-HEAD boxes when I get my two TE110Ps in for testing.
15:20.09*** join/#asterisk fugitivo (~ajf@201.255.104.144)
15:20.10fugitivohello
15:20.18*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@bzq-80-62-239.red.bezeqint.net)
15:20.24astoriaManxPower: bling bling
15:20.31anthmso when a guy says "I want foo" and someone says get CVS HEAD it
15:20.41ManxPoweranthm: I do that sometimes.
15:20.43PoWeRKiLLHi
15:20.50anthmis because someone already added that and it's the only way to get it
15:20.59PoWeRKiLLSomeone have an idea how to log cdr to 2 mysql server ?
15:21.00anthmunless you backport
15:21.14ManxPowerI've paid for at least 1 packport 8-)
15:21.19twisted[asteria]PoWeRKiLL, uhm, that's an oldie, and has quite a few steps involved... i reccommend the wiki
15:21.26Hmmhesayswiki wiki asterisk-addons wiki wiki
15:21.29anthmand if you backport everything you like in HEAD to stable you get a clone of HEAD
15:21.33*** join/#asterisk jsharp (~jsharp@65.88.255.132)
15:21.36twisted[asteria]anthm, yup
15:21.49twisted[asteria]anthm, how you doin btw?
15:21.51ManxPowerExcept it would not change every few days.
15:21.57anthmwhich proves that stable is indeed not the answer
15:22.01twisted[asteria]ManxPower, s/days/hours
15:22.07anthmthat is my theorm
15:22.15anthmtwisted, not bad you?
15:22.29jsharpOdd SIP problem.  When I call into Asterisk 1.0.7 with an ATA-186, I get a fast busy on the ATA and this on the Asterisk console:  chan_sip.c:7192 handle_request: Ignoring too old packet packet 1 (expecting >= 2)
15:22.46twisted[asteria]anthm, not bad, staying busy
15:22.51ManxPowerjsharp: any reason you are not using 1.0.9?
15:23.01jsharpOther than I haven't downloaded and installed it, no.
15:23.08twisted[asteria]PoWeRKiLL, DO NOT /msg W/o ASKING. that's bad form. I gave you what help i'm going to give
15:23.13twisted[asteria]PoWeRKiLL, you /msg me again, i will kick you.
15:23.22ManxPowerjsharp: I don't recall any specific change that woule fix that, but.....
15:23.33anthmtwisted, did you see app_rss I thought you'd like it being a cepstral fan
15:23.45leheltcpdump -i any port 4569 < this is correct?
15:23.51jsharpI am running it across a 600ms satellite link, so I figure that probably has something to do with it.
15:23.55*** join/#asterisk _omer (dfsdf@203.215.180.254)
15:23.55twisted[asteria]anthm, no, i have not yet... is it on pbxfreeware?
15:24.02PoWeRKiLLtwisted sorry
15:24.13ManxPoweranthm: I do admit to drooling over AEL. 8-)
15:24.23tzafrirlehel, does that dump both udp and tcp?
15:24.25twisted[asteria]PoWeRKiLL, s'ok.  that's why I warned rather than immediately taking action
15:24.32anthmyah
15:24.37sudhir492anyone using x-lite with asterisk
15:24.39coppiceManxPower: I drool over head :-)
15:24.51leheltzafrir: yes.. both, but i got no response
15:24.57tzafrirlehel, I also tend to add '-n' to avoid delays from name resolution
15:24.58lehelit's strange
15:25.13anthmwell it appears 1.2 is going to be done which is more like 1.0 if you ask me cos the 1.0 was not really ready
15:25.15ManxPowercoppice: realtime would be nice, the new jitterbuffer would be nice, auto priorities would be nice, etc, but the thing that is REALLY nice is AEL
15:25.20PoWeRKiLLI find a way to do it using cdr_mysql and odbc but in this install i can't use odbc
15:25.28anthmand it will actually be able to keep you busy for a while
15:25.34lehellast friday worked well the connection between.. and today.. it;s not working the IAX2
15:25.39anthmbut the same thing is gonna happen eventualyl
15:26.01sudhir492I am not able to get that work. In particular, what should I set in domain/realm?
15:26.09coppiceManxPpower: the jitterbuffer is what most people will notice, and the other thing mahy people need badly is VAD/DTX
15:26.16tzafrirlehel, not working=?
15:26.20lehelit is no difference ... no response. could be firewall problem?
15:26.43lehelnot- working = doesn't get (ping) response
15:26.46tzafririf it were a firewall on your system, you'd still see packets with tcpdump
15:27.01PoWeRKiLLis there another way ?
15:27.39tzafrirlehel, a lower level test is using netcat: echo test |nc -u asterisk_hostname 4569
15:28.21lters_PoWeRKiLL, write a script that watches your local cdr file and sends the records anywhere u want it to.
15:28.48*** join/#asterisk santiago (~santiago@63.245.86.188)
15:28.55izoManxPower: what is AEL ?
15:29.01PoWeRKiLLlters_ I want it to be via mysql
15:29.10leheltzafrir: forward host lookup failed: Host name lookup failure : Resource temporarily unavailable
15:29.13ManxPowercoppice: on the links I care about, there really isn't much jitter.  i.e. Point-to-Point T-1 w/QoS or local LAN.
15:29.15PoWeRKiLLAEL = Asterisk Entreprise Linux :D
15:29.24izo:-))
15:29.31izowhat about Asterisk Business Edition ?
15:29.32coppiceManxPower: so selfish :-)
15:29.35ManxPowerizo: it's a real programming language for extensions.conf.  It's included with CVS-HEAD.
15:29.49coppiceAsterisk End Life
15:29.56izoManxPower : really ? damn how come I didnt notice
15:30.06izoManxPower : what about overhead ?
15:30.09ManxPowerizo: because you were not reading the asterisk-cvs mailing list.
15:30.11_omerHow many calls at a time a pentium 4 machine with 1 GB ram could afford??????
15:30.20izoManxPower: i am but not all the changes :-P
15:30.22ManxPowerizo: all it does is translate the "code" into regular extensions.conf stuff
15:30.49ManxPowerthere is also a javascript and perl extensions.conf support on pbxfreeware
15:30.54_omerHow many calls (SIP - No Hardware) at a time a pentium 4 machine with 1 GB ram could afford??????
15:31.00lters_PoWeRKiLL, that is fine, make your script do mysql
15:31.44Qwell_omer: repeating your question over and over isn't the best way to get help
15:31.59brendaDoes anyone know how long ast_play_and_wait() waits?
15:32.02InfraRed_omer: check the wiki , under dimentioning server
15:32.27izoManxPower:damn i'm really backwards with this stuff
15:33.19PoWeRKiLLlters_ what do you mean what master.csv and do mysql ?
15:34.21anthmbrenda, probably till the file is done playing
15:34.30*** join/#asterisk gniretar (~mark@198.173.197.15)
15:34.45gniretarhe guys
15:34.53Qwellshe girls
15:35.04gniretari'm having a little trouble with my Aastra SIP phones
15:35.17gniretarthey arnt storing Caller ID info.
15:35.21gniretarand girls ;-)
15:35.52anthmbrenda, according to the source, it would be till you hanngup, press any digit, or the file is done playing
15:35.55gniretari need to be able to flip through previous calls and i cant. Does anyone know of something on the Asterisk side that would help me with this?
15:36.25brendaanthm: interesting... maybe I should add a timeout
15:37.02anthmbrenda, what's the goal ?
15:37.17elduis there a way to force a specific codec per telco number within the same sip trunk ?
15:37.58brendaanthm: background a file, wait X seconds
15:38.27anthmthen cancel the background ?
15:39.14anthmwith any intention of collection digits ?
15:39.27anthmcollecting digits
15:39.31brendathe timeout doesn't start until the file is done
15:39.43brendayes collect digits
15:40.00anthminside your app right ?
15:40.01brendaunless of course the timeout happens
15:40.03brendayeah
15:40.04anthmC code?
15:40.09brendayup
15:40.32anthmseconds or microsec ?
15:40.37brendaeither is fine
15:40.40*** join/#asterisk jimmybob46 (~jim@81.5.154.235)
15:40.42anthmor millisec
15:40.50Qwellpicoseconds
15:40.58jr352kanthm: are you collecting digits in the dial plan? or using agi?
15:41.01jimmybob46hello all
15:41.06gniretarhi
15:41.13anthm? when
15:41.33jimmybob46is this the main asterisk help channel? or have i gone wrong?
15:41.57brendaI'm the one collecting digits
15:42.18jr352kohh! branda! then the question is for you
15:42.21Qwellbrenda: I have a 6 you can add to your collection
15:42.23brendalol yeah
15:42.24tzafrirjimmybob46, it is. About you going wrong? I have no idea
15:42.51jimmybob46lol
15:42.53jimmybob46nice one
15:43.07jimmybob46thanks tzafrir
15:43.19brendaI'm playing my hand with apps so I can actually help with the asterisk code too
15:43.55jimmybob46I got a capi problem, I am behind a pbx, and have to dial 9 to get an isdn line
15:44.33jimmybob46I cant work out where to do this.. :(
15:45.13*** join/#asterisk akrall (user@201.128.92.42)
15:45.40tzafrirjimmybob46, where exactly is your problem? can you make calls inside the PBX?
15:46.06akrallWhat kind of phones are the panasonic KX-T7730  and  KX-T7030X  ones? digital or plain old analog? will they work on asterisk (of course, using an ATA)?
15:46.20tzafrirdoes the isdn/capi/whatever channel work?
15:46.32jimmybob46my asterisk pbx is fine, chan_capi is working fcpci is loaded, but i get Reason 0x3481
15:46.47jimmybob46how can i determine if the capi channel works?
15:48.27jimmybob46i can sip call inside the pbx to other sip numbers, its dialing out over capi that does not go anywhere
15:49.41jimmybob46I apologise for my ignorance, I have spent 5 days trying to figure this out.
15:49.53*** join/#asterisk _gigi_ (gigi@jabber.szczecin.pl)
15:50.04_gigi_Hello.
15:50.57jsharpWell, foo.  Upgrading to 1.0.9 didn't solve my SIP problem.
15:51.30_gigi_im looking for some software for analize RTP streams (delay, jitter, delta, and loss), someone know something ? :)
15:52.56lters_PoWeRKiLL, are u wanting to send the cdr's to 2 servers?
15:53.42Darwin35hmm
15:53.48Darwin35what phone to get
15:54.16InfraRedcisco 7960
15:55.30Darwin35have a office full of those
15:55.37Darwin35this is for home use
15:55.38*** join/#asterisk fugitivo (~ajf@168-226-245-129.mrse.com.ar)
15:55.44Darwin35x401 I think
15:56.08*** join/#asterisk Gunnar (~gunnar@62.97.243.70)
15:56.53*** join/#asterisk Gunnar (~gunnar@62.97.243.70)
15:57.00jimmybob46I am in the UK, is there anyone that knows about any capi specific problems? BT usually cause a few problems..
15:57.08gordonjcpBT always causes problems
15:57.16gordonjcpeven if you're not doing anything complicated
15:57.17anthmbrenda, off the top of my head, something like this http://66.250.68.190/eg/play_timed.c
15:57.23anthmdidnt try it tho
15:57.50puzzledjimmybob46: does capiinfo output a lot of info indicating that your card is active?
15:58.21jimmybob46capi info says lots
15:58.51puzzledjimmybob46: ok, and have you correctly loaded chan_capi.so in /etc/asterisk/modules.conf?
15:59.03*** join/#asterisk BoNaDs (~theplot@ool-44c53748.dyn.optonline.net)
15:59.17BoNaDsdoes "echotraining" when enabled, determine and override the setting for "echocancel"?
15:59.40*** join/#asterisk Andrezo (~www@217.129.208.124)
16:00.04jimmybob46Yes, I have made sure that I have that in my modukles. asterisk -vvvvvvgc shows me lots of info about capi loading
16:01.21Darwin35ok cool
16:01.22puzzledjimmybob46: ok, did you configure /etc/asterisk/capi.conf?
16:01.42Darwin35the x401 is the netweb 401 renamed and they are going to replace my broken oone
16:01.52jimmybob46I have configured capi.conf yes.
16:01.58HmmhesaysI love it when people call wanting general knowledge on voip
16:02.38puzzledjimmybob46: if all that is ok then I guess it must be something in the dialplan. are you using chan_capi-cm-0.5.4?
16:02.44_gigi_im looking for some software for analize RTP streams (delay, jitter, delta, and loss), someone know something ? :)
16:03.04jimmybob46I am using the chan_capi from sourceforge (o.54?)
16:03.19puzzledyes latest is 0.5.4
16:04.26*** join/#asterisk ArkyLady (ArkyLady@adsl-67-64-6-10.dsl.ltrkar.swbell.net)
16:04.41jimmybob46I would paste my extensions.conf in here, but dont want to upset anyone :)
16:04.50ManxPower~pastebin
16:04.50jbotrumour has it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
16:05.10ManxPower~mailinglist
16:05.10jbot[mailinglist] Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
16:05.11ManxPower~docs
16:05.11jboti guess docs is Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
16:05.29*** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || Asterisk 1.2 Janitors Needed! http://dev.asteriskdocs.org || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com ||Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - Speakers wanted
16:05.48*** topic/#asterisk by drumkilla -> Asterisk: The Open Source PBX || Asterisk 1.2 Janitors Needed! http://dev.asteriskdocs.org || Cluecon -- PBX Developers Conference in Chicago Aug 3-5 http://www.cluecon.com || Astricon 2005 in Anaheim Oct 12-14 http://www.astricon.net/2005/ - Speakers wanted
16:05.50*** join/#asterisk Gunnar (~gunnar@62.97.243.70)
16:05.52akrallWhat kind of phones are the panasonic KX-T7730  and  KX-T7030X  ones? digital or plain old analog? will they work on asterisk (of course, using an ATA)?
16:06.54ManxPowerakrall: plug one of the phone into an analog POTS line.  If you can use the phone then it will work with Asterisk
16:07.09Qwellif it starts smoking, you probably can't
16:07.25syle2do you use codecs on zap channels?
16:07.26akralljajaja not very scientific :)
16:07.43Qwellakrall: That is a very scientific method
16:08.40_omerdoes Asterisk Support Dual Processors??? fully utilize both processors???
16:08.56akrallQwell: :)
16:09.01*** join/#asterisk diegodi (~diegoluig@host-84-222-36-3.cust-adsl.tiscali.it)
16:09.04Qwell_omer: sure
16:09.23_omerQwell: sure for both questions? :)
16:09.37Qwellno to the first, yes to the second
16:09.53jimmybob46well, please could u check this simple config : http://pastebin.ca/18457
16:10.02*** join/#asterisk sloPP (~slepp@S0106000f663692da.ed.shawcable.net)
16:10.07diegodiHi all!!
16:10.14jsharpBuh.
16:10.25sloPPanyone know the difference between National 1 and National 2?
16:10.36essobiabout 10 dollars
16:10.45_omerQwell : :D
16:12.31jimmybob46what do u reckon puzzled?
16:15.13*** join/#asterisk agave-txlink (phanop@216.81.47.201)
16:15.21*** part/#asterisk akrall (user@201.128.92.42)
16:15.30agave-txlinkis it possible to do caller ID matching in realtime?  ex: exten => _NXXNXXXXXX/9725551212 ?
16:17.17*** join/#asterisk outtolunc (outtolunc@adsl-66-218-53-170.dslextreme.com)
16:17.21diegodisomeone can explain me how to modify the stripping 2 to 1 in the command {EXTEN:2}
16:17.25*** join/#asterisk nitram (foo@superblob.com)
16:17.54Qwelldiegodi: by replacing the 2 with a 1?
16:18.19diegodiyes, every time I save the configuration asterisk make this to 2!!!
16:18.28Qwellasterisk doesn't change your configs
16:18.39diegodiso...?
16:18.47Qwellso figure out what is
16:19.18*** join/#asterisk Beave (~beave@vistech.org)
16:19.39diegodiI don't know, I'm using asterisk@home and the command CAPI/XXXXXXB$OUTNAME}
16:19.43*** join/#asterisk joshpbx (~joshpbx@83.27.103.159)
16:19.47QwellThere you go.
16:19.52Qwellso amp is probably changing it on you
16:19.53*** join/#asterisk SteveL (~stephen@smtp.burlesonisd.net)
16:19.56QwellThats why we don't use amp. ;]
16:20.35diegodithanks very much!!!
16:23.08puzzledjimmybob46: let me have a look
16:23.14SteveLWhat do I need to do to access cvs HEAD?  I keep getting this error: cvs login: authorization failed: server cvs.digium.com rejected access to /usr/cvsroot for user anoncvs
16:23.40SteveLI'm using anoncvs for password.
16:23.56Qwellpassword is blank, right?  I forget, its been a while
16:24.10DarthClueanoncvs is the correct pw...let me try it
16:24.13Qwelloh
16:24.23SteveLblank password won't work either
16:24.45*** part/#asterisk diegodi (~diegoluig@host-84-222-36-3.cust-adsl.tiscali.it)
16:25.01MikeJ[Laptop]SteveL, I just tried and it is working.
16:25.03*** join/#asterisk grimse (~grimse@p5481D4AE.dip.t-dialin.net)
16:25.16MikeJ[Laptop]tripple check your CVSROOT
16:25.28MikeJ[Laptop]and make sure your not screwing up the pwd
16:25.58puzzledjimmybob46: the only difference is that I use "|" so e.g. exten => _00XXXXXXXXX,2,Dial(CAPI/contr1/b${EXTEN:1}|45|r)
16:26.14DarthClueworks here too.  follow the directions at http://www.darthclue.org/categories/3-Chalkboard-Examples
16:26.56SteveLSTRANGE
16:26.59*** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net)
16:27.06SteveLit's working now doing the exact same thing I was doing before
16:27.42DarthClueSteveL: sometimes it just takes a little evil to make things work.
16:27.51SteveLhaha
16:27.51SteveLtrue
16:28.07*** join/#asterisk Dave-- (~a@choloconsultancy.plus.com)
16:28.32Dave--evening
16:29.26Dave--dont suppose theres anyone awake that can tell me why....
16:29.26Dave--exten => _0.,1,Dial(Zap/1/${EXTEN},10)
16:29.26Dave--exten => _0.,2,NoOp(HANGUPCAUSE is ${HANGUPCAUSE})
16:29.36Dave--the second line is never got to
16:29.50QwellDave--: it probably jumps to 101 on failure
16:29.57Qwellor, 102?  I forget.
16:30.15florzDave--: Or to h upon hangup.
16:30.21Dave--lol
16:30.50Dave--so what should my next line be to catch it?
16:31.05MikeJ[Laptop]Dave--. what version?
16:31.09florzDave--: what exactly do you wanna catch?
16:31.27ManxPowerDial will jump to n+1 on busy if there is no n+101
16:31.37Dave--Asterisk 1.0.9-BRIstuffed-0.2.0-RC8h
16:31.52Dave--i need to catch hangup on busy
16:31.58Dave--so i can redial
16:32.15MikeJ[Laptop]see ManxPower's comment
16:32.46Dave--therefore it should be getting to the second line, correct?
16:34.25*** join/#asterisk DagMoller (~DagMoller@4b83b8828b0c682f.session.tor)
16:35.30DagMollerHave iaxtel problems? i'm from brasil, and canot connect to iaxtel
16:35.44brendaanthm: wow thanks!
16:36.47*** join/#asterisk SuPrSluG (~SuPrSluG@pool-129-44-144-219.buff.east.verizon.net)
16:36.55florzDave--: I'd say so, yeah.
16:37.34Dave--thanks
16:38.34florzDave--: So, what was it?
16:38.40Dave--ive no idea
16:38.51florzBut it works no?
16:38.52Dave--i cant get it to go anywhere beyond
16:38.52florz+w
16:38.52BoNaDsanyone here used fxotune?
16:38.53Darwin35man hop-op has made a sip/wifi phone priced at 39.95 but are unwilling to sell to the general public
16:39.08Darwin35they only want to sell to providers of voip service
16:39.16Dave--would they be willing to see to a call centre?
16:39.26Darwin35called the hop-on 1515
16:39.27*** join/#asterisk Goshen (~Goshen@67-40-107-29.slkc.qwest.net)
16:39.48InfraRedhttp://www.mosnews.com/news/2005/07/25/spammerdead.shtml
16:39.50InfraRedcoo
16:39.52Darwin35http://www.wifi-cell.com/
16:40.42Darwin35it just pisses me off all these companies now making good products for voip but unwilling to sell to the general user
16:40.59*** join/#asterisk Cresl1n (~Cresl1n@207.111.174.1)
16:41.10Darwin35and in nothing less then a bulk of 1000 units at a time to caompanies
16:41.11*** join/#asterisk Derkommissar (~alberto@66.64.215.6.nw.nuvox.net)
16:42.04brendaDarwin35: supporting end users is expensive
16:42.25Dave--florz - any idea why it might not be getting there?
16:42.38ManxPowerWe have way cool product, but you can't actually buy it!
16:42.58Qwellvaporware?
16:42.59Darwin35heck I would offer to do support for the units
16:43.05mkrufkyHow come channel topic says "Asterisk 1.2 Janitors needed" ... aren't we up to 1.09 or 1.10 or something
16:43.19Qwellmkrufky: 1.2 > 1.0.x
16:43.20mkrufky(i'm still using 1.07 on my server)
16:43.29*** join/#asterisk mjmac (~mjmac@mjmac.active.supporter.pdpc)
16:43.31mkrufkyaha... when did 1.2 come out?
16:43.40Darwin35it has not yet
16:43.52Darwin35there is a planned code freeze
16:43.53mkrufkyoh, okay
16:43.58Darwin35then a move to 1.2
16:44.03florzDave--: Nope, no clue.
16:44.07mkrufkyah, thats good news
16:44.27florzDave--: It's a BRI line?
16:44.37Dave--BRI line?
16:44.43file[desk]Cresl1n: Mattttttt
16:44.48florzDave--: ISDN
16:44.55Dave--nope, analogue line
16:45.04*** join/#asterisk ronn (ronn@host217-46-199-164.in-addr.btopenworld.com)
16:45.11Dave--thru a Wildcard X100P
16:45.31florzDave--: Ah, OK, maybe that's causing that the busy condition isn't correctly recognized?
16:45.52Dave--well i was thinking that
16:45.58Darwin35I just want to see usable products on the market
16:46.11florzDave--: I've got no experience with analog lines and asterisk, but it's quite likely to cause problems, I guess :-)
16:46.12Dave--and ive been thru all the gubbins in indicators.conf
16:46.34Dave--im wondering if BT's "You can press 5 to use ringback" over the top of the engaged tone
16:46.43Dave--might be fucking with the signalling
16:47.26Darwin35whoi is it that owns atcomm
16:47.31Darwin35I forget his name
16:48.55*** join/#asterisk coppice (~pocketirc@30.195.17.210.dyn.pacific.net.hk)
16:50.03*** join/#asterisk tarvid (~tarvid@68-67-192-202.chvlva.adelphia.net)
16:53.56*** join/#asterisk marc_in_lux (~gergesm@cable-83.217.135.132.coditel.net)
16:54.09*** join/#asterisk mut (~animenodv@65.111.201.79)
16:54.13marc_in_luxgood evening
16:54.17tarvidlooking for an 800 provider
16:54.38tarvidearly eveing in luxembourg?
16:54.41mutlocal telco?
16:54.47Darwin35astralink
16:54.50marc_in_luxtarvid, 7 pm... yes :-)
16:55.03*** join/#asterisk zoa (~k@213.91.216.136)
16:55.10*** part/#asterisk DagMoller (~DagMoller@4b83b8828b0c682f.session.tor)
16:55.15marc_in_luxlooking for a beginners introduction into dialplans. Replicating those festures found in SOHO PBX's.
16:55.28marc_in_luxlike transferring calls, parking them, picking up from another location etc...
16:55.58DarthCluemarc_in_lux: start with the wiki.
16:56.02emrahmarc_in_lux: What would you like to do?
16:56.08*** join/#asterisk loick (~loick@APuteaux-151-1-49-217.w82-124.abo.wanadoo.fr)
16:56.30emrahDo you want to be able to receive a call to a given extension, park it and take it in another station?
16:56.38marc_in_luxemrah, for example
16:56.53marc_in_luxor, doing a 3-way conference with two internal sip phones and an external call
16:57.07emrahOk...
16:57.17marc_in_luxor, routing all external call to one extension, but allowing others to pick up those calls on request
16:57.30Darwin35read the wiki
16:57.34emrahLet me talk with you in a private window without disturbing here
16:57.38Darwin35the wiki has all your answers
16:57.52Qwellemrah: best to answer these questions in public, so everybody benefits
16:57.53Darwin35the wiki is the book of the goods
16:57.54marc_in_luxDarwin35, it sure may. But you're new in telephony, it's hard to digest.
16:58.15emrahok
16:58.16DarthCluemarc_in_lux: start with the wiki, you'll be much happier later.
16:58.33Darwin35then sit your butt down and write a sterisk-for-dummies book
16:58.46emrahmarc_in_lux: It's good to read the wiki. It's perfectly explained. www.voip-info.org
16:58.53marc_in_luxDarwin35, DarthClue - happy to document where I'm getting.
16:58.58QwellI don't know about perfectly... ;]
16:59.09emrahYou have also a good example files.
16:59.18emrahJust make them with the appropriate cmd
16:59.24marc_in_luxI'll restudy the wiki.
16:59.27[TK]D-FenderDarthClue : We as a community really should make a much "fuller" sample dialplan with comments to give people as a place to start for understanding how things work.  Going right to the Wiki to break apart every command can be frustrating to newbs...
16:59.45marc_in_lux[TK]D-Fender, thanks :-)
16:59.50astoriaIt really does help to have a working extensions.conf to work from.
17:00.00astoriaBut once you pick it up, the wiki organization is purrrfect
17:00.14marc_in_luxthe problem for newbies in telecoms is that most concepts are not really understood. I can use my little home pbx.
17:00.21[TK]D-Fenderastoria : I agree, but we didn't do that groundwork yet.
17:00.24Darwin35I have posted a few on pastebin.ca
17:00.29lathos42wiki good
17:00.31Darwin35goo lok for darwin35
17:00.36astoriaYeah, thats true. I am working on a degree in telecom, so I guess it was easier.
17:00.40marc_in_luxbut I have a tough time following how a 3 way conf. should go in extensions.conf
17:00.41Darwin35its got loads of extensions
17:00.45[TK]D-Fendermarc_in_lux : Feel frr to PM me for the time being.
17:01.04Qwellmarc_in_lux: meetme
17:01.06marc_in_lux[TK]D-Fender, thanks.
17:01.06[TK]D-Fenderfree*
17:01.12Darwin35but I an still adding functions as I can
17:01.27*** join/#asterisk brettnem (~brettnem@207.90.232.34)
17:01.51*** join/#asterisk denisgalvao (~Denis@200.146.0.254)
17:02.01[TK]D-Fendermarc_in_lux : 3-way calling is often nothing more that being able to accept another call while on one already and using the IP phones "conference feature".  That isn't even in the dialplan.
17:03.30marc_in_luxDarwin35, found lots of stuff from you. will now go and study that
17:04.00denisgalvaocoppice: How may I get the unicall logs? They will go to the same files of Asterisk logs?
17:04.14coppiceyes
17:04.48denisgalvaocoppice: Could you help on a problem?
17:04.59coppiceok
17:05.33denisgalvaocoppice: Jul 25 09:50:07 WARNING[3200] channel.c: Exception flag set on 'UniCall/13-1', but no exception handler
17:06.14denisgalvaocoppice: Im running FC2 with * HEAD and unicall0.0.3pre3
17:06.34denisgalvaocoppice: CPU 100%
17:06.40coppicei saw your email about that   i have to check
17:06.55denisgalvaocoppice: Hmmm.. Ok.
17:07.39*** join/#asterisk xcore (~xcore@200.175.93.58.tbprof.gvt.net.br)
17:07.42denisgalvaocoppice: I can test your unicall with a lot of E1 PABX.
17:08.28xcorepeople, anybody here know how to eliminate the echo on asterisk+IAX... i have echo when i speak, i listen my voice
17:09.17*** join/#asterisk denisgalvao (~Denis@200.146.0.254)
17:09.32denisgalvaocoppice: Sorry... I lost my conn.
17:10.35denisgalvaocoppice: I was writing that Im on a telephony company, so we have a lot of equipments to test it out.
17:11.29*** join/#asterisk craziman2 (~donnie@boromir.apid.com)
17:12.11denisgalvaoHow may get mor information(logs) from UNICALL?
17:12.31craziman2does anyone know why I can't get anything when I do a cvs checkout -r v1-0_stable zaptel
17:14.16*** join/#asterisk craziman2 (~donnie@boromir.apid.com)
17:16.15*** join/#asterisk craziman2 (~donnie@63.238.52.21)
17:16.17coppiceadd the log control statement to unical.conf
17:16.35*** part/#asterisk craziman2 (~donnie@63.238.52.21)
17:16.45denisgalvaologlevel=255 is the higher level?
17:17.14coppiceyes
17:17.20ronnhi guys.. i'm lookin for polish and hungarian DIDs
17:17.29anthmbrenda, did that code work?
17:17.40denisgalvaocoppice: Ok.
17:17.41Qwellanthm: fn != filename
17:17.41Qwellp
17:17.43*** join/#asterisk craziman2 (~donnie@63.238.52.21)
17:17.44Qwell:p rather
17:18.24anthmdoh!
17:18.27denisgalvaocoppice: Why I can call from PABX to the Asterisk but I cannot call from Asterisk to PABX?
17:18.34craziman2Sorry to ask again.. but I dc'ed... any one have any idea why I can't pull anything with, "cvs checkout -r v1-0_stable zaptel"
17:18.51denisgalvaocoppice: From Asterisk to PABX I got the 100% CPU usage.
17:19.09joshpbxcraziman2: u cant download nighly snapshots?
17:19.50denisgalvaocraziman2: try it: cvs checkout -r v1-0
17:19.59craziman2Trying to setup a semi production system... so I figured the stable would be better.
17:20.02denisgalvaowithout _stable
17:20.07craziman2k
17:20.21denisgalvaov1-0 will poitn you to the satble.
17:20.54craziman2I get "cvs [server aborted]: no such tag -v1-0"
17:21.21denisgalvaoexport CVS_RSH=
17:21.22denisgalvaoexport CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
17:21.31denisgalvaocvs login      (the password is: anoncvs)
17:21.37MikeJ[Laptop]cvs co -r v1-0 asterisk
17:21.39denisgalvaocvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons
17:22.53craziman2seams to be working... thanks
17:24.31*** join/#asterisk joerg (~joerg@p54889B45.dip0.t-ipconnect.de)
17:25.23Darwin35when are sip phones going to come down to the cost of a normal home phone
17:26.18*** join/#asterisk mithro (~tim@83-169-171-16-dynip.superkabel.de)
17:26.33*** join/#asterisk jsharp (~foo@65.90.64.82)
17:26.46QwellDarwin35: sometime in 2036 I'd imagine
17:28.05*** part/#asterisk xcore (~xcore@200.175.93.58.tbprof.gvt.net.br)
17:30.00MikeJ[Laptop]Darwin35, check tomorrow.. maybe by then.. if not, lather, rinse, repeat.
17:30.19*** join/#asterisk rashid (~rashid@63.133.150.3)
17:30.38QwellI wouldn't mind paying $10 at Radio Shack for a SIP phone
17:30.52rashidneither would i
17:31.04*** join/#asterisk oej (~oej@213.204.186.40)
17:31.16Qwelloej: evening
17:31.25oejEvening
17:31.30oej~drumilla
17:32.04oej~drumkilla
17:32.04jbotdrumkilla is, like, Russell Bryant, the Asterisk release branch maintainer <russelb@clemson.edu>
17:33.01astoriaok?
17:35.12zoahey ho oej
17:35.22Andrezowhere can i get the list of new features on 1.2? the changelog on cvs, says nothing more than a few lines
17:35.27oejHej zoa! Dinner time. Brb
17:35.30anthmok
17:35.31anthmhttp://66.250.68.190/eg/play_timed.diff
17:35.36zoahey ho anthm
17:35.49anthmhttp://66.250.68.190/eg/app_testit.c
17:35.51anthmhey zoa
17:35.52brendaNEAT!
17:36.05anthmtestit(demo-congrats|5)
17:36.35Qwellanthm: would be a nice head feature
17:36.37rashidso, I have an, hmm, interesting problem. We have a 24 port sip gateway at one location that registers to an asterisk box in our data center, calls then go out to the PSTN. When you call certain PSTN numbers the called party can not hear the caller. Other numbers work fine and once in awhile even the affected number work fine
17:37.01brendayou're tellin me
17:37.12rashidi didn't believe it until I played with it for myself
17:37.26*** join/#asterisk Barmal (~info@c-24-30-75-206.hsd1.ga.comcast.net)
17:38.07rashidbecause my cellphone, my bosses number and our consultants numbers all work fine. My girl friend's office doesn't, neither does her cell phone, my co workers house number doesn't work.
17:38.34brendaanthm: your skills make me jealous
17:38.41rashidits insanity
17:38.55Qwellbrenda: he has "other skills" too, I hear
17:39.03rashidmy boss and I sat in the pbx room for 5 hours on friday trying to figure it out
17:39.16ManxPowerrashid: remove the "r" command to the Dial line.
17:39.17brendaQwell: lol... as in what?
17:39.30Qwellbrenda: dunno
17:39.34essobiAnyone played with the new H323 drivers?
17:40.48astoriaha ha. that would suck if you spend all that time just to find out it was one letter!
17:41.21rashidwe're not using the 'r' command
17:42.22anthmdo you need to use it from the dialplan or just in C ?
17:43.01craziman2loader.c:440 load_modules: Loading module app_realtime.so failed!
17:43.01craziman2[root@smokey lib]# Ouch ... error while writing audio data: : Broken pipe
17:43.01craziman2loader.c:440 load_modules: Loading module app_realtime.so failed!
17:43.01craziman2[root@smokey lib]# Ouch ... error while writing audio data: : Broken pipe
17:43.01craziman2trying to go from latest CVS to 1.0... now I get this message when I try an asteris -cvvv "loader.c:440 load_modules: Loading module app_realtime.so failed!
17:43.03craziman2[root@smokey lib]# Ouch ... error while writing audio data: : Broken pipe
17:43.05craziman2"
17:43.26*** join/#asterisk Blackthorn (blackthorn@ws-10.smyth.net)
17:43.38Qwellcraziman2: remove all your old modules
17:44.19joshpbxand include`s too
17:44.36ManxPowercraziman2: thats a mpg123 error
17:44.42*** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com)
17:44.59rashidyup, not using it anywhere :-/
17:45.00ManxPowerthe Ouch is, the other one is not.
17:45.15ManxPowercraziman2: run "make install" and READ THE MESSAGE AT THE END
17:45.22BlackthornHello.  About every 5-7 days my sip and disa users experance poor voice quality and I can reboot the server and everything clears up. How can/should I trouble shoot this issue?
17:45.32rashidnot using any options in any of the Dial commands actually
17:45.38ManxPowerya know the message that talks about what you have to do do downgrade to 1.0.x
17:45.53craziman2where do I find the include files to remove?
17:45.53rashidBlackthorn, do you have to reboot the box or just restart asterisk
17:46.02ManxPowercraziman2: run "make install" and READ THE MESSAGE AT THE END
17:46.21*** join/#asterisk dan2_ (~foobar@dan2.professional.supporter.pdpc)
17:47.03rashidBlackthorn, we had a similar issue, ended up just making a cronjob that ran 'restart when convenient' at 3am each day. Its a hack solution, but it worked
17:47.20ManxPowerthe part where it says " ** NOTE FOR DOWNGRADING FROM CVS HEAD ** "
17:47.33ManxPowerBlackthorn: using analog cards?
17:48.22*** join/#asterisk file (~jcolp@mctnnbsah25-142166093154.nb.aliant.net)
17:48.49Blackthornrashid: i have to reboot the box. manixpower: no just running one 4 port pri card.
17:49.02rashidany other ideas for the random no voice situation?
17:49.07*** join/#asterisk DagMoller (~DagMoller@4b83b8828b0c682f.session.tor)
17:49.15Barmalwhat application is beeing used for voice Forbidden
17:49.15BarmalYou don't have permission to access /admin/ on this server.
17:49.15Barmal--------------------------------------------------------------------------------
17:49.15BarmalApache/1.3.33 Server at amp.eiktel.com Port 8080
17:49.18rashidany commands we _should_ be using in the Dial command?
17:49.23BarmalSORRY
17:49.26*** join/#asterisk wshs (screwy@69-168-246-252.bflony.adelphia.net)
17:49.56ManxPowerrashid: connect a SIP phone directly to the SIP/PSTN gateway.  If the problem still occurs, contact the gateway vendor
17:50.02craziman2okay... thanks for the help guys... sorry I ask ?? when I should have RTFM... I appreciate the help.
17:50.22ManxPowercraziman2: Did you read the README in the zaptel directory too?
17:50.49craziman2uh... I am now :)
17:51.23ManxPowerand the stuff in the doc/ directory of Asterisk?
17:51.33rashidManxPower, so here's the other part of the story. At this location we have another 24 port sip gateway, that registers to a local asterisk box, and then passes the call off to the datacenter to go out over the same connection to the PSTN
17:51.38rashideverything works fine here
17:51.40rashidhowever
17:52.01ManxPowerrashid: same vendor/make/model?
17:52.02rashidif we change the sip gateway to register directly to the datacenter asterisk, the problem rears its ugly head once again
17:52.04rashidyes
17:52.12Barmalwhat application is best to use for voice activated auto attendant?
17:52.24ManxPowerBarmal: I can't think of any.
17:52.56BarmalManxPower: but there is one made it should be....
17:53.16craziman2One more question... everything is working now I get these messages for my IAX Trunks... "chan_iax2.c:5067 socket_read: meta trunk cmd 1 received, I only understand 0 (perhaps the remote side is sending trunk timestamps?)
17:53.16craziman2"
17:53.38craziman2the reading I have done on this show that it's a stable for latest head issue.
17:53.48rashidits a real head scratcher
17:54.04craziman2Is there a way to talk IAX between a 1.0 box and a Latest CVS?
17:54.04ManxPowerBarmal: Why?  There's only one open source voice recognition project that I'm aware of and it sucks.
17:54.35ManxPowercraziman2: turn off trunking and jitterbuffer on the cvs-head box
17:54.55Barmalmanxpower, whats the name of it?
17:55.06rashidwe've been completely unable to replicate the problem reliably
17:55.09ManxPowerBarmal: Sphinx
17:55.44Kattyis firefly any good?
17:55.48Kattyor is there a better one?
17:55.57zoaidefisk!!!!
17:55.58zoa:)
17:56.35fugitivofirefly has a problem with cpu usage
17:56.40lathos42oops, wrong Firefly :D
17:56.56fugitivoidefisk seems to work well, but i don't like the interface
17:58.33Corydon-wSpeaking of speech recognition, ever been overhearing a conversation, and only discover about 5 minutes in that they're actually speaking English?
17:59.15zoafugitivo, what dont you like about the interface ?
17:59.34Corydon-wThat's Sphinx, only it's more picky
18:00.10doolphmmm
18:00.18doolphbehind nat is a problem eh
18:01.31Kattynat=yes
18:02.44*** join/#asterisk Juxt (~Juxt@64.135.20.202)
18:02.52Juxtgood day
18:03.02Juxtdoes anyone have any experience with chanspy?
18:03.49Juxti need to know where chanspy writes it's temporary file
18:04.04craziman2Have any of you seen sip 'lock up' using the latest cvs?  I see this thing where sip show peers doesn't list anything... then when I restart asterisk everything is there?
18:04.06Corydon-wHah, temporary file?
18:04.17Corydon-wWhat do you think this is, Windows?
18:04.20Juxtlol
18:04.29Juxtwell i believe that chanspy does write something
18:04.32Juxtsome sort of a stream file
18:04.38Juxtcause i've seen it before
18:04.38*** part/#asterisk DagMoller (~DagMoller@4b83b8828b0c682f.session.tor)
18:04.38*** join/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com)
18:04.40essobiuhhh.
18:04.40Corydon-wNo, it writes to a socket
18:04.44essobiNo ddude.
18:04.49essobiNo files.
18:04.57Juxtok i am wrong then
18:05.12Juxtthen is there a reason why my system studders sometimes while monitoring a channel via chanspy?
18:05.12essobiyes
18:05.18essobiyes you are
18:05.24essobiumm.
18:05.37essobitoo slow to multiplex/mixing/encoding?
18:05.41ManxPowercraziman2: at least in 1.0.x if you do a "stop when convienent" or "stop gracefully" you can't do anything at the console except exit and reconnect
18:05.46essobipossibly a bug
18:06.06Juxtthe system is a pretty powerful athlon and sometimes it studders even when there's only 1 channel going
18:06.46essobithen it's a bug or it's hardware.
18:06.56essobiwhat's "stuttering" ?
18:07.14Juxtwell the sound starts crackling
18:08.37rashidoh
18:08.47rashidand the problem only occurs with outgoing calls
18:08.56Kattyoh noes! Hmmhesays fell off teh interweb!
18:13.18Kattyhmm
18:13.28Kattyi don't understand this, in asterisk, zap show channels shows i have 4 channels.
18:13.36Kattybut in zapata.conf i said channels is 1-8
18:13.51ManxPowerKatty: put your zapata.conf on pastebin
18:15.45*** join/#asterisk lehel (~Lehel@82.79.20.17)
18:15.52lehelhello again
18:16.07*** join/#asterisk _deg_ (~deg@200.146.0.254)
18:16.18*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-75-202.cust.bezeqint.net)
18:16.23lehelrequested format = ilbc, ?? "ilbc" <?? what's this?
18:16.42leheli set up ulaw,gsm
18:17.00BoNaDsargh!
18:17.02lehelwherefrom is coming this "ilbc" ?
18:17.03rashidperhaps the requested codec from the sip client?
18:17.04*** join/#asterisk bsd3 (~bsd@203.134.192.221)
18:17.15lehelit's IAX2
18:17.19tzafrir_laptopilbc is a codec. format_ilbc.so or something similar
18:17.21rashidor IAX2 client
18:17.23BoNaDsanyone successfully get fxotune to work?
18:17.26*** join/#asterisk pa (~Paolo@pa.user)
18:17.47leheland why it is requsting in this format?.. i never specified it
18:17.51BoNaDsi stop asterisk, but it skips my fxo devices on my tdm04b cards
18:17.57*** part/#asterisk Juxt (~Juxt@64.135.20.202)
18:17.58rashidmaybe the client defaults to that
18:18.01*** part/#asterisk bsd3 (~bsd@203.134.192.221)
18:18.26tzafrir_laptopwhat is fxotune?
18:18.44rashidin any case, if you disallow ilbc in your asterisk configuration the codec actually being used will be whatever you allowed
18:19.01BoNaDsfxotune turns the hardware echo cans on in the tdm 400 cards
18:19.39tzafrir_laptopBoNaDs, I wish those small utilities were simply integrated into ztcfg
18:20.00BoNaDsheh i'll settle for just wishing that it would work
18:20.01BoNaDs=P
18:20.08lehelrashid: i found ilbc in my sip.conf.. but i'm not using SIP
18:20.10tzafrir_laptopsimply one binary that has all the ioctls
18:20.35lehelhowto stop requesting "ilbc"?
18:20.38ManxPoweranyone know of a web/domain hosting company that will host e-mail and allow access via imap
18:20.38BoNaDsits not detecting my FXO devices when clearly it should
18:20.48joshpbxdisallow this codec?
18:20.58lehelyes
18:21.00tzafrir_laptopdisallow=all, allow=certain codecs
18:21.14lehelin my iax.conf
18:21.21tzafrir_laptopBoNaDs, strace will show you where it looks
18:21.43BoNaDswhat is strace?
18:22.08yaaarBoNaDs: it's a stack tracer
18:22.14joshpbxtzafrir_laptop: but he can find it in default iax conf.. but if somone are lazy to read all conf.. ;/
18:22.22yaaarhelps you track down what a crashing program is doing
18:22.32lehelppl i have: diasallow=all, and allow=ulaw&gsm
18:22.38BoNaDsoh its not exactly crashing
18:22.40BoNaDs=P
18:22.46BoNaDsits just not detecting my boards
18:22.51BoNaDsasterisk sees them fine however
18:23.10BoNaDsinstructions say to stop asterisk before running it which i did
18:23.42ManxPowerBoNaDs: did you do a "ps -ax | grep asterisk" to make SURE asterisk is not running?
18:24.07BoNaDsyep
18:24.12joshpbxlehel: then paste u call to pastebin. maybe gw support only ilbc.
18:24.30BoNaDsSkipping non-TDM / non-FXO
18:24.38BoNaDsfxotune outputs that 8 times
18:25.01Kattyanthm: www.copi-rite.com/zapata.txt <- did i goof my channels up at the bottom for that second card somehow?
18:25.06BoNaDswhich i presume is once for each of thr 4 fxo resources on my 2 TDM04b cards
18:25.12ManxPowerBoNaDs: you have a 4-port analog card with RED modules?
18:25.28BoNaDsi have 2 brand new digium TDM04b cards
18:25.39ManxPowerwith red modules?
18:25.43BoNaDsyes
18:27.00ManxPowerand you can dial out of them with Asterisk?
18:27.00BoNaDsyes
18:27.00ManxPowerno idea
18:27.00BoNaDsme either =P
18:27.00ManxPowerYou didn't find anything helpful when you searched the mailinglist archives?
18:27.00ManxPower~mailinglist
18:27.00jbotit has been said that mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search.  Browse the mailing list archive at http://lists.digium.com/
18:27.08QwellManxPower: how much volume?
18:27.14tzafrir_laptopstrace is s system-calls trace. nothing to do with the stack
18:27.17Qwellfor the web/domain hosting
18:28.19*** join/#asterisk hex_ffff (~0x3e44d@67-41-182-243.slkc.qwest.net)
18:29.02lehelhere is my call: http://pastebin.ca/18471
18:29.02leheljoshpbx: the call
18:29.02tzafrir_laptopjbot, strace is a system-calls tracer. spits out a useful trace to stderr. To trace the command 'cmd with params' use: 'strace cmd with params'
18:29.03jbot...but strace is already something else...
18:29.06tzafrir_laptop~strace
18:29.06jbotsomebody said strace was now in a .ipk in the bleeding feed.
18:29.12Qwellheh
18:29.24*** part/#asterisk hex_ffff (~0x3e44d@67-41-182-243.slkc.qwest.net)
18:29.34tzafrir_laptopjbot, forget strace
18:29.44lehelhow's my call ppl?
18:29.57tzafrir_laptopjbot, no strace
18:30.13Qwelltzafrir_laptop: no, strace is x, should work too
18:30.27tzafrir_laptopjbot, no, strace is a system-calls tracer. spits out a useful trace to stderr. To trace the command 'cmd with params' use: 'strace cmd with params'
18:30.27jbotokay, tzafrir_laptop
18:30.28joshpbxlehel:  u use stable or cvs heads?
18:30.45*** join/#asterisk Recursion (~0x3e44d@67-41-182-243.slkc.qwest.net)
18:30.46lehelCVS HEAD
18:30.54*** join/#asterisk Nebukadneza (~daddel9@i3ED6E386.versanet.de)
18:30.56Nebukadnezahi
18:31.20Nebukadnezais there a way to execute a system command (like "sh /some/path/to/a/shellscript.sh") with asterisk?
18:31.24Nebukadnezawithout a agi?
18:31.27Qwellsystem
18:31.29leheljoshpbx: mention: i can make the call in the other way
18:31.34Qwellshow application system
18:31.42tzafrir_laptopNebukadneza, help application system
18:31.52tzafrir_laptopbah, show
18:32.05Nebukadnezahmm ... okay - ill check
18:32.06Nebukadnezathanks
18:32.09joshpbxlehel: find in u default conf and read about codecpriority
18:32.13*** join/#asterisk jarrod (anon@juniperyour.net)
18:32.13BoNaDsbbib
18:32.15*** part/#asterisk BoNaDs (~theplot@ool-44c53748.dyn.optonline.net)
18:33.23paI can formerly announce: asterisk + i4l is now working :-)
18:33.25joshpbxtry with host, if u really dont have any allow=ilbc in u iax conf it can work.
18:33.46Qwelljoshpbx: unless there is an allow=all
18:34.04joshpbxQwell: true.
18:35.04jarrodwhen using sipusers (realtime config) should i be able to see my sip users via 'sip show users' that are located in sql?
18:35.25payeah, i4l has a bit of echo.. not the best quality i think.. can be lot better with other drivers?
18:35.56*** join/#asterisk meppl (~mephisto@87.193.4.139)
18:35.57*** join/#asterisk bscheller (~none@corpwall.gsi-kc.com)
18:36.14ManxPowerpa: That depends on a lot of things, but other drivers might be better
18:36.30zoais this a chan_i4l ?
18:36.48payes, ithink
18:36.52paw8 1 moment
18:37.07pachan_modem_i4l.so
18:37.34jarroddoes anyone suspect t38 support being built into asterisk for faxing?
18:37.39jarrodthat seems to be the only thing lacking
18:38.48*** join/#asterisk jaimeco (~chatzilla@216.230.138.103)
18:38.58Delta34does anybody know what can be causing this msg
18:39.07brettnemdream on for t38
18:39.11Delta34<PROTECTED>
18:39.11Delta34<PROTECTED>
18:39.20jarrodbrett: what do you do for faxing?
18:39.24ronni'm using Playtones(ring) in  my dialplan but the ringing do not stop when i start dialing extensiions ... any idea?
18:39.25Nebukadnezahm
18:39.30brettnemDelta34: perhaps a malformed request
18:39.38brettnemjarrod. G.711 and a whole lot of faith.
18:39.50mutanyone know of any scripts/programs to analyze CDR records, stored in a database..
18:39.53jaimecosomebody know where I can download a good manual to configurate the sip extensions
18:40.02jarrodbah i think im going to spring for the cisco voice t1 card to use t38
18:40.04Delta34i get it when i call cisco iphone to iphone sip to sip
18:40.10Nebukadnezaexten => 6265494,1,System(sh /root/stable_01_ast&)   << this should execute a script that runs for ... lets say 10 minutes, but proceeds in the asterisk dialplan, right (becaus of the &)
18:40.20brettnemjarrod: running t.38 through asterisk will crush it
18:40.52jarrodyea i am just going to use a cisco then
18:41.12brettnemjarrod: you mean CCM?
18:41.32jarroda 7206 with a PA-VXB-2TE1 and t.38 to a t.38 compatible ATA
18:41.34brettnemI haven't found a good faxing solution yet.. however fax over g.711 doesn't work too bad if you control the network
18:41.59brettnemjarrod: t.38 implementation vary wildly.. no guarentee that it'll work
18:42.22jarrodi have seen it work
18:42.30jarrodusually to another cisco
18:42.37brettnemwell cisco to cisco should work
18:42.37jarrodim going to experiment with various ATA devices
18:42.55brettnemlet us know how it goes.. I've only seen trouble with t.38
18:43.06ronni'm using Playtones(ring) in  my dialplan but the ringing do not stop when i start dialing extensiions ... any idea?
18:43.07joshpbxsomone maybe have idea, why i hear voice ok, but when i record even local it`s very slowly.. when i play it 2 time fasters it`s ok..
18:43.19ManxPowerronn: you need StopTones
18:43.19brettnemronn, why are you using playtones(ring) ??
18:43.24jarrodright now i just forward the inbound faxes out another pri channel to the POTS line installed for the fax heh
18:43.38*** join/#asterisk harryvv (~none@S0106006097af532d.vs.shawcable.net)
18:43.54brettnemjarrod: embarassingly enough, that's about all I could come up with now for an alternative too..
18:44.14ronnbrettnem:  i wanted to give a dialtone as soon as some one enters an extensions and allow user dial different exten
18:44.31brettnemronn, use DISA for that
18:44.32leheli can;t imagine.. why is this ilbc codec is active?? i give now disallow=ilbc.. still requested format=ilbc
18:44.55brettnemlehel: I phone can still request it even if it's disallowed
18:45.00brettnemer I=a
18:45.12brettnemronn: really -> show application DISA
18:45.26ronnbrettnem :  DISA .. is that a command?
18:45.51jarrodthis realtime config is pimp++ with the ability to load extensions and sip/iax users into sql
18:46.06brettnemronn: it's a dialplan application.. like "playtones" is..  from the CLI type: show application DISA.. it plays dialtone, authenticates (or not) and then connects to an extension in a given priority..
18:46.31ronnthanks brettnem, i got it now.
18:46.45*** join/#asterisk darkskiez (~darkskiez@host-84-9-85-42.bulldogdsl.com)
18:47.02brettnemronn: It does exactaly what you are talking about.. keep in mind the security problems.. it's meant to give EXTERNAL users INTERNAL dialtone.. but you can do lots with it..
18:47.27Delvarjarrod: loading extensions.conf into mysql isnt so good... but the other stuff saves time++ makes it so much easyer for us to handle customers
18:47.55Delta34i did a debug and i found this Warning: 399 Bad Request - 'Malformed/Missing FROM: field'
18:48.13Beirdohmm
18:48.14*** join/#asterisk frogy (~edmund@cm222-167-86-25.hkcable.com.hk)
18:48.26BeirdoI might just change to using DISA
18:48.47brettnemBeirdo: what are you doing now?
18:48.50Beirdoright now I do the same thing with an Authenticate and then go to the internal context and let them dial
18:49.01brettnemDelta34: yep
18:49.04*** join/#asterisk CONTRABANDA (~M@213.131.37.202)
18:49.10CONTRABANDAHi all
18:49.11Beirdopretty much the same effect though
18:49.20Beirdobut DISA is the "right" way
18:49.24Delta34where do you define the FROM: field in a sip conversation
18:49.27brettnemBeirdo: do you get 2nd dialtone that way>?
18:49.38brettnemDelta34: what are you dialing from?? what is the UA?
18:49.49CONTRABANDAHow can i get h323 termination calls end then send it to various gateways ?
18:49.53Beirdonot a dialtone, but I make it say "please enter the extension you wish to dial"
18:49.54Delta34cisco 7960 to cisco 7960
18:50.09brettnemah
18:50.11Delta34sip to sip local net
18:50.17Beirdowhich can include external calls
18:50.24*** part/#asterisk perlmonky (~perlmonky@pix.benchmark-systems.com)
18:50.26brettnemDelta34: it's one of the "name" fields when you setup the SIP stuff
18:50.35Delta34in sip.conf?
18:50.49bschellerHas anyone had problems with an asterisk box with Linux 2.6 kernel freezing after running ztcfg?
18:50.51ManxPowerWell THIS isn't good.
18:51.15ManxPowerPrelim indications are that exten => fax does not work inside a macro.  Anyone else have this problem?
18:51.17*** join/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net)
18:51.32obsidian-studiosgreetings, what would cause the CALLERID var to be the value specified in a zap channel? I have * set to email me caller id info just after a zap channel is answered. Most times I get the CID/ANI info of the caller via the CALLERID var. However sometimes it returns the zap channels caller id info?
18:51.37Beirdonever tried, ManxPower
18:52.02ManxPowerobsidian-studios: It didn't get the callerid info then
18:52.18ManxPoweryou want usecallerid=yes and callerid=asreceived
18:52.51obsidian-studiosok, I have another setup using a TDM400p that always returns the zap caller id info in voicemails and etc. I have never been able to get it to get the ANI/CID info.
18:52.59brettnembscheller: are you using hdlc data or anything fancy like that?
18:53.06harryvvThis is bad, when calling in on my ivr the calling party gets the typicall ivr to call one of two extentions but then thay get a all circuits are busy please call again. No one is using the phones at the time. What would most likly cause this.
18:53.25brettnemharryvv: show us the CLI output
18:53.33ManxPowerharryvv: you have been at this long enough to know that we need a pastebin of the cli output to help you
18:53.47harryvvsure
18:53.49harryvvi know
18:53.52bschellerNo, I am using pretty standard stuff. The hardware is a T410P. No other cards. Only fix has been to power cycle the machine to recover
18:54.23jarrodi just need a way for all of the servers in this asterisk environment to read from the same extensions
18:54.30jarrodthe realtime seems like the best way
18:55.05bschellerbrettnem: sorry TE410P
18:55.10brettnemjarrod: dundi can do some of that too
18:55.40brettnembscheller: I had some problems like that with doing hdlc data on asterisk.. never got it to work right.. I'd call digium about it..
18:55.43obsidian-studiosManxPower: thanks for the info, we will see how it goes
18:55.57*** join/#asterisk dacrazyz (~www@217.129.208.124)
18:56.09brettnemthey'll want to know about a card locking up a box
18:56.25bschellerbrettnem: thanks. I had seen some weird stuff, but this one tops it. I will give them a call
18:56.25Nebukadnezahm ... i need to test some sipgate testing ... if anyone of you guys is using sipgate ... could you try calling 6265494 please? :P
18:56.35jarrodim lookin more to where if one server dies the other has an exact copy of its extensions and not where it queries
18:56.55jarrodor does 'lookups' on another server and their extensions differ
18:57.06brettnemah
18:57.12*** part/#asterisk obsidian-studios (~obsidian-@c-66-177-188-197.hsd1.fl.comcast.net)
18:57.16brettnemyeah.. trying to come up with a good solution for that myself as well..
18:57.26jarrodsql is the best way ive found
18:57.35brettnemunfortunately, I've had way too many deadlocks with all of the asterisk database components.. so I don't trust them
18:58.14_deg_Anyone using Unicall with Asterisk?
18:58.42*** part/#asterisk thal (~thalunil@walledcity.de)
18:58.50brettnemjarrod: maybe you'd have better luck ditching asterisk and going to SER? :)
18:58.55greg_workjarrod: how about lowtech, like  cron + scp
18:58.59greg_workor rsync
18:59.23brettnemyeah.. I wrote a rsync + cron utility to centrlize my configs
18:59.59jarrodthat is lame
18:59.59brettnemhmm what is that config pushing utility out there.. I forget it's name.. very powerful package..
19:00.06Delta34I think the problem is  I am getting three "Name" 4088" in the from field, not sure how thats getting inserted into the sip conversation
19:00.23Delta34three " in the sip From field
19:00.29brettnemjarrod: having asterisk hang calls because the network connectivity to the database dies is lame.
19:00.36ManxPowerDelta34: by the SIP device usually.
19:00.54ManxPowerOr you have quotes in your sip.conf, which is bad.
19:01.02brettnemyeah that's gotta be your phone, I'd think
19:01.04jarrodif its on the same LAN
19:01.04*** part/#asterisk lehel (~Lehel@82.79.20.17)
19:01.07jarrodim not worried about that
19:01.25ManxPowerI suspect I'll have to post another bounty to get the fax problem fixed
19:01.32Delta34ohh my caller id has "username" extension
19:01.32*** join/#asterisk Ash (~aaron@outofband.org)
19:01.36*** join/#asterisk craziman2 (~donnie@boromir.apid.com)
19:01.37Delta34i should remove the " right?
19:01.43brettnemyes
19:01.57ManxPoweralways remove quotes from sip.conf
19:02.02bkw_Russia's Biggest Spammer Brutally Murdered <-- see what spamming gets your ass
19:02.10*** join/#asterisk pa (~Paolo@pa.user)
19:02.13brettnemwow
19:02.16bkw_quotes?
19:02.26Ashhas anybody seen this error on debian-stable before? libgcc_s.so.1 must be installed for pthread_cancel to work
19:02.27brettnemyeah
19:02.29ManxPowerbkw_: Have you ever had an exten => fax inside a macro?
19:02.30brettnemknow any good quotes?
19:02.38bkw_ManxPower, I think it works in cvs-head
19:02.42Ashthe only problem is that libgcc_s is installed and ld knows about it
19:02.53bkw_callerid="username" <number> <-- is the righ tway
19:02.54ManxPowerbkw_: I think it doesn't in 1.0.9
19:02.59bkw_and is what the callerid parsing is looking for
19:03.01_deg_ManxPower, I will do the same bounty to Unicall
19:03.02blitzrageanyone have an idea why AGI(myscript.agi,1234) would pass the value in HEAD, but not in 1.0.9 ?  I just get "" in the same script when calling from stable, as opposed to 1234 in HEAD.
19:03.19ManxPowerbkw_: callerid= doesn't care about quotes.
19:03.31ManxPowerAnd having them will cause problems with at least some SIP clients.
19:03.39CONTRABANDAHow can i get h323 termination calls end then send it to cisco access server ?
19:03.42brettnemoh this ought to be good
19:03.44blitzrageand I think Asterisk puts the quotes around the string for you (you'll get ""name"" if you add quotes)
19:04.16jr352khi there!!
19:04.46_deg_#asteriskbrasil
19:04.55jr352khas anyone worked w/ adit 600 CMG card + asterisk?
19:05.03Delta34Yes it worked =) thxs manxpower and brettnem
19:05.11Delta34using 1.0.9
19:05.45brettnemjr352k: I have..
19:06.59bkw_snprintf(buf, bufsiz, "\"%s\" <%s>", name, num);
19:06.59ManxPowerDelta34: bkw_ is USUALLY technically correct.
19:06.59ManxPowerDelta34: but as you can see in this case.....
19:07.00blitzragelol
19:07.05brettnemhmm
19:07.10Delta34it must of changed when i upgraded to 1.0.9, was working on 1.0.2
19:08.17bkw_ManxPower, ast_callerid_parse I think isn't as picky about it.. but you should still quote it to stop it from trying to figure out exactly what you ment
19:09.10harryvvManxPower take a look at this. have both cli output and the extentions below it. I have come to relize for some odd reason when the calling party calls in on the zap hears the ivr to press one of two extentions that asterisk instead dial up the main phone number the calling party is calling on. I have not made any changes to really reflect this. http://pastebin.ca/18476
19:11.18ManxPowerbkw_: if you quote it the quotes are passed to the destination SIP device and as Delta34 has seen, some of them don't like that
19:12.25harryvvmanx, ever see a bug where a extention is pressed and it dials the main zap dial un number?
19:12.34ManxPowerharryvv: no.
19:12.36*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
19:12.43harryvvThats what this is doing.
19:12.49ManxPowerSince Asterisk doesn't even know what the main zap number is
19:12.49harryvvnever done this before.
19:13.21anthmi'd say the thing setting callerid should nuke quotes as a policy I recall making a dequote func in app.c
19:13.29anthmfor the gotoif stuff
19:13.38*** join/#asterisk dant (~dan@81-86-69-213.dsl.pipex.com)
19:13.38ManxPower<PROTECTED>
19:13.50ManxPowerharryvv: the SIP client has call forwarding turned on
19:13.57harryvvahhh
19:14.02harryvvmabey wife did that
19:14.47harryvvso asterisk does not automaticly send that to the phone company then
19:15.05Nuggetprobley not.
19:15.52anthmhttp://66.250.68.190/eg/tmp.c
19:16.06harryvvits a feature my wife wants
19:16.26anthmdata = ast_strip_quoted(data, "\"", "\"");
19:17.00anthmyou can just do that on the name portion and it will strip whitespace and stay inside quotes if the exist and eliminate them
19:19.07ManxPowerOr you can just not use quotes in sip.conf, iax.conf, zapata.conf, etc
19:19.36harryvvI guess the only way around my case is to route it out to iax.cc or another zap line.
19:22.55anthmkinda like you can just opt to not hit tab in the cli cos you never know when it will crash your box
19:23.18ManxPowerExactly!
19:23.18*** join/#asterisk Darwin35 (~darwin@ip70-186-117-198.ma.dl.cox.net)
19:25.50essobiheh
19:25.50anthmor always say disallow=all cos it makes one feel they worked around a codec bug =>
19:25.50*** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net)
19:25.50ManxPowerGranted it should be fixed in the source code, but that doesn't help anyone not running CVS-HEAD
19:25.50Darwin35ManX did it I saw him
19:25.50ManxPoweranthm: I do that too.
19:25.50fockswhat would cause sporatic echo on a PRI Zap channel? Using Polycom Soundpoint 501
19:25.50jarrodbrett: do you use ser?
19:25.50anthmwell I suggested that code cos it could be easily patched into stable
19:25.50ManxPowerthese days I do allow=all in [general] then disallow=all and allow=myhappycodec in each sip.conf section
19:25.50anthmit's like a 10 line func in utils.c
19:25.50Dovidhello all
19:25.50ManxPowerfocks: lower your txgain on the pri.  the exho is caused by the remote analog line
19:25.50Dovidanyone know of any software for phone spoofing ?
19:25.50anthmwhen you get to 1.2 you will be in for a treat cos I reworked all the codec stuff to work right
19:25.50focksManxPower, that's in zaptel.conf right?
19:25.51harryvvdovid, you mean cid spoofing?
19:25.51ManxPoweranthm: Hmm?  How do?  Does allow=all no longer say Asterisk supports G723.1 or G729 (without a license)?
19:25.51*** join/#asterisk ai-a[afk] (~gandalfii@81.168.0.204)
19:25.51Dovidharryvv: yes
19:25.51anthmallow=all means whatever you said in general now
19:25.51ManxPowerDovid: there is no such application.  I'm sorry.
19:25.59Dovidkk
19:26.02harryvvI suspect register with a iax service that allows you to set your own cid.
19:26.06ManxPoweranthm: Um, it seems to me that that happens in 1.0.x
19:26.09Dovidcause i know people that did it with asterisk
19:26.20Nebukadnezaexten => 6265494,1,System(sh /root/stable_01_ast&)
19:26.20anthmso you setup general the way you like it and it's inherited to each peer
19:26.20Dovidbut i want the caller to set it when they call
19:26.23Nebukadnezahm .. why doesnt this work
19:26.24anthmmaybe that made it in stable
19:26.27Nebukadnezait seems its just not executed
19:26.30anthmi get lost on the timeline
19:26.43tzafrir_laptopNebukadneza, * runs as root? bad
19:26.50Nebukadnezai know :P
19:26.57Nebukadnezathis is going to change soon ...
19:27.06anthmif show codecs has this
19:27.08anthm<PROTECTED>
19:27.17anthmit the top its got my patch
19:27.25Nebukadnezabut still ... why isnt this working?
19:27.27DovidManxPower: I just found an AGI for it
19:27.57ManxPoweranthm: the entire reason I have allow=all in [general] is because disallow=all in [general] not allow ANY codecs in each sip.conf stanza
19:29.45anthmwith the new way, if you allow=a,b,c,d instead of all it preserves the order
19:29.50*** part/#asterisk dan2_ (~foobar@dan2.professional.supporter.pdpc)
19:30.32anthmit works better in iax than sip cos mark vetoed the bit that enforced the order based on the config in favor of the stupid sip way
19:30.56Delta34in the sip show peers, how do u get the status code to "monitored" mine shows as Unmonitored
19:30.57anthmbut in iax you have a ton of control over codec negotiation now
19:32.29ManxPowerDelta34: qualify=yes in each sip.conf entry
19:32.36*** join/#asterisk R3DB0x (nobody@66.142.28.36)
19:33.19Delta34i have that set in global settings for sip.conf
19:33.27Delta34doesnt it apply to all sip clients?
19:36.56ManxPowerDelta34: it might in cvs-head, but not in 1.0.x
19:37.38twisted[asteria]anyone know if zapata can successfully be read from realtime?
19:38.22ManxPowertwisted: it doesn't really matter since chan_zap.so has to be unloaded and reloaded anyway
19:38.47zoadoes somebody know what esf is exactly ?
19:38.57zoaextended super frame, but what does it do ?
19:39.02zoaand how does it relate to asterisk ?
19:39.19anthmisnt that one of the pri choices ?
19:39.22twisted[asteria]zoa, it's t1 framing
19:39.30ManxPowerzoa: ESF is what lets you have 64k clear channel on T-1
19:39.40anthmyah that's it , t1 the physical layer option
19:39.41twisted[asteria]ManxPower, no it doesnt, ninny, I made chan_zap reloadable last year
19:39.47twisted[asteria]reload chan_zap.so
19:40.00twisted[asteria]it will reload everything except the signalling
19:40.03ManxPowertwisted:
19:40.05ManxPowerpbx-1*CLI> reload chan_zap
19:40.05twisted[asteria]and modify the settings accordingly
19:40.05ManxPowerpbx-1*CLI>
19:40.06anthmits the one you pick between d4 and esf it's usually on channel banks as one or the other
19:40.12ManxPowertwisted: nifty.
19:40.16twisted[asteria]chan_zap.so
19:40.23twisted[asteria]don't forget the .so
19:40.31ManxPowerpbx-1*CLI> reload chan_zap.so
19:40.32ManxPowerpbx-1*CLI>
19:40.32jarrodanyone running a multiple server environment?
19:40.35ManxPowernothing.
19:40.38twisted[asteria]ManxPower, heh... what version?
19:40.40jarrodwith multiple entry points and config sharing
19:40.41*** join/#asterisk pfn (~pfnguyen@netblock-66-245-252-239.dslextreme.com)
19:40.42ManxPowerDoes it reload with no output?
19:40.44MikeJ[Laptop]that's silly that it needs the .so.. silly
19:40.45ManxPowertwisted: 1.0.x
19:40.46twisted[asteria]no, it gives output
19:40.52twisted[asteria]but remember, realtime isn't in stable
19:40.56twisted[asteria]and that may not be either
19:41.00ManxPowerSpecifically 1.0.9
19:41.05ManxPowertwisted: *nod*
19:41.11zoais it like a specific version of AMI and B8ZS ?
19:42.50Kattytwisted[asteria]: WAKE UP
19:43.12twisted[asteria]Katty, i'm awake
19:43.14twisted[asteria]barely
19:43.15twisted[asteria]heh
19:43.20MikeJ[Laptop]:)
19:43.41Kattyeww :<
19:43.44Kattyi got 8ish
19:43.46twisted[asteria]yeah, it sucks
19:43.52twisted[asteria]i usually get anywhere from 6-8
19:43.57twisted[asteria]but tonight i feel will be a 12 hour night ;)
19:44.49Kattyyum.
19:45.18Kattyi'll spend it for you
19:45.20fileI got a new geek toy today but I can't play with it until work is over :( makes me sad
19:45.21Kattyso you feel important, etc.
19:45.28Kattyfile: :<
19:45.45Kattyfile: :>
19:45.52filehaha
19:46.04twisted[asteria]Katty, hmm?  spend what?
19:46.16*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
19:46.22Kattytwisted[asteria]: your 12 hour night's worth of pay :>
19:46.40twisted[asteria]oh, i meant 12 hour night's worth of sleep :P
19:46.51Corydon-wfile:  yeah, I think the boss would look poorly upon playing with a dildo at the office...
19:46.52Kattyoh
19:46.55Kattywelllllll
19:46.56twisted[asteria]i'm salaried, i get paid the same regardless
19:46.57filehaha
19:46.57Kattynevermind then!
19:47.01Kattyi'll go spend my own money!
19:47.04Kattyand yours too
19:47.07MikeJ[Laptop]Corydon-w, bad!
19:47.09jarroddoes a module need to be loaded for extconfig.conf to be parsed
19:47.12jarrodor is that default
19:47.25KattyCorydon-w: dirty.
19:47.32*** join/#asterisk neonet2006 (~icechat5@iphost-64-56-140-28.wpg.wiband.net)
19:47.36neonet2006hello
19:47.44twisted[asteria]Katty, if you can find a way to spend my money, heh, you'd deserve it
19:47.46neonet2006how can set the npi and ton in asterisk
19:47.49Corydon-wKatty: not with proper soap and water cleaning, it won't be...
19:48.05pfnaw shit, I deleted my old box and deleted my asterisk diffs, boooo
19:48.24Corydon-wNow you know why most of us submit patches...
19:48.25Kattytwisted[asteria]: i shall attempt sleeve tugging, and then the pout.
19:48.31Kattytwisted[asteria]: if that does not work, then i shall whine.
19:48.38Kattytwisted[asteria]: and if that does not work, then i shall tickle
19:48.43Corydon-wIt's not because we feel like contributing; it's so we don't lose them
19:48.48Corydon-w;-)
19:48.51Kattytwisted[asteria]: there's a whole list of tricks ;)
19:49.26neonet2006how can change the ton and npi in asterisk?
19:49.34twisted[asteria]Katty, hehe.. sleeve tugging and pouting usually only gets evil looks and the occasional head patting
19:49.37Corydon-wKatty: you forgot finding twisted wallet while he's sleeping...
19:49.42twisted[asteria]whining usually makes me turn on the ipod
19:49.47twisted[asteria]and tickling usually makes me tickle back
19:50.03Corydon-wOoooh... I can get twisted to tickle me?
19:50.12Kattytwisted[asteria]: what about The Whimper(tm)?
19:50.13twisted[asteria]Corydon-w, uhm... no
19:50.30twisted[asteria]Katty, lol... i haven't had that one tried on me yet
19:50.34Kattytwisted[asteria]: k
19:51.53neonet2006?any help?
19:53.00Kattymy jewish blood is quite handy at times (=
19:53.09Kattyand bloody annoying at others
19:53.37Corydon-wneonet2006: pridialplan in zapata.conf
19:54.22focksI've got Polycom SoundPoint 501's setup to auto-answer for 4 digit internal extensions. Problem is, when I perform a transfer (blind or supervised) it doesn't distinguish between a transfer and dialing normal internal extension to intercom. How can I work around this?
19:54.28Corydon-wThat specifies the outgoing TON
19:54.28*** join/#asterisk Jearil (~Jearil@67.151.27.214)
19:54.36twisted[asteria]Katty, you're jewish?
19:54.52focksaside from assigning a second extension like 8XXXX for intercom and leaving XXXX for normal dialing
19:54.59neonet2006is there a way to force the ton and npi in the extensions.conf
19:55.06neonet2006for specific dial plans
19:55.17Kattytwisted[asteria]: about 45% german, 45% irish, and 10% jewish
19:55.22Corydon-wneonet2006: why would you want to?
19:55.23twisted[asteria]hehe word
19:55.24*** part/#asterisk bscheller (~none@corpwall.gsi-kc.com)
19:55.25tzangerKatty: rearrange some over to mine
19:55.25*** part/#asterisk illek (~mike@ip68-227-104-152.ok.ok.cox.net)
19:55.29Kattytwisted[asteria]: you could call me aggressively thrifty (=
19:55.46Kattymy savings account is fluffy
19:55.49Kattybut i have no credit
19:55.51*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
19:55.52twisted[asteria]Katty, i'm 25% irish, 25% scottish, 25% english, and 25% cherokee indian
19:55.56ManxPowerzoa: http://en.wikipedia.org/wiki/Digital_Signal_1
19:56.02Kattysoon i shall have credit card and then i shall aquire credit
19:56.06neonet2006i would like to set specific caller ID Number with proper TON and NPI to be terminated on another gateway
19:56.21Kattytwisted[asteria]: k (=
19:56.29Kattymy hair has streaks of red in it
19:56.36Kattyi guess that's called auburn
19:56.36fileKatty rocks my world
19:56.53Katty..!
19:56.57twisted[asteria]Katty, hehe, yeah, my hair is like a dirty red, and my facial hair is red.
19:57.05Kattytwisted[asteria]: post gifs :>
19:57.06twisted[asteria]whoa
19:57.09filedeath by ticklking?
19:57.10twisted[asteria]Katty, there are some already
19:57.11Darwin35my replacement and new x401 phones are on thier way
19:57.11fileer tickling?
19:57.14Kattytwisted[asteria]: post url
19:57.16Darwin352 phone
19:57.17twisted[asteria]not gifs, but jpegs
19:57.19Kattytwisted[asteria]: k
19:57.20*** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk)
19:57.27twisted[asteria]Katty, http://gallery.indigent-networks.com
19:57.29twisted[asteria]somewhere in there..
19:57.31Kattyfile: death by SOY
19:57.34Kattytwisted[asteria]: k
19:57.37fileooh
19:57.55ManxPowerzoa: or http://www.dcbnet.com/notes/9611t1.html
19:58.07izoanybody using ata 186 ?
19:58.17Kattytwisted[asteria]: in friends?
19:58.35Darwin35now to get the firmware src code
19:58.41neonet2006??
19:58.51twisted[asteria]Katty, nah, that's mostly friends of mine.  look in VON for one or two
19:58.55Kattytwisted[asteria]: k
19:58.56twisted[asteria]i know there's a good one of me and drumkilla
19:59.22paI have a debian dist, and i changed in /etc/defaults/asterisk RUNASTERISK from no to yes, but if i issue "/etc/init.d/asterisk start", it does not start... :-/
19:59.25Corydon-wneonet2006: pridialplan=dynamic
19:59.50neonet2006and how do i set the NPI and TON in extensions.conf
20:00.05pai can still launch it by hand..
20:00.27Kattytwisted[asteria]: oooh, pretty female at VON :>
20:00.42pai could add it to inittab, but i think that letting start it to rc is better
20:00.45*** join/#asterisk darkskiez (~mhb@host-84-9-85-42.bulldogdsl.com)
20:00.51twisted[asteria]Katty, hmm?
20:00.51ManxPowerCorydon-w: What version of Asterisk supports pridialplan=dynamic?
20:01.45Corydon-wManxPower: dunno... it's been in there a while
20:01.50neonet2006and how do i set the NPI and TON in extensions.conf
20:01.51ManxPowerpa: can you start Asterisk with "asterisk -cvvv"?
20:02.18Qwellhmm
20:02.19Corydon-wSomebody's switch wouldn't support setting pridialplan=unknown, so he went ahead and made it work
20:02.26*** part/#asterisk Nebukadneza (~daddel9@i3ED6E386.versanet.de)
20:02.32ManxPowerCorydon-w: Well it's not documented in zapata.conf.sample in 1.0.9
20:02.41Qwellforget this whole "bootup services" stuff
20:03.01Corydon-wManxPower: Dunno... I don't run 109
20:03.15Corydon-wManxPower: I don't use it, either
20:03.30Kattytwisted[asteria]: neat sunglasses
20:03.46ManxPowerThen it's prolly not in 1.0.x
20:04.16paManxPower: yes
20:04.27twisted[asteria]Katty, thx
20:04.36Corydon-wManxPower: yep, it's in HEAD only
20:04.38paManxPower: asterisk -vvvc works great
20:04.39neonet2006so no doc for this feature?
20:04.48Corydon-wManxPower: ain't the source grand?
20:06.34ManxPowerneonet2006: what version of Asterisk are you running
20:06.44Corydon-wneonet2006: who needs documentation when you have the source?
20:07.15Darwin35coool
20:07.29Darwin35I will have 2 new phones this rocks
20:07.45neonet20061.0.8
20:07.58neonet2006i still need documentation.....
20:08.11neonet2006a project with docs....means nothing
20:08.12Darwin35for what
20:08.15ManxPowerneonet2006: I strongly doubt that feature is supported in 1.0.8
20:08.24ManxPowerneonet2006: You're not running BRIStuff, are you?
20:08.26Corydon-wManxPower: in fact, it is not
20:08.34neonet2006no I am not running it
20:09.05Corydon-wForget support... it's not even in 108
20:09.09neonet2006what I am trying to do is set the ton and npi when I force the dial plan to show specific callerID number
20:09.16ManxPowerApparently Wait(1) will trigger a faxdetect if it's run
20:09.18neonet2006SetCallerID()
20:09.23ManxPower(and a fax machine is calling)
20:09.38neonet2006is there  a way to do it in the dial plan (extensions.conf)
20:09.55*** join/#asterisk T-Squared (~ted@hidden.serreyn.com)
20:09.58ManxPowerneonet2006: 1) Why? and 2) I don't think you can do what you want to do using the version of Asterisk you are using.
20:10.42Corydon-wneonet2006: no, there is no way to do that with the version you're using
20:11.08neonet2006I am trying to terminate on a gateway (hooked up to a PRI) the PRI provider needs to see specific ton and npi
20:11.12neonet2006national/isdn
20:11.20neonet2006what version should I get?
20:11.35ManxPowerneonet2006: What happens if you just set pridialplan=unknown?
20:11.40zoawhat is npi and ton ?
20:11.57ManxPowerzoa: You saw my ESF links?
20:12.20neonet2006i am not originating the calls on the zaptel cards
20:12.27neonet2006stricly SIP
20:12.30ManxPowerneonet2006: %90 of people's TON problems is that they set prodialplan= something other than "unknown"
20:12.31zoayeah thanks
20:12.53ManxPowerneonet2006: You can't send that information over SIP as far as I know.
20:13.09neonet2006that what I think as well
20:13.31neonet2006so I need to be using zaptel hardware to get it working or what?
20:14.00neonet2006the info as far as i know is related from one gateway to the other? no SIP intervention?
20:16.16neonet2006how would transmitt that info, then?
20:16.24GoshenNufone.net crapping out again?
20:16.24zoatwisted, do you have a link explaining that ESF makes sure you can use 64kbit ?
20:16.47Goshenthe voice is really choppy again, like it was when they were having problems last week....
20:16.48*** join/#asterisk PakiPenguin (uppal@202.61.58.73)
20:17.05*** join/#asterisk __kop__ (~kop@71-35-174-95.tukw.qwest.net)
20:17.17DarthClueGoshen: if it's a tollfree, come to asterlink.
20:17.46Goshenurl?
20:17.50*** join/#asterisk TripleFFF2sdf (~TripleFFF@modemcable131.156-131-66.mc.videotron.ca)
20:17.59DarthCluehttp://www.asterlink.net
20:18.02TripleFFF2sdfhey..
20:18.21GoshenDarthclue: not resolving
20:18.24DarthClueGoshen: if you're interested, msg me and we can get you on the express route.
20:18.37pai think i will let asterisk start from inittab, so i can redirect it to a virtual terminal
20:18.49GoshenDarthClue: asterlink.com?
20:18.52DarthClueGoshen: sorry, need more caffiene, asterlink.com
20:19.12neonet2006how would transmitt that info, then?
20:19.13darkskiezpa, safe_asterisk script does all that for you
20:24.37padarkskiez: but i cant start it from rc script :-(
20:24.40neonet2006is there a way to see what npi and ton is trasmitted from asteriks
20:24.48neonet2006how do i enable the log
20:25.00neonet2006what log should i enable
20:25.01TripleFFF2sdf./usr/lib64/gcc-lib/x86_64-suse-linux/3.3.3/../../../../x86_64-suse-linux/bin/ld: cannot find -lssl
20:25.01TripleFFF2sdfcollect2: ld returned 1 exit status
20:25.06TripleFFF2sdfhmm
20:25.25fileyou don't have the library
20:26.07*** join/#asterisk L|NUX (linux@202.5.146.154)
20:26.19TripleFFF2sdfyes itsthere
20:27.05ManxPowerTripleFFF2sdf: in /usr/lib ?
20:27.06*** join/#asterisk craziman2 (~donnie@boromir.apid.com)
20:27.13TripleFFF2sdfy
20:27.19TripleFFF2sdfadded
20:27.20TripleFFF2sdf-ldl -lpthread -lncurses -lm -lresolv   -L /usr/local/lib -L /usr/lib -L/usr/pkg/lib  -lssl
20:27.20ManxPoweror somewhere else that's listed in /etc/ld.so.conf
20:27.36ManxPowercan't have a space after -L I don't thin
20:27.40TripleFFF2sdfhmm
20:28.08TripleFFF2sdffixed that
20:28.10TripleFFF2sdfbut still
20:28.17fileit can't find it...
20:28.25TripleFFF2sdfnow.. b1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv   -L/usr/local/lib -L/usr/lib -L/usr/pkg/lib  -lssl
20:28.38kFuQ<PROTECTED>
20:28.59ManxPowerTripleFFF2sdf: well put the directories in /etc/ld.so.conf and rerun ldconfig -v then you don't have to put all that extra shit in the makefile
20:32.44*** join/#asterisk Hmmhesays (negative3k@66.173.103.108)
20:33.01*** join/#asterisk BleedingMe (~BleedingM@ppp-69-238-171-80.dsl.scrm01.pacbell.net)
20:33.56TripleFFF2sdfstill no go
20:33.57TripleFFF2sdfoh
20:33.59TripleFFF2sdfhmm
20:34.42BleedingMedoes anyone know anything about www.iax.cc / www.sixtel.net  ?
20:34.50TripleFFF2sdfwould suse compile as ifeq (${OSARCH},SunOS) ?
20:35.06ManxPowerTripleFFF2sdf: If it does, it would be a major bug in Suse
20:35.43*** join/#asterisk loick (~loick@APuteaux-151-1-20-68.w82-124.abo.wanadoo.fr)
20:36.02*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
20:36.46TripleFFF2sdfahah
20:40.02TripleFFF2sdfstill not
20:41.28filethere's a readme for suse...
20:41.42filemight yield some info
20:42.31TripleFFF2sdfk
20:42.38TripleFFF2sdf./usr/local/ssl/lib/ is where it is
20:42.44TripleFFF2sdfand thats in ld.conf
20:42.52TripleFFF2sdf,res_crypto.c:556: warning: implicit declaration of function `SSL_library_init'
20:42.52TripleFFF2sdfres_crypto.c:557: warning: implicit declaration of function `ERR_load_crypto_strings'
20:42.52TripleFFF2sdfmake[1]: *** [res_crypto.o] Error 1
20:42.52TripleFFF2sdfmake[1]: Leaving directory `/usr/local/src/asterisk/res'
20:42.53TripleFFF2sdfnow
20:42.54TripleFFF2sdflol
20:43.02TripleFFF2sdfman this IS not a plug and play
20:43.04TripleFFF2sdf;l)
20:43.54*** join/#asterisk wunderkin (~kev@12-215-218-160.client.mchsi.com)
20:43.58mishehuTripleFFF2sdf: I put my monopoly money on the problem being something to do with includes.
20:45.04TripleFFF2sdflol yeah
20:45.17TripleFFF2sdfthing is.
20:45.29TripleFFF2sdfis i add that dir to libs of the ifeq (${OSARCH},SunOS) then i goest a bi further but dumps
20:48.09*** join/#asterisk jsmith (~jsmith@smithfam.dsl.xmission.com)
20:52.29TripleFFF2sdfhmm if i addd -L/usr/lib
20:52.37TripleFFF2sdf./usr/lib64/gcc-lib/x86_64-suse-linux/3.3.3/../../../../x86_64-suse-linux/bin/ld: skipping incompatible /usr/lib/libdl.so when searching for -ldl
20:52.40TripleFFF2sdfi get these
20:52.43TripleFFF2sdf64
20:52.52TripleFFF2sdfso .. i trie /use/lib64 then no luck at all
20:53.14RaYmAn-Bxhow is adding libraries to the command line going to help with problems finding declarations that has to do with includes? :>
20:53.25TripleFFF2sdfnot sure
20:53.27TripleFFF2sdflol
20:53.39TripleFFF2sdfalso its says while searching for  searching for -lm
20:53.43TripleFFF2sdfwell
20:53.49TripleFFF2sdfim at a dead end then
20:53.52TripleFFF2sdfany dsuggestio ?
20:54.55RaYmAn-BxDo you have a library called "libssl.so" in any of the paths you pass? Like, check that it's actually there and stuff
20:55.32TripleFFF2sdf<PROTECTED>
20:55.52TripleFFF2sdf<PROTECTED>
20:55.58puzzledanyone use MMX optimizations enabled in zconfig.h in zaptel (stable)? positive difference?
20:56.07*** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com)
20:56.27zoadunno if there is a difference, but it doesnt seem to crash :)
20:56.34puzzledhehe
20:56.46puzzledzoa: so you usually have it enabled on prodction boxes?
20:57.29jsmithpuzzled: I've been using it on production boxes for years.
20:57.52*** join/#asterisk asterisk99 (~chatzilla@modemcable111.209-131-66.mc.videotron.ca)
20:57.54puzzledjsmith: ok thanks
20:57.56file[laptop]oh no... IT'S JARED SMITH!
20:58.01RaYmAn-BxTripleFFF2sdf: was that an answer or? Because it doesn't really give any information whatsoever...
20:58.25jsmithfile[laptop]: That would be me -- scary, isn't it?
20:58.30file[laptop]jsmith: quite
20:58.39mutthere a way to turn off verbosity no the cli at all?
20:58.42mutset verbose 0
20:58.45jsmithGood thing they're not putting my picture on the book...
20:58.50mutand i still see warnings when i reload the config
20:58.53file[laptop]jsmith: it would scare people away
20:58.54asterisk99anyone know how to specify specific days of week in GotoIfTime (e.g. Mon+Sat+Sun) ????
21:02.08MeatyGotoIfTime(*|sat-mon|*|*?context,extension,priority)
21:02.13Meatyasterisk99 ^^
21:02.27Meatyor Maybe GotoIfTime(*|sat-sun-mon|*|*?context,extension,priority)
21:02.38Meatydunno
21:02.46Meatytest it !
21:02.59jsmithasterisk99: If I remember correctly, it's mon,tue,wed
21:03.09ManxPowerIt uses the same syntax as include =>
21:03.11*** part/#asterisk mkrufky (~mk@68.160.103.77)
21:03.49asterisk99Meatty: sat-sun-mon is invalid
21:03.57asterisk99Meaty: sat-sun-mon is invalid
21:04.09Recursionjsmith: You're West's cousin, right?
21:04.11Meatysat-mon ?
21:04.21asterisk99jsmith: can't use commas either
21:04.32Meaty?
21:04.43ManxPowerperhaps it's 0123 for sun,mon,tue
21:04.44Meatyyes
21:04.47Meaty\,
21:04.55jsmithRecursion: Man, I'm famous!
21:04.56ManxPowerOr you can look up the syntax for include =>
21:04.59jsmithRecursion: Yes, I am.
21:05.09Recursionjsmith: soon to be even more famous!
21:05.20Meatytry sat-mon
21:05.41jsmithRecursion: Exactly!
21:06.06ManxPowerday-day is the correct formaty
21:06.30jsmithManxPower: So if I want Monday, Friday, and Saturday, how do I do that?
21:06.42ManxPowerjsmith: Did you look it up?
21:06.47jsmithNot yet...
21:06.54ManxPowerPerhaps now is the time to do so.
21:06.58darkskiezjsmith: you could always do multiple lines of gotoiftime
21:07.00asterisk99jsmith: it'll do range, but nothhing else
21:07.17*** join/#asterisk criptos (~criptos@201.138.231.189)
21:07.28asterisk99jsmith: yeah.... looks like I have to tetest 1 day at a time
21:07.35criptosanyone using snom190 phones? I want to know how, from asterisk, specify the ring tone to use...
21:07.38jsmithI was afraid of that...
21:07.41Delta34so i removed the " " in the callerid field under sip.conf but now my direct to voicemail buton is not working now, i guess it is looking for "Username" 4088 but its getting Username 4088 instead
21:07.56Delta34the voicemail button now is asking for extension then pwd
21:08.02Delta34rather then just asking for pwd
21:08.09ManxPowerDelta34: that's a callerid problem.
21:08.18ManxPowerWhat is the Voicemailmain line show on the CLI?
21:09.12Delta34exten => 4199,1,VoicemailMain(${CALLERIDNUM}@default)
21:09.20*** join/#asterisk santiago (~santiago@63.245.86.188)
21:09.37ManxPowerNo, from the CLI!  And your voicemail button should be dialing 4199
21:09.56*** join/#asterisk zotz (~zotz@24.231.36.100)
21:10.01ManxPoweri.e. AFTER CALLERIDNUM is replaced with the correct value during the dialplan processing
21:10.03Delta34whats the command in CLI
21:10.24darkskiezwouldnt be hard to rewrite the get_dow function in pbx.c to take comma seperated days.
21:10.26ManxPowerno, reproduce the problem.  You'll see in the CLI the voicemailmain being run
21:10.48Delta34ohhh one sec
21:10.53ManxPowerdarkskiez: Since Asterisk replaces "," with "|" internally, it's a lot harder than you think.
21:11.29darkskiezManxPower: err, fuck.
21:11.43ManxPowerWhenever I need to concatinate things I do it using &
21:11.48ManxPowerLike Dial and others do.
21:11.53Delta34Executing VoiceMailMain("SIP/4088-c09f", "Jackie Lau 4088@default") in new stack
21:12.12ManxPowerDelta34: Egads!  It should be 4088 right?
21:12.21ManxPowerWhat have you done do your callerid?
21:12.29ManxPowerDelta34: paste the sip.conf line for callerid=
21:12.44darkskiezwouldnt be hard to rewrite the get_dow function in pbx.c to take & seperated days.
21:12.47darkskiez:)
21:12.50Delta34callerid=Jackie Lau 4088
21:12.53ManxPowerdarkskiez: yeah
21:12.56Delta34i removed " "
21:13.08ManxPowerDelta34:  callerid=Jackie Chan <4088>
21:13.10ManxPowerNotice the <>
21:13.17ManxPowerwhatever is in <> should be CALLERIDNUM
21:13.49ManxPowerIt's callerid=calleridname <calleridnum>
21:15.18jr352kguys is there any way on the dial plan to limit the number of digits a user enters in order to get them capture: ex dial your 8 digits pin number
21:15.36Hmmhesayswhen did chanspy make it back into head?
21:15.48ManxPowerHmmhesays: Dunno.  Was news to me too a few weeks ago
21:15.56anthmwhen did it leave ?
21:16.15Hmmhesayshrm...i'm pretty sure it wasn't in there for awhile
21:16.27Delta34cool thxs manxpower
21:16.28anthmit's been there since it was added
21:16.34Delta34was a syntax error on my part
21:16.37Hmmhesayshrm, I must be losing my mind
21:16.43Hmmhesayswhich is completely possible
21:16.58anthmrevision 1.1
21:16.58anthmdate: 2005/03/24 01:19:02;  author: markster;  state: Exp;
21:16.58anthmAdd chanspy (bug #3836)
21:17.35Hmmhesayshasn't chanspy been around a lot longer than that?
21:17.52anthmnot in CVS
21:18.01anthmon my pc it's been there for a year
21:18.27anthmyou'll want 2005/07/20 or better for the best version of it
21:18.29ManxPowerHmmhesays: I recommend doing a "show applications" ever few months 8-)
21:19.18Hmmhesaysof course
21:19.27anthmthere is also app_muxmon on deck which is a call recorder that uses the chanspy hooks
21:20.44*** join/#asterisk caramb1 (~alfons@c-f8ef70d5.09-237-73746f34.cust.bredbandsbolaget.se)
21:21.35ManxPowerI really like being a consultant.
21:22.04ManxPowerI just called up my largest customer, asked if anything was happening this afternoon, he said there isn't, so I said I was taking the rest of the afternoon off.
21:22.39caramb1I read that using mpg123 is not recommended. So I wonder, is there a gstreamer module for asterisk? That would be far more versatile than mpg123.
21:22.50ManxPowercaramb1: where did you read THAT?
21:22.52file[laptop]mpg321 is not recommended
21:22.55file[laptop]mpg123 IS recommended
21:22.56puzzledyou can use madplay too
21:23.03file[laptop]0.59r to be exact
21:23.09file[laptop]or you can use native audio if you really want...
21:23.12file[laptop]or something else...
21:23.19ManxPowerfile: You sure are getting smart in your old age.
21:23.32file[laptop]haha
21:23.47puzzled-O6 optimization in asterisk. do people leave that in or tone it down to something less aggressive?
21:23.59SwK[Work]most people leave it
21:24.01puzzled(in stable)
21:24.08ManxPowerpuzzled: I never touch the makefile, except to enable the option to make G726 work with SIPura
21:24.14Nuggetthat's nutty.  I'd never noticed the -O6
21:24.22emrahHello again everyone
21:24.26file[laptop]make thing go fasta!
21:24.35SwK[Work]Nugget: doesnt d.net use like -O99? heh
21:24.45puzzledNugget: line 66 of the Makefile
21:24.52Nuggetno, d.net doesn't use compiler optimization
21:24.53caramb1ManxPower: At http://www.mpg123.de/ it says:  It is highly recommended to not use the source code you can download from this site
21:24.57Nuggetdnetc is -by--hand
21:24.58Nugget:)
21:25.01Nuggeter, --by-hand
21:25.04SwK[Work]thats like -funroll-loops
21:25.16Qwellcaramb1: from the asterisk source dir, do make mpg123
21:25.17SwK[Work]see www.funroll-loops.org for definitions
21:25.19ManxPowercaramb1: ignore that.
21:25.22Nuggetis -O6 even valid?
21:25.25SwK[Work]yes
21:25.28caramb1Qwell: Aha! Thanks.
21:25.35emrahI'm having a problem with AreskiCC. The script is just like doing nothing. May I past a part of the error message, or a link to pastebin?
21:25.48tzafrir_laptop~pastebin
21:25.48jboti heard pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
21:26.03ManxPowercaramb1: if you can install mpg123 0.59r from a package do that.  If not do a "make mpg123" in the Asterisk source directory before running "make install"
21:26.09tzafrir_laptoppaste your stuff there, paste link here
21:26.15Nuggetwhat nutball gentoo-addled mouth breather put -O6 in the Makefile?
21:26.22puzzledlol
21:26.25emrahOk
21:26.29ManxPowerWhat's wrong with -O6?
21:26.43eKo1isn't that for optimizing?
21:26.44puzzledback at 0.7 or so I always changed it back to -O2 or -O3
21:27.03tzafrir_laptopNugget, I think it is taken from a kernel tree build command
21:27.05ManxPowereKo1: -Ox is optimization level
21:27.12Nugget-O6 is totally a "this one goes up to eleven" act.
21:27.27Nuggethell, -Os is proably the best bet for most cases
21:28.39ManxPowerNugget: What problems might -O6 cause?
21:28.52emrahthere is the link
21:28.54emrahhttp://pastebin.ca/18489
21:29.06puzzledManxPower: in gcc4 all hell will break loose and you will not have the afternoon off :)
21:29.07*** join/#asterisk Ayano (~erik_leee@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
21:29.16Nugget-O7 is where the gcc documentation says to never tread, so presumably someone saw that note and decided that -O6 must be the best choice.
21:29.18emrahNo logs
21:29.20Nuggetbut it often isn't
21:29.21emrahanything
21:29.26AyanoAny of the asterlink guys around?
21:30.04bkw_yes
21:30.06puzzledNugget: what do you think is a good value? -O2, -O3? never heard of -Os. I have gcc 3.4.3
21:30.06DarthClueyes
21:30.08bkw_Ayano, whas up
21:30.19bkw_-06 baby
21:30.21Nuggetlots of code runs faster at -O2 or -O3 than at -O6.  it's not like a big "code speed" volume knob
21:30.22bkw_er O
21:30.29bkw_haha
21:30.37bkw_ya it might make smallercode
21:30.38AyanoI wanna go to cluecon, do you know if there are any rooms still available?
21:30.38bkw_but not faster
21:30.44bkw_Ayano, yes
21:30.46DarthClueAyano, yes
21:30.51MikeJ[Laptop]echo
21:30.54bkw_need to get you registered and paid ASAP
21:30.59MikeJ[Laptop]echo
21:31.00AyanoWhat do I need to do?
21:31.14DarthClueAyano: go to cluecon.com and register
21:31.15file[laptop]go to http://www.cluecon.com/ and register
21:31.17emrahPlease, anyone can help me?
21:31.21MikeJ[Laptop]hehe
21:31.32file[laptop]I have a beep, someone beeped me...
21:31.35Nugget-Os optimizes for size, which can reduce cache pressure and in many cases is a much bigger win than trying to optimize for speed.
21:31.42BeirdoBEEP!
21:31.50Beirdoit wasn't me though
21:32.15ManxPower+zaptel 1.0.9.1
21:32.16ManxPower+ -- continue fxo operation after the magical 25 days
21:32.21Nuggetheh
21:32.26*** join/#asterisk stkn (nobody@stkn-active-pdpc.developer.gentoo)
21:32.38iCEBrkrSTFU LILO
21:32.39ManxPowerNugget: Don't we only care about speed?
21:32.53emrahIs there anyone wh could help me?
21:33.05Nugget-O is about specific optimizations, not speed.  Sometimes they translate to speed, sometimes they harm speed.
21:33.11DarthClueemrah: be patient, someone will eventually come along who understands areskicc
21:33.18emrahThanks
21:33.44puzzledlike areski
21:33.48iCEBrkrWhat the heck is areskicc? LOL
21:33.56iCEBrkrThe calling card thing?
21:33.58puzzledcalling card app
21:34.03*** join/#asterisk drooth (~drooth@ip68-111-235-172.sd.sd.cox.net)
21:34.11HmmhesaysI had all kinds of trouble with areskicc
21:34.36ManxPowerI am SO glad I don't have to bill for calls.
21:34.49iCEBrkrI wrote my own CC thing before he came out with a dedicated project on it. I was just tinkering.
21:34.55Hmmhesayspost paid isn't so bad
21:35.05asterisk99why not bill based on the info in thhe CDRs?
21:35.12iCEBrkr...and to be honest, I'm not sure mine worked 100%
21:35.50asterisk99you gottum da number called... youo gottum the billsecs... you gottum the source... what more could you want?
21:36.04iCEBrkrasterisk99: It's not about what to bill on, it's how to do all the IVR stuff.
21:36.10ManxPowerasterisk99: the account information 8-)
21:36.18asterisk99thilly me!!!!
21:36.42ManxPowerI just let my LD carrier bill for the calls based on the account code they enter in
21:36.56ManxPowerGranted, this is in an enterprise enviroment
21:37.16ManxPowerI'll be the first IAX2 provider that does something similar will make a fortune.
21:37.48Darwin35?
21:38.11fugitivocan i use a fax machine with the tdm400?
21:38.13*** join/#asterisk Habakuk (~andy@64.1.15.130.ptr.us.xo.net)
21:38.57Darwin35re think that question and think what ports are on the tdm400
21:39.31*** join/#asterisk IRCMonkey0815 (~asterisk-@dsl-084-056-187-232.arcor-ip.net)
21:39.40fugitivomy question is "does fax work with the tdm400 or it just receives noise"?
21:39.47emrahNo-one can help me?
21:39.49Delta34when using the Queue cmd, is there a way of actually ringing the queue instead of playing the moh for the enduser
21:40.19*** join/#asterisk Robot_ (~Robot_@84.47.4.242)
21:40.28ManxPowerDelta34: "show application queue"
21:40.37*** join/#asterisk Trifixion (~trif@c-24-23-130-6.hsd1.ca.comcast.net)
21:40.41*** join/#asterisk justin_e (~justin_e@c-67-180-16-102.hsd1.ca.comcast.net)
21:40.51ManxPowerDelta34: pay SPECIAL attention to the "r" option of queue
21:40.52TrifixionQuestion for the Asterlink crew - how do I connect to Asterlink using SIP instead of IAX?
21:40.58Delta34cool thxs
21:41.20DarthClueTrifixion: msg me your userid and i'll get file right on that.
21:41.24denonTrifixion: you're probably better off asking asterlink support
21:41.41denonor him
21:41.54IRCMonkey0815hi all, i've problems with my TE411P (Zaptel-1.0.9), when overlapdial=yes  * get killed when someone calls on my 3 digit extension, can anyone help me ?
21:42.53emrahanyone still?
21:43.01ManxPowerIRCMonkey0815: turn off overlapdial
21:43.47IRCMonkey0815thats not really an option! because only phone which calls with complete Number will be able to call
21:44.00rashidstill haven't figure out my silence problem
21:44.03ManxPowerIRCMonkey0815: You don't need overlapdial for that.
21:44.11rashidmight have to give up and just put a local asterisk box at that site
21:44.24ManxPowerIRCMonkey0815: Are you using PRI or E&M for your DIDs
21:44.56justin_eHey all, I just installed Asterisk CVS-v1-0-07/25/05-11:15:51, when I try to dial from my 7960 to Ast Demo Ext I get no audio, if unload the wct1xxp and try again it works fine, ideas?
21:44.59Trifixionwhats overlapdial?
21:45.04IRCMonkey0815hmm, really. But if not incoming calls are being rejected after the first number,
21:45.23ManxPowerIRCMonkey0815: Are you using PRI or E&M for your DIDs
21:45.51NuggetI suspect the choice of -O6 was from someone who stumbled across the old docs which warn you to never use O7 or higher and the person figured that O6 must be optimal.
21:45.57IRCMonkey0815ManxPower: which zapata or zaptel option are you talking about
21:46.06Trifixionlol O6 doesn't do any more than O3.
21:46.20ManxPowerIRCMonkey0815: I cannot help you any further until I know what signalling the line is set for.
21:46.49tzafrir_laptopNugget, check how modules are being compiled in kernel 2.4. IIRC it is with -O6 as well
21:47.03IRCMonkey0815ok what i know - German E1, ccs,hdb3,crc4
21:47.04Nuggetwell, that's linux for you.  :)
21:47.30ManxPowerIRCMonkey0815: in /etc/asterisk/zapata.conf what is the signaling= line?
21:47.34*** join/#asterisk hound (~AirSaniti@1feaf9de96afd7da.session.tor)
21:47.53*** join/#asterisk DannyF (~dannyf@h194n1fls32o865.telia.com)
21:48.03ManxPowerNot knowing the signaling of a line is like not knowing your default route or netmask
21:48.05IRCMonkey0815ManxPower: euroisdn
21:48.10Kattyoh
21:48.13KattyJul 25 16:46:27 WARNING[879]: chan_iax2.c:7357 socket_read: Received mini frame before first full voice frame
21:48.16Kattywhat's that mean?
21:48.21ManxPowerIRCMonkey0815: I've NEVER gotten overlapdial working with a USA PRI.
21:48.24*** join/#asterisk bonez41 (~aint@c-67-166-77-14.hsd1.ut.comcast.net)
21:48.27Nuggetso, I'm relieved to learn that the ridiculous -O6 in the asterisk makefile hasn't been producing shit code for us all, but I'm dismayed to learn that it's there to begin with.  :)
21:48.30ManxPowerKatty: harmless
21:48.35KattyManxPower: uhm
21:48.39KattyManxPower: let me rephrase that
21:48.42KattyManxPower: WHAT DOES IT MEAN
21:48.46KattyManxPower: m'kay? (+
21:48.46Katty(=
21:48.57ManxPowerIRCMonkey0815: make SURE immediate=no and overlapdial=np
21:49.05Qwellit means a mini frame was received before the first full voice frame was
21:49.10Kattyi see.
21:49.10Qwellbut, what do I know?
21:49.17IRCMonkey0815ManxPower: is configured definetly
21:49.31ManxPowerKatty: it means that Asterisk got a miniframe (which can NEVER be the first frame) before it got the first full frame for a call.
21:49.43KattyManxPower: and what causes this?
21:49.53ManxPowerKatty: random routing oddities
21:49.58Kattyk
21:50.07ManxPowerKatty: out of order packets are not all that unusual.
21:50.39KattyManxPower: thank you (=
21:51.55tzafrir_laptopso why warn about this?
21:52.11IRCMonkey0815ManxPower: with that config: calls to 'XXXX0' ok! calls to 'XXXX11' will be cut to 'XXXX1'
21:52.45IRCMonkey0815ManxPower: but only in slow dial
21:53.26*** join/#asterisk Robot_ (~Robot_@84.47.4.242)
21:54.19emrahPlease, I'm desesperatting with this strange problem.
21:54.34Trifixionemrah - whats the problem?
21:54.51Trifixionif you want asterisk to run faster do the -mcpu=686 stuff.
21:54.56Trifixionnot -O99999
21:55.08*** join/#asterisk d00gster (~doughant@toronto-HSE-ppp4334604.sympatico.ca)
21:55.12QwellTrifixion: can-tehl-pself-itis?
21:55.25*** part/#asterisk d00gster (~doughant@toronto-HSE-ppp4334604.sympatico.ca)
21:55.28Qwellthel rather, heh
21:55.30doolphanyone can help me with gnugk?
21:55.38doolphor where to get that kind of help?=
21:56.17Trifixiondoolph - i'm not ashamed to admit that i'm very intimate with gnugk.
21:56.25puowvipI'm starting a company and you are all hired.
21:56.28Trifixionin fact, i use gnugk source code as erotic fiction every now and again.
21:57.27doolphTrifixion I didnt understand you lol
21:57.38Trifixioni'm saying that gnugk gets my rocks off!
21:57.40Trifixiongot it?
21:57.41puowvip#asterisk <Trifixion> in fact, i use gnugk source code as erotic
21:57.41puowvip+fiction every now and again.#asterisk <Trifixion> in fact, i use gnugk source code as erotic
21:57.58*** join/#asterisk pigpen (~mark@fw.seamans.cc)
21:58.16doolphit means that you are good or bad
21:58.24Trifixioni'm badgood/goodbad.
21:58.32IRCMonkey0815ManxPower: any hints
21:59.13puowvipTrifixion: you're a zebra-hat?
21:59.29Trifixionyup
21:59.36ManxPowerIRCMonkey0815: those two things are all I had to set to make my PRI work, other than the obvious things
22:00.03doolphTrifixion do you have time to me?
22:00.12Trifixionyup
22:00.35opus_hey
22:00.35IRCMonkey0815ManxPower: hmm! what is your option at pridialplan "unknown" ?
22:00.41opus_hmmtl
22:01.15*** join/#asterisk Oryn (oryn@falcore.fsck.tv)
22:01.58Orynanyone know if its possable to use ogg vorbis for voicemail attachments?
22:02.06Trifixionhttp://www.crapville.com/media_videos12/treat_her_right.wmv
22:02.08Trifixion(fyi)
22:03.17Nuggetis there a techical reason to prefer vorbis, or is it just a political desire because vorbis makes your shit smell like roses?
22:03.25*** join/#asterisk aminorex (~tony@66-191-69-132.dhcp.dlth.mn.charter.com)
22:03.50Nugget"RMS refuses to leave voicemail until we stop using GSM"  :)
22:04.08*** join/#asterisk hypa7ia (~leigh@33051397cc8359ec.session.tor)
22:04.24*** join/#asterisk Coriantum (~asdfkle@67-41-182-243.slkc.qwest.net)
22:04.42Orynyeah its easyer to play than wav files, for some reason the player that my email client uses wont play wav files
22:04.55Orynplus I'm hoping they will be smaller
22:05.24OrynNugget: mp3 would be ok I guess
22:05.25puowvipeh
22:05.33CoriantumI just started using AEL, could someone recommend a website I could look at for some documentation?  Other then the Readme.
22:06.15jsmithCoriantum: If you find something good, jump in to #asterisk-doc and let us know, please...
22:06.25*** join/#asterisk Milligan (~stephen@208.189.212.200)
22:07.06CoriantumIf I cant find something then I will start writing my own
22:07.22Darwin35AEL is new not well documented yet
22:07.27Darwin35its only in head
22:07.33CoriantumI noticed
22:07.34Darwin35if you use it document it
22:07.43Darwin35then put it in a wiki
22:07.51jsmithAnd in #asterisk-docs, please...
22:08.18Trifixiondid you guys check out that URL i posted?
22:08.20Trifixionit's hilarrrrrious.
22:08.22Darwin35I take the .ael must be asterisk-extensions-logic language
22:08.34CoriantumWill do
22:08.48*** join/#asterisk pdracevich (~bob@210.54.249.228)
22:09.16pdracevichis the Asterisk CVS server down?
22:09.57Darwin35it should not be
22:10.29Orynso is ogg a dirty word in here?
22:10.29pdracevichtryed downloadin from it.  It timed out at the PW stage tried to ping it no luck as well
22:10.52fileworking for me
22:10.57*** join/#asterisk gast (~riemensc@83-169-155-92-dynip.superkabel.de)
22:11.04gastguten abend
22:11.24pdracevich*sigh* must be my ISP
22:12.14gastich habe mal ne frage bezüglich asterisk mit zaphfc
22:12.38gastbei mir klappt zwar die eingehende telfonie aber nicht die ausgehende
22:13.01NuggetWir sind alle Kuhe.
22:13.09eKo1lol
22:13.23gastich erhalte immer die fehlermeldung extensions in context ´default´ from ´meine gesetzte msn´ does not exists rejeting call on channel 0/2, span 1
22:13.45pdracevichexit
22:13.52gastkennt jemand von euch die fehlermeldung und wie kann man ihn beheben?
22:14.04eKo1bitte frag auf englisch weil die meisten hier verstehen dich nicht.
22:14.17gastspreche leider nicht so gut english
22:14.39gastkönnte es aber mal schnell durch babelfish senden oder meinst du das bringt nichts
22:14.57eKo1versuch es mal
22:15.08florzgast: Wenn Du hier fuer Belustigung sorgen willst ... =:-)
22:15.17Nuggetdeine english ist besser dan meine deutcsh.
22:15.36Nuggetund ich have zwei jahre deutschklasse!
22:15.36gastsorry, jeder kann sich mal vertippen
22:16.16gastda ich mir ja schon den spot auf mich gezogen habe, kann mir jemand schildern wie man das problem löst?
22:16.37gastasterisk mit hfc karte -> soll sich am sip anmeldungen
22:16.38florzgast: Was Dein Problem angeht: Naja, Du hast halt im Dialplan nicht spezifiziert, was passieren soll, wenn die betreffende MSN angewaehlt wird.
22:16.50gastdas telefon klingelt
22:17.04gastkann ja nur nicht abgehend telefonieren, das macht mich ja so verrückt
22:17.21*** join/#asterisk zagaya972 (~d2s-compa@APointe-a-Pitre-102-1-7-21.w81-248.abo.wanadoo.fr)
22:18.01florzgast: Aehm, oder so - halt die Nummer, die Du waehlst, ist im Dialplan nicht spezifiziert.
22:18.09tzafrir_laptopgast, isn't there any #asterisk-de or something?
22:19.03gast[general]
22:19.03gaststatic = yes
22:19.03gastwriteprotect=no
22:19.03gast[globals]
22:19.03gastIAXINFO=guest   ; IAXtel username/password
22:19.03gast[default]
22:19.05gastinclude=>calls
22:19.07gast[calls]
22:19.09gastexten => 1015396,1,Dial(Zap/2/30690116,60tT)
22:19.11gastexten => 1015396,2,Hangup
22:19.13gast;exten => s,1,NoOP
22:19.15gastexten => _9.,1,Dial(SIP/${EXTEN:1}@sipgate,60,tT)
22:19.17gastexten => _9.,2,Congestion
22:19.19gastexten => _9.,3,Busy
22:19.21tzafrir_laptop~pastebin
22:19.21jbothmm... pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
22:19.21gastexten => _9.,4,Hangup
22:19.23gastdie 30690116 sollte meine msn sein
22:19.27puzzledgast: please use pastebin.com
22:23.01Darwin35~jbot sex
22:23.01jbotupdatedb; locate; talk; date; cd; strip; look; touch; finger; unzip; uptime; gawk; head; apt-get install condom; mount; fsck; gasp; more; yes; yes; yes; more; umount; apt-get remove --purge condom; make clean; sleep, or super extractor, http://sf.net/projects/sex/
22:24.42eKo1hahaha
22:25.44Nuggetthat command line is fundamentally flawed.
22:25.49Nuggetthey should be &&'s, not ;'s
22:26.09Nuggetyou sure as hell wouldn't want to proceed to strip if the cd failed!
22:26.34*** join/#asterisk Ahewes (~rsb@209.133.58.210.aaph.com)
22:26.48AyanoAlright, I'm signed up to get a clue at cluecon!
22:27.59*** join/#asterisk pfn (~pfnguyen@netblock-66-245-252-239.dslextreme.com)
22:28.22niZondoes nufone allow you to set the outgoing callerid?
22:28.39DarthClueAsterlink does.
22:28.54Nuggetasterlink rocks my socks.
22:30.02*** join/#asterisk jeffgus (~jeffgus@greengables.zimage.com)
22:30.18niZonthey seem a little coming soon..
22:30.29DarthClueniZon: who? asterlink?
22:30.52niZonyeah
22:30.53niZonlol
22:31.00AyanoThese guys at asterlink worry me.  They are too on top of thier game!  They are too good!  :)
22:31.01Ayanolol
22:31.22niZonlol
22:31.31niZoncan they do paypal?
22:31.33DarthClueniZon: we aren't coming soon.  we are here.  unfortunately, i think the web guys went on vacation and forgot to finish the site.
22:31.40niZonlol
22:31.43DarthClueniZon: if that's what you use, i'm sure we can.
22:32.05niZonUS50/Canada toll free? prepaid?
22:32.07DarthClueniZon: might cost a little more though since we would have to pay a fee to paypal.
22:32.28DarthClueUS48 right now, we are working towards Canada, etc.  yes, 2c prepaid.
22:32.46niZonaw i want canada
22:34.09*** join/#asterisk focks (~cbruender@nsc66.147.95-93.newsouth.net)
22:34.34fileCanada is expensive
22:34.52AyanoA lot of taxes right?
22:36.21justin_eHey all, I'm running CVS-v1-0-07/25/05-11:15:51, SIP->External Sip Provider works, but SIP->AST (Playback) doesn't send audio (tethereal shows no outbound audio). If unload the wct1xxp and try again it works fine. Ideas?
22:36.21niZonhm
22:36.21file[laptop]no, it's just expensive...
22:38.00niZonlink2voip claims they can do it for 3 cents per min
22:38.27niZonproblem is, they have almost nonexistant support and they don't seem to actually have DIDs
22:38.49file[laptop]all I can tell you is that for people who have their own PRIs and circuits, it's expensive
22:39.15syle2thats why you hookup to tandum switches file
22:39.26syle2PRI is wasteful money hehe
22:39.37file[laptop]PRI was just in my head
22:39.48file[laptop]doesn't mean I have a PRI or something :P
22:41.43*** join/#asterisk danalien (~danalien@danalien.user)
22:49.12fockson an incoming context, why won't s,1,... match any incoming trunk? I see an error about no matching extension in [incoming] but i didn't think s cared about the 4 digits coming in off the PRI
22:50.01*** join/#asterisk meppl (~mephisto@87.193.6.7)
22:50.27file[laptop]s is when asterisk doesn't know what number was dialed
22:51.01focksfile[laptop], so if it does know the extension, i have to explicitly use it?
22:51.22*** join/#asterisk daniel101 (~daniel101@dsl15-088.express.oricom.ca)
22:51.43focksie, instead of s i'd use like _XXXX to match any digits for the [incoming] context
22:52.10file[laptop]exactly!
22:52.41focksfile[laptop], hmm, wonder why this example only listed s and nothing about actual extensions
22:53.29focksoh well
22:54.05file[laptop]bkw!
22:54.11*** part/#asterisk justin_e (~justin_e@c-67-180-16-102.hsd1.ca.comcast.net)
22:56.58syle2i;m having a problem where can barely hear the caller on the pap2-na
22:57.38*** join/#asterisk kslater (~kslater@24.svnf1.xdsl.nauticom.net)
22:57.40syle2ulaw from termination provider to me-> tried ulaw to pap2-na device and g729 without any result changes
22:57.41eKo1turn the volume up
22:57.45syle2any ideas what else i could try
22:57.54syle2turn volume up where
22:58.05eKo1on your phone
22:58.06*** join/#asterisk iswm (iswm@iswm.user)
22:58.24syle2there is no volume on the phone
22:59.00eKo1sucks to be you then
22:59.22syle2sucks to be every use that buys a 20 dollar phone at walmart you mean
22:59.27syle2thats pretty much everyone
23:00.20syle2no personally i have the latest and greatest, but thats not the point, its the enduser i;m talking about :)
23:01.20eKo1Ditto.
23:02.39*** join/#asterisk btm (~b1ueemu@70-33-140-162.agstme.adelphia.net)
23:03.00*** join/#asterisk justin_e (~justin_e@c-67-180-16-102.hsd1.ca.comcast.net)
23:03.55*** join/#asterisk colinm_ (~colol@VDSL-130-13-9-155.PHNX.QWEST.NET)
23:03.59*** part/#asterisk justin_e (~justin_e@c-67-180-16-102.hsd1.ca.comcast.net)
23:05.08*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
23:09.38timecophm nice
23:09.47timecopever since i changed fwd to IP address i dont see it failing to register anymore
23:09.53timecopDNS IN ASTERISK SUCKS.
23:10.56opus_Jul 25 16:07:26 WARNING[2252]: rtp.c:950 ast_rtp_settos: Unable to set TOS to 184
23:11.47puzzledtry tos=0x18
23:11.57*** join/#asterisk Romik_ (~romik_@1.fix.netvision.net.il)
23:12.12*** join/#asterisk stock (~stock@stokkie.demon.nl)
23:12.23Romik_somebody can advice what DEAD PACKAGE means?  iax2 show stats
23:12.25Romik_<PROTECTED>
23:12.27Romik_---------------------
23:12.29Romik_Outstanding frames: 6 (0 ingress, 6 outgress)
23:12.31Romik_Packets in transmit queue: 6 dead, 1 final, 6 total
23:12.41Romik_packets i rather.
23:15.12opus_is there a problem with SER where the enduser hangs up and SER never sends the hangup back to asterisk?
23:17.14*** join/#asterisk Syncros (~sysop@noc.routermonkey.net)
23:18.23*** join/#asterisk craziman2 (~donnie@boromir.apid.com)
23:18.58Ahewesanyone else getting a lot of errors trying to compile chan_zap.c in cvs head?
23:20.26Ahewesooops, my bad. libri.h no such file or directory. bonehead mistake.
23:21.52*** part/#asterisk Ash (~aaron@outofband.org)
23:22.52kslatertrying to find an answer on E911 for a server for use at home
23:23.40niZonget a cheap basic landline and an FXO interface
23:23.50Nuggetdoes "trying to find an answer" ever actually involve, you know, asking a question?
23:23.54*** join/#asterisk Sedorox (brandon@sedorox.staff.smartserv)
23:24.04niZonor if you're cheap, get a big red phone and plug it in to the landline
23:24.06Nuggetthat's what most of us do.
23:25.02newlor just don't provide it..you're not a carrier.
23:26.13kslaterok. so since it's a home system and I have a TDM with both FXO and FXS modules, I should be fine, eh?
23:26.49kslaterNugget: I suppose you're correct, I should have asked for a pointer at least.
23:26.59gordonjcpkslater: what's the problem?
23:27.15newlSure.  Though if you're concerned with E911 service, you should be speaking with your VoIP provider instead. :)
23:27.27*** part/#asterisk caramb1 (~alfons@c-f8ef70d5.09-237-73746f34.cust.bredbandsbolaget.se)
23:28.06kslaterno problem yet. I intend to be my own voip provider
23:28.24kslaterso my concern is that people in my house still have access to 911 service
23:28.46gordonjcpkslater: ?
23:29.18gordonjcpkslater: do you have a landline coming in at all?
23:29.34kslateryes
23:29.51gordonjcpgreat, plug an el-cheapo POTS phone in
23:29.58gordonjcpjob done
23:30.14gordonjcpor use your mobile
23:30.23gordonjcpno sense hunting around for a landline in an emergency
23:30.26gambolputtyswitch => Realtime isn't working for me.  Failed for anyone else?
23:30.34kslaterso they'd have to use the POTS phone for 911, couldn't use any phone connected to * for 911?
23:30.46colinm_kslater: They could use a phone connected to *, so long as there's power.
23:30.49gordonjcpkslater: yes, if it dialled out over the normal line
23:31.12kslaterok. so if I lose power, then they'd need to the landline
23:31.13Nuggetif you buy a sipura to hook up the pots line to asterisk, then you can just route 911 out over the pots line
23:31.21Nuggetbuy a UPS.  they're like $99
23:31.30*** join/#asterisk tessier (~treed@146.82.146.22)
23:31.39kslaterNugget: have a honkin' ups already on the server with *
23:31.46gordonjcpor, as I say, just don't worry about it
23:31.50kslateralso have a TDM400 card
23:31.51tessierAnyone know the trick to configuring a Handytone-486? I plug in my analog phone, press the button, but don't hear an IVR.
23:31.53gordonjcpuse your mobile
23:32.00Nuggetso what's the problem?
23:32.06Nuggetsounds like you know all the answers already
23:32.27kslaterjust trying to be sure I understand. I don't know much as it turns out. that's why I'm asking here.
23:33.40kslaterI suppose having all cordless phones it's no different in a power failure situation.
23:34.53opus_ever tried calling 911 from a cellphone? 9 out of 10 times it won't wor
23:34.54opus_wor
23:34.57opus_work
23:35.01tessieropus_: Why not?
23:35.05opus_dunno
23:35.07gordonjcpopus_: wtf?
23:35.13Nuggeti've never had a problem with 911 on a mobile.
23:35.28opus_how many times have you called 911?
23:35.32Nuggetthree or four times
23:35.38gordonjcpno problems either time
23:36.04gordonjcpI wouldn't even consider using a landline
23:36.12Nuggetonce to report a car crash and three times to report drunk drivers.  no problems at all
23:36.42syle2power failure at home?
23:36.52gordonjcpI don't think I've even had a landline for about 10 years
23:36.56gordonjcpmaybe 8
23:36.56kslatergordonjcp: wouldn't consider a landline?
23:36.58syle2just install a solar power panel
23:36.59kslaterah
23:36.59gordonjcpnope
23:37.05gordonjcpkslater: not these days
23:37.10kslatergot it
23:37.11syle2run all your electricity in your house off that :)
23:37.12Delta34anybody know much about the tranfer cmd
23:37.12gordonjcpkslater: maybe if I was getting ADSL
23:37.14kslatersingle?
23:37.36gordonjcpsingle what?
23:37.38Delta34if someone is calling thru zap channel to sip line
23:37.52kslatergordonjcp: are you single?
23:38.00Delta34can i transfer the call back out the zap channel to external number
23:38.05gordonjcpkslater: no
23:38.15gordonjcpkslater: why?  and what kind of question is that anyway?
23:38.36kslatergordonjcp: too many ppl in this house for cellular to be affordable
23:38.44kslaterthat's why I asked.
23:39.04gordonjcpkslater: it's practically free, my bill is about £15 a month
23:39.20gordonjcpI don't think i could even get a line for that, never mind call anyone
23:39.36syle2kslater just clone your smartcard
23:39.38gordonjcpI'm not even sure it's as much as that, tbh
23:39.44kslatergot it. different situation here. sorry if I offended you
23:40.14opus_yeah, i need a sim card cloner
23:40.20opus_can i borrow yours syle2?
23:40.21gordonjcpkslater: put it this way, the direct debit for my phone bill isn't enough to show up on a monthly bank statement
23:40.25opus_does it work off the air?
23:40.43gordonjcpand I use my phone a lot
23:41.53kslaterok. you guys answered my questions. time to push forward with the asterisk project. I have 4 kids that could use it..
23:43.59Sedoroxtoo bad there isn't a iden kit to connect to asterisk :p
23:45.37*** join/#asterisk wolfson (~hehe@68-187-187-034.dhcp.mant.nc.charter.com)
23:45.57puzzledasterisk on slashdot: http://joybubbles.telephreak.org/papers/vpa/
23:48.12PyroStevehey guys
23:48.49PyroStevei just implentmented concurrent call limits in my asterisk install by using the SetGroup stuff
23:48.57PyroSteveseems like its work great
23:49.22PyroStevebut here is an a question that I guess could be answered by expermintation ...
23:49.43PyroStevewhen a call is added to a group .. then the call is finished ...
23:49.58PyroStevedoes the call simply get removed from the group
23:50.27Sedorox0_o
23:50.50PyroSteveis my keyboard working ?
23:50.56Sedoroxno.. it isn't
23:50.57Sedorox:p
23:50.59Kattythere are 4 T in 1/4 C, right?
23:51.03SedoroxI duno the answer.. sorry
23:51.14Sedorox-_-
23:51.19KattyWHAT
23:51.29Kattyhave you insaned?
23:51.41SedoroxI've been insane
23:51.46Kattyoh
23:51.50Kattywell that explains everything
23:51.51Sedoroxlol
23:51.53Kattyanyone else know?
23:51.54Sedoroxyup
23:52.24gordonjcpfour what in a quarter what?
23:52.39Kattynevermind ;)
23:52.42Sedoroxlol
23:52.59*** join/#asterisk hypa7ia ([U2FsdGVkX@a7ac4e237de46e4f.session.tor)
23:53.58twistedwheeee.
23:54.08KattyBEEP BEEP
23:54.16twistedbeep beep :)
23:54.19Kattytwisted: don't call, i'm working out
23:54.28twistedyou're working out on irc?
23:54.28Sedoroxbackuping up?
23:54.29Kattyi shall not interrupt teh reverse wrist calls.
23:54.39Kattywell im sorta between sets
23:54.45twistedahh
23:54.56*** join/#asterisk litage_ (~nick@ws01.5749.dsl.winshop.com.au)
23:55.02twistedi consider IRC a workout sometimes
23:55.21Qwelltwisted: Want a real workout?
23:55.26SedoroxHmmm
23:55.29Kattyi hear sex is a good workout
23:55.32Sedorox0_o
23:55.37twistedQwell, uhm, if I wanted that, I'd ask Katty if i could join her set
23:55.37Kattysomeday i'll find out (=
23:55.39Sedoroxsay... Katty wanna work out with me
23:55.40Sedorox?
23:55.41Sedorox:p
23:55.41Qwellput documentation in one room, a shitload of cables in front of you, and make sure you have a computer that you can't get to the back of
23:55.45KattySedorox: uhh, no
23:55.57twistedbut my workout today has been staying awake
23:56.15Qwell(in my case, a shitload is 1)
23:56.26twistedQwell, like the size of a volkswagen/
23:56.43Qwelltwisted: 1/8" stereo cable :p
23:56.54twistedQwell, ah. close enough.
23:57.10QwellI had a space about 11" wide, 3' deep, trying to plug shit in blind...
23:57.19twistedQwell, welcome to racks
23:57.20SedoroxKatty: darn :(
23:57.23Qwelltwisted: indeed
23:57.43KattySedorox: sorry, i'm not a casual sex type person
23:57.47Sedoroxthats the best.. plugging stuff in blind
23:57.51KattySedorox: in fact, i'm still one of the innocents (=
23:57.53twisteddon't harass Katty, she's a sweetheart
23:57.55Kattyand prefer to keep it that way
23:57.59SedoroxKatty: hey.. so I am...
23:58.00Sedorox:/
23:58.02Kattyk
23:58.05Delta34anybody know if a 2.4 or 2.6 kernel better for asterisk? or really doesn't matter
23:58.13SedoroxI use 2.6
23:58.25opus_2.6
23:58.45syle2what the HELL is wrong with casual sex!!!
23:58.47opus_2.4 requires a timing device,
23:58.58Sedoroxno one said there was
23:58.59opus_well, you can use the dummy one.
23:59.00niZonanyone used asterisk_php?
23:59.01Kattysyle2: i didn't say anything was. i just said i didn't have it
23:59.02*** part/#asterisk hound (~AirSaniti@1feaf9de96afd7da.session.tor)
23:59.03filecasual sex where?
23:59.04Sedoroxwe're just stating that we're not into it
23:59.04Sedorox:p
23:59.07Sedoroxlol
23:59.12twistedfile, your hand
23:59.15Kattyfile: over there, in the corner
23:59.15Sedoroxfile: see #causual_sex
23:59.18Delta34doesnt digium card have a timer device?
23:59.23Sedoroxsp
23:59.28SedoroxDelta34: yes
23:59.31twistedKatty, ew.. I don't wanna clean that up...
23:59.41gambolputtyI compiled the newest CVS *, and suddenly my phones won't register.  http://pastebin.com/320657  could someone take a look?

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