irclog2html for #asterisk on 20050810

00:00.18Ariel_SkramX, you know speanish?
00:00.25Maxx4lifercam, Inbound/Outbound?
00:00.29SkramXNot much.
00:00.30RCAMMaxx4life Both.
00:00.32Maxx4lifeSi Ariel... Poco
00:00.36RCAMMaxx4life Mostly in.
00:00.45SkramXRCAM: so like DID's?
00:01.01Maxx4lifeOut isn't much of a problem... In on the other hand is a bit harder....
00:01.01NuggetI speak functional "lawn care and food ordering" spanish, as does anyone who lives in texas for more than a month.
00:01.15DonXGot a quick ? that maybe some of you guys will have a quick workaround for. I'd like to create and extension that when dialed, will prompt for a pin and upon successfull authentication, will drop to a dialtone or provide another means for the user to dial out to the PSTN
00:01.23RCAMSkramX Yes with Vanity 800s.
00:01.27Strom_CDonx: easy
00:01.27DonXIs there a semi easy way to do that?
00:01.35Maxx4lifedonx, www.voip-info.org look for DISA
00:01.38SkramXnufone.net, asterlink.com, link2voip.com.... also voip-info.org has many resources listed
00:01.45tzangerDonX: show application DISA
00:01.57RCAMSkramX I know... But I am just wondering if anyone has a recommendation
00:01.59DonXthanks :)
00:02.03tzangerI never saw the joy in vanity 800
00:02.19Ariel_DonX, there is a sample extension setup with DISA on the wiki
00:02.25RCAMtzanger Well for product support and sales it's great.
00:02.32SkramXDonX, not sure about the Dialtone.. but I can give you code for Authenticate, then set var $num, and then dial IAX2/username:password@youroutboundcarrier/$num
00:02.40tzangerRCAM: that's not vanity
00:02.42tzangerthat's utility
00:03.00Strom_Cdial tone is an easy kludge - just play a recording of dial tone
00:03.03RCAMtzanger Well they call it vanity.
00:03.09SkramXtzanger..I have used nufone.net, iax.cc and asterlink.com for 800 inbound.. I like asterlink.com the most.
00:03.20Ariel_argh bad play recording of a dial tone
00:03.20*** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka)
00:03.22Ariel_why
00:03.23SkramX( #asterlink you can talk to some of the guys who work there)
00:03.27SkramXhaha
00:03.27RCAMSkramX I have beenusing Nufone.
00:03.29tzangerSkramX: that's good, coz' the guys who run that lurk here
00:03.42SkramXOk.
00:03.43Maxx4lifeI like iax.cc for 800 inbound
00:03.45RCAMseen jerjer
00:03.50Ariel_nufone is great for outbound and 800 number service
00:04.03DonXSkramX that would be great
00:04.05JunK-Yhere's the my best pics of cluecon (and trip in chicago) http://www.midsouthmarketplace.com/~krice/gallery/albums.php  , a great thanks to ken for hosting.
00:04.05RCAMAriel_ It has been pretty good.
00:04.10Ariel_~seen jerjer
00:04.13jbotjerjer is currently on #asterisk (5h 25m 6s).  Has said a total of 17 messages.  Is idling for 45m 18s
00:04.19RCAMO yeah.
00:04.21Netgeeksanyone here do some "VoIP handholding" in the SF Bay area?
00:04.30Ariel_Netgeeks, yes I do
00:04.54SkramX~seen SkramX
00:04.54jbotskramx is currently on #asterisk.  Has said a total of 12 messages.  Is idling for 0s
00:04.58Ariel_I have a few customers there and one of my partners is in South SF location.
00:05.03SkramX~seen skram|
00:05.04jbotskram| <n=SkramY@cpe-70-112-81-84.austin.res.rr.com> was last seen on IRC in channel #asterisk, 8m 39s ago, saying: 'Nugget, was 'kaus' random or are you in the Cen-Tex area?'.
00:05.08_ioscannerDonX I am in DFW too
00:05.10SkramXok.
00:05.16*** join/#asterisk invisblewoman (n=foobar@67.71.159.87)
00:05.23SkramXI am in t.eh 512, Represent.
00:05.34SkramXDonX do you want some code for it or have you figured it out?
00:06.53glm2khmmm ./clidtest /dev/zap/1 just sits there....
00:08.25tzangerhey is Hmmhesays here?
00:08.39ataraxisjust want to share a happy life sign: Accepting data call from '74' to '0815' on channel 0/2, span 1 :)
00:08.45Hmmhesaysmaybe
00:08.48Hmmhesaysdepends on who you ask
00:09.07*** part/#asterisk debian (n=foobar@67.71.159.87)
00:09.25tzangerHmmhesays: my gf thinks you're hot
00:09.35Hmmhesaysheh, thanks
00:09.37*** join/#asterisk debianE (n=foobar@67.71.159.87)
00:09.47JunK-YdebianE!!!
00:09.48Hmmhesaysumm... as long as by "my gf" you don't me "I"
00:09.52RCAMWhat is the best control panel software for Asterisk these days? IPSwitchboard?
00:09.54tzangerHmmhesays: hahaha no
00:10.08Hmmhesaysnothing wrong with that, I just don't swing that way
00:10.13Hmmhesaysexcept for bkw_
00:11.10ManxPowerWell, it was a 2 hour conference call, but it was a good 2 hour conference call
00:11.46RCAMCan asterisk easily be set to record all calls by default?
00:12.14ManxPowerRCAM: "show application monitor"
00:12.24tzangerRCAM: yes
00:12.25ManxPowerRCAM: you just have to be sure to run Monitor before ANY Dial line
00:13.19RCAMAnything better than Diax for IAX Softphone?
00:13.40SkramXIAXComm is what I think is the best
00:13.50SkramXworks on Mac, Unix, Windows.
00:13.51RCAMSkramX What makes it better?
00:13.53SkramX*Windows.
00:13.59RCAMSkramX That does help.
00:14.02ManxPowerAll softphones suck.
00:14.07RCAMSkramX Open source?
00:14.09ManxPowerYou will understand that when you get a hardphone
00:14.11SkramXNice and simple and compatible with many OS's
00:14.19SkramXManxPower: Indeed.
00:14.25SkramXRCAM.. im pretty sur
00:14.26SkramXe
00:14.30*** join/#asterisk jeffi1 (n=Jeff@toronto-HSE-ppp3943426.sympatico.ca)
00:14.31SkramX~google IAXComm
00:14.33SkramXoh
00:14.53RCAMManxPower O I Know... But I have people all over the place without hardphones.
00:15.07SkramXhttp://iaxclient.sourceforge.net/iaxcomm/
00:15.14SkramX:D
00:15.38RCAMSkramX Gotta love SF.
00:15.50SkramXYeah..
00:16.06Ariel_I found if you are going to setup a softphone you need to get a good usb headset/mic setup for it to work correctly.
00:16.27Ariel_since most laptops sound card and mic suck.
00:16.46SkramXRCAM I am going to try out DIax.
00:16.48SkramXNever heard of it.
00:17.22RCAMSkramX I had about 20 people using it in a virtual call center for 4 months last year. It was great.
00:17.29Ariel_diax is nice and works small setup but sometimes a little confusing.
00:17.32SkramXCool, Cool.
00:18.02RCAMWhat kills me about Diax is that it's not open source... and Windows only.
00:18.27Ariel_RCAM, yes that is correct it's not open source
00:18.42*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-209-191-131.dsl.sfldmi.ameritech.net)
00:19.00SkramXSee yall.. priv msg me if youneed anything.
00:19.10RCAMBye.
00:19.49ManxPowerI discovered a really silly Polycom error
00:19.56ManxPoweror bug or problem or whatever.
00:20.17ManxPowerIf the polycom cannot do a DNS lookup for the configured server names it will lock up after a few moments
00:20.31*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
00:20.39ManxPowerWe spent about 45 mins on this one.
00:21.37Ariel_ManxPower, yes I have seen that problem as well
00:21.55ManxPowerAriel_: We are did a VLAN test tonight.  Worked pretty good.
00:22.04*** join/#asterisk Beccara_ (n=tristram@222-152-13-41.jetstream.xtra.co.nz)
00:22.07Ariel_it happens more on boot up as well. it can hang there if it does not find the dns server.
00:22.25*** join/#asterisk supaigtr (n=yurplsl@152.53.16.10)
00:22.29supaigtrHello.
00:22.52Ariel_we did some test with eithereal and found the reason there would hang some times on boot up was trying to find the dns.
00:23.07Ariel_seems they did not build a good error out in there firmware
00:23.21supaigtrAnyone know how to make bindaddr with multiple ips?
00:23.43Ariel_supaigtr, like bindaddr=0.0.0.0
00:24.06supaigtrWell that used to work till I added vlan support.  Now 0.0.0.0 picks the non-vlan interface.
00:24.13*** join/#asterisk |cleric| (n=dacleric@p5482B1B7.dip0.t-ipconnect.de)
00:24.46supaigtr:(  It seems the code that get the all ip list didn't account for vlan.
00:25.14supaigtrIs there a way to specifiy multi like 192.168.1.1,192.168.1.2?
00:25.16Ariel_supaigtr, take it out there leave it out of the config file
00:25.33ManxPowersupaigtr: that would be a bug in Linux's networking code.
00:25.34supaigtrHuh??
00:25.58ManxPowersupaigtr: since bindaddr=0.0.0.0 makes asterisk to the equiv of "bind to all IPs" networking call.
00:26.20ManxPowerIf it's not binding to the right IPs, I would assume it's an OS issue.
00:26.21supaigtrManxPower:  I belive you but apache seems to work.
00:26.38ManxPowersupaigtr: Asterisk does a lot of weird shit. 8-)
00:26.56supaigtrIs there support for a multi ip list anyone knows about?
00:27.14supaigtrYea tell me about it. This new echo card is throwing a fit.
00:27.17ManxPowersupaigtr: I believe CVS-HEAD supports it.
00:27.41supaigtrI have CVS-HEAD as of about 2 hours ago.  2.6.12.4 and 2.6.9 tried
00:28.17ManxPowersupaigtr: how are you determining what IPs Asterisk is bound to?  netstat -an?
00:28.43SkramXFor somereason head kept on shitting out on me.. and I had to re-start asterisk every hour or so, im on Stable..
00:28.48supaigtrIf i do 0.0.0.0 it goes one place if i specified it goes to the other.
00:28.59ManxPowerSkramX: I use 1.0.x as well.
00:29.03SkramXyea.
00:29.04NuggetI recall talk of asterisk mishandling bindaddr=0.0.0.0 when there aren't different physical interfaces.  It's been months since it was discussed, but someone in here reported problems when the multiple IPs were aliased to the same physical interface.
00:29.21ManxPowersupaigtr: rtp.c has historically had issues with multiple IPs
00:29.31NuggetI've never had a problem, even back with 1.0.1, but I've always had multiple physical interfaces.
00:29.43ManxPowerNugget: in other words "search the mailing list archives"
00:29.50ManxPower~mailinglist
00:29.50jbothmm... 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/
00:29.50Nuggetsomething like that, yeah.  :)
00:29.52supaigtrNugget: Same boat here but VLANs are now in the picture.
00:30.29ManxPowersupaigtr: I just started with VLANs (never needed them before) and I'm already starting to dislike them.
00:31.00supaigtrVLAN"s are cool. Just forces me to write things down carefully.
00:31.33ManxPowersupaigtr: we are putting a polycom phone and a PC on the same switch port.  That was interesting.
00:32.06ManxPowerthey needed to be in different vlans
00:32.22ManxPoweron a 5509 switch running catos with an external router.
00:32.26supaigtrpoly supports vlan.  Computer on 1 phone on 2
00:32.43*** join/#asterisk Kumbang (n=ecvs@167.205.24.5)
00:32.49*** part/#asterisk Kumbang (n=ecvs@167.205.24.5)
00:32.49ManxPowersupaigtr: Actually, phone on the configured vlan and the PC traffic is just passed thru
00:33.06ManxPowerThen the switch had to tag traffic that's not 802.1q into the correct VLAN
00:33.09*** join/#asterisk Kumbang (n=ecvs@167.205.24.5)
00:33.16supaigtrDefault 1 phone 2
00:33.32Kumbangguys, how to disable callpickup on zap channel
00:33.38supaigtrworks good here.  Have like 100 or so networks.
00:33.57Kumbangis there option callpickup=no on zapata.conf
00:33.57ManxPowersupaigtr: since we never ever set up a vlan before......
00:34.13supaigtrIt took me about a week to get it.
00:34.32ManxPowerI tried and failed to get VTP working between the router and the switch that we called in a 2nd consultant and we got it working in less then 2 hours.
00:34.37supaigtrOnce I realized what tags where it got alot easier
00:34.51Moc___I gota find a good powerfull cheap switch
00:35.01supaigtrWe were setting up old 3coms and the UI sucked.
00:35.09ManxPowersupaigtr: My problem is that Cisco calls any port that supports more than 1 vlan a "trunk port".  Stupidest name I ever heard.
00:35.11supaigtrDELL = Powerful cheap.
00:35.18ManxPowerTo me TRUNKS ONLY GO BETWEEN SWITCHES.
00:35.26supaigtrSame with 3 com
00:35.47supaigtrI actually like dell and have grown quite fond of the Netgear 7312
00:35.47ManxPowerKatty: that's just SICK.
00:35.56twistedwheeee
00:35.57twisteddrugs
00:36.00KattyManxPower: i wonder about the psychology of those people
00:36.03ManxPowerI'm starting to wonder if people have "Brand Karma".
00:36.05twistedhydrocodone makes one loopy
00:36.20ManxPowerevery single netgear switch we have bought was thrown away since they kept locking up.
00:36.26niZonanyone used a good wifi voip phone lately?
00:36.54Kattytwisted: mrow?
00:36.59niZoneep?
00:37.00twistedKatty, i have a hip problem :/
00:37.01supaigtrManxPower: same here with the exception of the 7312. Linux and has VLAN routing
00:37.12twistedKatty, pretty severe if the doc diagnosed it correctly
00:37.23Kattytwisted: can you be fixed with hugs? :<
00:37.25twistedso i'm heavily medicated/sedated
00:37.32Kattyi see.
00:37.43*** part/#asterisk Kumbang (n=ecvs@167.205.24.5)
00:37.44Kattymishehu: whatsammater?
00:37.46twistedKatty, i doubt fixed, but hugs always improve things
00:37.52mishehuhow long does it take asus to get me a frickin replacement hsf for this motherboard?
00:37.55Kattytwisted: i don't hug medicated males.
00:37.55twistedKatty, it's not contagious..
00:38.02mishehunow it's all fucked and won't boot.
00:38.03Kattytwisted: it could be Dangerous
00:38.22twistedKatty, nah.. this stuff just makes me feel like i'm floating
00:38.24Kattymishehu: yeah, i don't like getting jerked around by companies either.
00:38.34twistedbut makse pain go byebye
00:38.40mishehuKatty: over two weeks ago they told me it would be about two weeks for the replacement.
00:38.52Kattymishehu: :<<<<
00:38.57mishehuit's a shame I can't find the right size fan here, I'd have replaced it myself already.
00:39.23twistedaww, no love from Katty ;.(
00:39.46Hmmhesaysheh
00:39.50Kattykatty doesn't lurve on people
00:39.58Hmmhesaysyou're like my favorite damn disease
00:40.03twistedHmmhesays, ;)
00:40.10twistedKatty, meh?
00:40.11Hmmhesaysyo
00:40.26Hmmhesaysi had to decline asteria twisted
00:40.35Hmmhesaysbecause I don't like you
00:40.38twistedHAH
00:40.50Kattytwisted: what's confusing about my statement?
00:41.23KattyHmmhesays: put your arm back on dude, and stop bleeding on the floor.
00:41.38Hmmhesaysfine FINE.
00:41.43Kattykthx
00:41.43Hmmhesays0_o
00:42.13Hmmhesaysrumple mints?
00:42.35Kattythey want to cut this guy's hair on tv..the How Do I Look thing from style network
00:42.39Kattyit makes me want to pout.
00:42.55Hmmhesayshttp://www.deesnider.com/members/images/3Dee/3DeeImage300.jpg
00:43.04Hmmhesaysthat should put you over the top
00:43.11Ariel_Katty, well you could always watch cartons
00:43.46Kattyhmm, i guess
00:45.07twistedHmmhesays, that was scary.
00:45.17Ariel_lumpy = pooh hefalump movie
00:45.22Hmmhesaysthe result of a face melting guitar solo i'm afraid
00:45.34Hmmhesaysthey ran rampant in the 80's
00:45.46twistedHmmhesays, yeah, I know...  but still :P
00:45.54KattyHmmhesays: grow your hair out
00:46.08Hmmhesaysi'm bald now Katty
00:46.46Ariel_hair loss is a way of life it shows your smart.
00:47.09twistedAriel_, haha... my grandpa must be an uber-genious
00:47.12HmmhesaysI decided scaring the stripper was the last straw
00:47.44Kattyaww, they just cut the hair off :<
00:47.57KattyHmmhesays: you did what?
00:48.16Hmmhesays"i'm afraid i'm going to impale myself on your hair"
00:48.30Kattydid you really shave it all off?
00:48.38Hmmhesayspfffft
00:48.40Hmmhesaysof course not
00:48.59Kattyphew.
00:50.20*** join/#asterisk AYANO (n=erik@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
00:50.22twisteddoes 1/4" count as cutting it all off?
00:50.33Katty1/4" doesn't count as hair
00:50.38twistedoh
00:50.51twistedbut I still shampoo it!
00:51.01Kattyk
00:51.23AYANOshampoo is always a plus.  :P
00:51.30twistedoooh... hydrocodone is kicking in
00:51.35twistedmust lay down now
00:51.48Kattybye
00:52.04KattyHmmhesays: grow hair out plskthx
00:52.19Ariel_twisted, what did you do to your self. (what's your pain)?
00:53.54Moc___woohoo.. my loan for my appliance are over... woohoo hehe
00:54.12*** join/#asterisk RoyK (n=roy@142.80-203-78.nextgentel.com)
00:54.13Hmmhesaysmotorcycle time
00:54.22Hmmhesayslater
00:54.36*** part/#asterisk pur-angst (n=fool@adsl-66-72-91-74.dsl.applwi.ameritech.net)
00:54.37*** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net)
00:54.38Kattyk
00:54.50KattyMoc___: yay!
00:55.03*** join/#asterisk Defraz_ (n=t0tal@tim.ibccom.net)
00:55.24Moc___that mean I'll have my new camera len this year... + more geek stuff..
00:55.31Katty:>>
00:55.46AYANOlol
00:57.28*** join/#asterisk kajtzu_ (n=kajtzu@shell1.fi.basen.net)
00:58.08litagewould asterisk be considered a "floating point intensive" app?
00:58.19Moc___litage, depend what you do..
00:58.31Moc___transcoding is floating point intensive... I think..
00:58.54tzangerMoc___: depends on the codec
00:58.58tzangerif it's a floating point codec then yes
00:59.00litageMoc___: there's no way of avoiding transcoding if you use an FXO/FXS/etc card, right?
00:59.03*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)
00:59.08tzangerlitage: use ulaw/alaw
00:59.18litagetzanger: which codecs are floats?
00:59.27Moc___yes ulaw and alaw is fine..
01:00.02litageulaw/alaw is float or non-float?
01:00.18Moc___ulaw/alaw are the less cpu intensive codec
01:00.33Moc___well signlear too.. I gues..
01:00.38MikeJ[Laptop]ulaw to alaw is a table lookup, not even math
01:00.55MikeJ[Laptop]that being said, I beleive the code in asterisk does not work properly
01:01.09Moc___MikeJ[Laptop], alot of code in asterisk doesn't work properly..
01:01.28MikeJ[Laptop]slin to ulaw is also just a lookup
01:01.38Moc___wonder if something like chan woomera could be used for Transcoding only..
01:01.43litageulaw and alaw are both 64kbps?
01:01.44MikeJ[Laptop]all the other codecs require some math
01:01.48tzangerslinear is just the raw audio.  ulaw and alaw are jsut "conversions" from 16-bit signed linear to 8-bit "companded"
01:01.55tzangercoppice would be able to explan better
01:02.02mishehubreaking the ulaw! breaking the ulaw!
01:02.09tzangerulaw/alaw isn't really a codec it's just a different-formatted slinear
01:02.12SkramXhaha mishehu
01:02.30MikeJ[Laptop]tzanger, you did a decent job....
01:02.35Moc___isnt ulaw a lower quality than slinear ?
01:02.43MikeJ[Laptop]well, it is diff...
01:02.51MikeJ[Laptop]but it is a direct corolation...
01:03.12tzangerMoc___: no
01:03.15*** join/#asterisk sambal_ (n=sambal@213.148.236.189)
01:03.17mishehuMoc___: I do not believe so
01:03.19tzangerMoc___: it's lossless
01:03.25mishehuit's just a formatting difference
01:03.37Moc___what is higher than ulaw ? PCM16 or something.. what is it call ?
01:03.40MikeJ[Laptop]I am not sure if that is true or not
01:03.41tzangerMoc___: basically it's companding the 16-bit signed to 8-bit unsigned, alaw is 16-bit signed to 8-bit signed (IIRC)
01:04.03tzangerMoc___: higher quality means sampling at better than 8kHz.  if you're dealing with FXO/FXS you'll never hear a difference
01:04.24tzangerulaw-slinear-ulaw will produce the exact same results no matter how many conversions
01:04.28tzangersame with alaw
01:04.31tzangerit's lossless and totally reverseable
01:05.13*** join/#asterisk Xeaded (i=Xeaded@69-88-216-229.thewavz.com)
01:05.57Moc___anyway...
01:06.04Moc___I use ulaw ;)
01:06.16mishehuI use ulaw internally
01:06.22mishehuexternally it could be speex, g729, or gsm
01:06.39*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) [NETSPLIT VICTIM]
01:06.43Moc___I loved g726 until I heard the huge noise it generate on high volume :(
01:06.51*** join/#asterisk j0sephi` (i=j0e@2001:6f8:99e:243:0:0:0:32)
01:06.57*** join/#asterisk Corndawg_ (i=whoisit@c-66-176-249-51.hsd1.fl.comcast.net) [NETSPLIT VICTIM]
01:06.58Moc___not noise but distortion..
01:07.17*** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net) [NETSPLIT VICTIM]
01:07.29tzangerI like g729 and gsm
01:07.41Moc___they both sound too digital :(
01:07.54*** join/#asterisk asasdaa (n=ticqoctz@203.167.245.16)
01:08.06MattLambHeya
01:08.15tzangerMoc___: compared to ulaw/alaw yes
01:08.21tzangerbut it's easy to get uysed to
01:08.23*** join/#asterisk dtpetty (n=petty@CPE000d8897ef6b-CM00407b878bdb.cpe.net.cable.rogers.com)
01:08.29MattLambI got a q
01:08.29tzangerhell music on hold with g729 is acceptable to me for MOST types
01:08.37MattLambsimple one
01:08.41Moc___g726 is perfect.... EXCEPT the distortion at high volume :(
01:08.42*** join/#asterisk rajo (n=rajo@bfs.cs.uni-sb.de) [NETSPLIT VICTIM]
01:08.48*** join/#asterisk l-fy (n=pchitesc@yate/developer/l-fy)
01:08.49MattLambHow do i access my voicemail?
01:08.50l-fyhello
01:08.58Moc___might be a asterisk g726 problem thought..
01:09.55*** join/#asterisk AYANO (n=erik@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
01:10.21*** join/#asterisk gravemind (n=omgwtfbb@freenode/developer/GXTi) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk kajtzu_ (n=kajtzu@shell1.fi.basen.net) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk bjohnson (n=bjohnson@i216-58-17-181.igs.net) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk Craziman2 (N=Craziman@boromir.apid.com) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk nwhit (N=nwhit@ns1.whittrio.com) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk nextime (N=nextime@213-140-6-96.ip.fastwebnet.it) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk jdiskywlkr (N=kvirc@ip68-0-90-1.tu.ok.cox.net) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk _mwoodj_ (N=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk iswm (I=iswm@unaffiliated/iswm) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk Saaib (N=nabudoco@ns1.ensenada.gob.mx) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk Luke-Jr (N=luke-jr@user-0c938q3.cable.mindspring.com) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk MajestiK (N=MajestiK@S0106000ea6572b5f.ed.shawcable.net) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk ecto (N=ectospas@69.85.202.242) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk marv (N=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk Silik0n (N=krice@newrso.suspicious.org) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk ManxPower (N=eric@dsl-209-16-67-160.datasync.com) [NETSPLIT VICTIM]
01:11.14*** join/#asterisk _hannes (N=hannes@port-212-202-55-197.dynamic.qsc.de) [NETSPLIT VICTIM]
01:11.18*** join/#asterisk Himeko (n=himeko@S01060040ca128fc3.ed.shawcable.net) [NETSPLIT VICTIM]
01:11.20*** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox) [NETSPLIT VICTIM]
01:11.20*** join/#asterisk j0sephi (I=letmeebo@2001:6f8:99e:243:0:0:0:32)
01:11.20*** join/#asterisk flynux (I=notqrw1@cl-8.bru-01.be.sixxs.net) [NETSPLIT VICTIM]
01:11.20MattLambheya?
01:11.33*** join/#asterisk xADDY (n=xAD_nFL@host144-199.pool8290.interbusiness.it) [NETSPLIT VICTIM]
01:11.36MattLambi got a newbie qustion, i cant figure out how to get my voicemail
01:12.07Moc___MattLamb, you need to create an extention that call the right application with the rights paramaters
01:12.09SkramXman.. there have been atleast 2 netsplits today..
01:12.29l-fySkramX > you know people are trying to make things stable
01:12.33l-fybut  is a lot of work
01:12.51l-fy~h323
01:12.51jbotwell, h323 is An ITU-T standard for packet-based multimedia communications systems. This standard defines the different multimedia entities that make up a multimedia system - Endpoint, Gateway, Multipoint Conferencing Unit (MCU), and Gatekeeper - and their interaction. This standard is used for many voice-over-IP applications, and is heavily dependent on other ...
01:12.51MattLambOh ok. Is there a page that can show me how?
01:12.57Nugget~docs
01:12.57jboti 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
01:12.59l-fy~openh323
01:12.59jbotH.323 protocol stack. URL: http://www.openh323.org/
01:13.08*** join/#asterisk DonnieMoss (n=Craziman@boromir.apid.com)
01:13.18SkramXOh, I know.. mad props to the freenode peoples and contributors, just saying its been more than usual lately
01:14.19*** join/#asterisk Sedorox (n=Brandon@UFO.client.wlgrv.pa.sed6.net)
01:14.32NuggetI don't see why freenode deserves "props".  They beg and demand payment for a service which plenty of other networks happily provide for free (and often with better reliability)
01:15.04l-fyNugget > you are free to not use freenode
01:15.09*** join/#asterisk BoNaDs (i=THEPLOTE@pool-68-162-3-185.nwrk.east.verizon.net)
01:15.15Nuggetnot until digium decides to go somewhere friendlier.
01:15.32mishehul-fy: and he is free to voice his opinion.
01:15.36Nuggetbut I'm sure not here because I want to help keep lilo from becoming homeless.
01:15.43*** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox)
01:15.46BoNaDsi just upgraded to cvs-head and after rebuilding i get this
01:15.51BoNaDs[chan_zap.so] => (Zapata Telephony w/PRI)
01:15.51BoNaDs[root@asterisk1 src]# Ouch ... error while writing audio data: : Broken pipe
01:15.51BoNaDs[chan_zap.so] => (Zapata Telephony w/PRI)
01:15.51BoNaDs[root@asterisk1 src]# Ouch ... error while writing audio data: : Broken pipe
01:15.57*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it)
01:16.03BoNaDswtf did i hose?
01:16.17*** join/#asterisk _hannes (n=hannes@port-212-202-55-197.dynamic.qsc.de)
01:16.19*** join/#asterisk kajtzu (n=kajtzu@shell1.fi.basen.net)
01:16.20mishehuBoNaDs: did you clear out all of your previous modules?
01:16.22*** join/#asterisk flynux (i=f85avst@cl-8.bru-01.be.sixxs.net)
01:16.27mishehuand do you also realize how instable HEAD is?
01:16.28l-fyBoNaDs > did you upgrade the zaptel and libpri libs also?
01:16.38BoNaDsmishehu: i ran the rm_astsrc  shell script 1st
01:16.40*** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net)
01:16.49BoNaDsbut i didnt delete any compiled binaries
01:16.52mishehuI'm not familiar with that script.  heh.
01:16.56BoNaDsdo i need to wipe them too?
01:16.57mishehuI just make my own packages all the time.
01:17.04*** join/#asterisk brookshire (n=matt@206.15.88.228)
01:17.19mishehuBoNaDs: I make packages, so nothing stale is left over.
01:17.46brookshirelinuxworld is hot :)
01:17.53brookshireand cold if you walk outside :)
01:17.56*** join/#asterisk mhnoyes_ (n=mhnoyes@user-2ivfmv5.dialup.mindspring.com)
01:18.02*** join/#asterisk syle (n=blah@wnpgmb06dc1-33-192.dynamic.mts.net)
01:18.11BoNaDs[root@asterisk1 src]# asterisk -r
01:18.11BoNaDsUnable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?)
01:18.23BoNaDsbut i think thats cause asterisk isnt running
01:18.34*** join/#asterisk Saaib (n=nabudoco@ns1.ensenada.gob.mx)
01:18.56*** join/#asterisk nwhit (n=nwhit@ns1.whittrio.com)
01:18.56*** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-90-1.tu.ok.cox.net)
01:18.56*** join/#asterisk iswm (i=iswm@ip68-0-136-205.tc.ph.cox.net)
01:18.56*** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net)
01:18.57BoNaDsdo i need to kill files other than the ones in /usr/src/zaptel & asterisk etc before i download from cvs and compile?
01:20.53*** join/#asterisk santiago (n=santiago@63.245.86.153)
01:21.21BoNaDsand if i shouldnt use cvs-head which one should i grab?
01:22.28*** part/#asterisk angom (N=Administ@red-corp-200.38.16.10.telnor.net)
01:22.54*** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk iswm (i=iswm@ip68-0-136-205.tc.ph.cox.net) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-90-1.tu.ok.cox.net) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk nwhit (n=nwhit@ns1.whittrio.com) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk flynux (i=f85avst@cl-8.bru-01.be.sixxs.net) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk kajtzu (n=kajtzu@shell1.fi.basen.net) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk _hannes (n=hannes@port-212-202-55-197.dynamic.qsc.de) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk bjohnson (n=bjohnson@i216-58-17-181.igs.net) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk _mwoodj_ (N=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk ecto (N=ectospas@69.85.202.242) [NETSPLIT VICTIM]
01:22.54*** join/#asterisk ManxPower (N=eric@dsl-209-16-67-160.datasync.com) [NETSPLIT VICTIM]
01:22.58*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
01:23.02Moc___wb everyone
01:24.04*** join/#asterisk Luke-Jr (n=luke-jr@CPE-65-26-132-140.kc.res.rr.com)
01:26.46*** join/#asterisk kajtzu (n=kajtzu@shell1.fi.basen.net)
01:27.01brimstoneBoNaDs: you don't need to remove /usr/src/asterisk or zaptel unless you want to make sure you have the latest and gratest
01:27.12brimstoneand be totally positive that the old is gone
01:27.19BoNaDsthe thing seems to compile fine
01:27.21BoNaDsit just wont run
01:27.21*** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net)
01:27.28brimstoneand if you don't use HEAD, then use "cvs -r v1-0 asterisk zaptel" etc
01:27.39brimstoneBoNaDs: what's the error when you run "asterisk -c" ?
01:27.44BoNaDsv1-0 will give me 1.0.9?
01:27.58brimstoneBoNaDs: v1-0 will give you cvs stable, which currently is 1.0.9
01:28.13*** join/#asterisk Umaro (n=umaro@209.140.74.64)
01:28.14*** join/#asterisk Xeaded (i=Nafis@69-88-216-229.thewavz.com)
01:28.15BoNaDs-c gives me
01:28.16BoNaDs[ Booting...................Warning, flexibel rate not heavily tested!
01:28.16BoNaDs....[root@asterisk1 src]# Ouch ... error while writing audio data: : Broken pipe
01:28.39brimstoneok, the error is prolly in a config file then, run "asterisk -vvvc" and use pastebin
01:29.13BoNaDsok sec
01:29.21BoNaDsi had 1.0.7 before
01:29.55blitzragegood eve all
01:30.05brimstoneword up blitzrage
01:30.11Qwellblitzrage: y0
01:30.21blitzrageQwell: hey hey!
01:30.31MattLambwell im stumped. How do I set it up so i can get my voicemail messages using AMP?
01:30.52blitzrageMattLamb: ummm... don't use AMP? :)
01:30.56*** join/#asterisk SplasPoo1 (n=jwb@brooklyn.paravolve.net)
01:31.00litagewhat advantages are there for getting 2x1GB ram sticks rather than 5x512MB?
01:31.04brc_anybody know the default password for a polycom 600?
01:31.14BoNaDsbrimstone: http://pastebin.com/333380
01:31.16*** join/#asterisk flynux (i=3otqr0t@cl-8.bru-01.be.sixxs.net)
01:31.20brc_litage, none
01:31.26MattLambBlitzrage: Ok. how do i do it manually?
01:31.30Nuggetwith 2x1gb you'll have three slots open for three more 1gb sticks later.
01:31.32mishehulitage: saving space is about it
01:31.53blitzrageMattLamb: use the Voicemail() application
01:32.07blitzrageerrr.... VoicemailMain()
01:32.13brimstoneBoNaDs: which zap hardware do you have? and does ztcfg -vv run without errors?
01:32.22blitzragebrc_: isn't it just Polycom ?
01:32.25BoNaDsi have 2 TDM04b cards
01:32.39blitzragebrc_: oh wait.... maybe its 432
01:32.40*** join/#asterisk _hannes (n=hannes@port-212-202-55-197.dynamic.qsc.de)
01:32.48MattLambhmm aplication.
01:32.51brimstoneBoNaDs: something's off with zaptel*.conf
01:33.10brimstonemake sure that each file has the right signalling set before the channel that it's suppose to affect
01:33.43BoNaDsztcfg finds 8 channels but says
01:33.44BoNaDsZT_CHANCONFIG failed on channel 1: No such device or address (6)
01:34.02brimstoneBoNaDs: ok, your module isn't loaded
01:34.16BoNaDswhich module? zaptel?
01:34.17brimstonemodprobe -vv wctdm or wcfxs or whatever
01:34.58BoNaDsok i ran modprobe on wctdm
01:35.09BoNaDsbut asterisk still wont load although ztcfg works
01:35.28brimstoneok, so ztcfg -vv ran peachy keen?
01:35.38BoNaDsyessir
01:35.42brimstonecheck /etc/asterisk/zapta*.conf
01:35.51BoNaDsasterisk -cvv ends on this new error now
01:35.52BoNaDs[pbx_wilcalu.so]Ouch ... error while writing audio data: : Broken pipe
01:36.18brimstoneBoNaDs: you acn remove /usr/lib/asterisk/modules/pbx_wilcalu.so
01:36.30brimstonei don't think maybe poeple use it anymore
01:36.32BoNaDsjust delete it?
01:36.53brimstonesure
01:37.04brimstoneif you really need it, run make install in asterisk source directory again
01:37.31*** join/#asterisk flynux (i=efkxqnw@cl-8.bru-01.be.sixxs.net)
01:38.12mishehuBoNaDs: I already suggested that you dump all the old modules and reinstall the new ver
01:38.26*** part/#asterisk MattLamb (n=ticqoctz@203.167.245.16)
01:38.45brc_blitzrage, thanks
01:39.01brimstoneBoNaDs: prolly wouldn't be a bad idea either, what mishehu said
01:39.17BoNaDsyeah trashed that module and we're all good now
01:39.19BoNaDsdanke
01:39.24brimstoneno prob
01:39.37BoNaDswe're trying to eliminate echo on our pots calls
01:39.43BoNaDsits a real pain in the ass
01:40.02brc_wtf
01:40.03BoNaDsis wctdm much better than wctfx?
01:40.06brc_I can't connect to cisco's ftp
01:40.12brimstoneBoNaDs: heh, tell me about it
01:40.18brimstonei troubleshoot echo all day :/
01:40.44BoNaDswe got echo to go away on calls we make out over pots
01:40.53BoNaDsbut those settings made it 100 times worse on incoming calls
01:41.11ManxPower"Ouch ... error while writing audio data" is a message from mpg123
01:41.11ManxPoweryou MUST have EXACTLY mpg123 v0.59r
01:41.11ManxPowerIf you don't want to load a module put noload => my_module.so in /etc/asterisk/modules.conf
01:41.12BoNaDsso i figured i'd upgrade to 1.0.9 and see what happens
01:41.22*** join/#asterisk _hannes (n=hannes@port-212-202-55-197.dynamic.qsc.de) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk Umaro (n=umaro@209.140.74.64) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk kajtzu (n=kajtzu@shell1.fi.basen.net) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk iswm (i=iswm@ip68-0-136-205.tc.ph.cox.net) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-90-1.tu.ok.cox.net) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk nwhit (n=nwhit@ns1.whittrio.com) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk bjohnson (n=bjohnson@i216-58-17-181.igs.net) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk _mwoodj_ (N=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk ecto (N=ectospas@69.85.202.242) [NETSPLIT VICTIM]
01:41.22*** join/#asterisk ManxPower (N=eric@dsl-209-16-67-160.datasync.com) [NETSPLIT VICTIM]
01:42.49blitzrageBoNaDs: wctdm == wcfxs, just been renamed to avoid confusion when the FXO modules came out for the TDM400P cards
01:42.54*** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net)
01:43.31BoNaDshave any of ya'll tried fxotune?
01:44.00brimstoneBoNaDs: i've seen fxotune do crazy things before
01:44.12brimstonetakes about 5min a fxo to run
01:44.18brimstonecrazy as in, work
01:44.26BoNaDshm
01:44.34BoNaDswell cross your fingers cause i'm trying
01:44.38BoNaDshehe
01:44.53brimstonecrossed fingers can't type well
01:44.54BoNaDsi used -i 1 does it matter what i put after the -i?
01:44.57brimstoneand the mouse is just out of the question
01:45.02brimstoneshouldn't
01:45.17BoNaDsit initially told me "could not fill input buffer"
01:45.22BoNaDsand now started Dialing...
01:45.25BoNaDsover and over
01:45.44*** join/#asterisk _mwoodj_ (n=mwoodj@24.96.145.218) [NETSPLIT VICTIM]
01:45.48*** join/#asterisk bjohnson (n=bjohnson@i216-58-17-181.igs.net) [NETSPLIT VICTIM]
01:46.01tuxinator_linuxMbye bye people
01:46.05*** join/#asterisk Umaro (n=umaro@209.140.74.64) [NETSPLIT VICTIM]
01:46.05*** join/#asterisk iswm (i=iswm@ip68-0-136-205.tc.ph.cox.net) [NETSPLIT VICTIM]
01:46.05*** join/#asterisk fanguin` (n=user@p548F146F.dip0.t-ipconnect.de)
01:46.32*** join/#asterisk nwhit (n=nwhit@ns1.whittrio.com) [NETSPLIT VICTIM]
01:46.45*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it) [NETSPLIT VICTIM]
01:47.36jsaundersAnyone encountered this?  chan_sip.c:10709 reload_config: Section 'macro-stdexten' lacks type
01:47.48*** join/#asterisk ecto (n=ectospas@69.85.202.242) [NETSPLIT VICTIM]
01:48.34dijungalhello
01:48.40dijungalwhat port does iax2 run on...?
01:49.05*** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) [NETSPLIT VICTIM]
01:49.08dijungali can call my iax clients but when i connect i do not hear anything...  i think my client's are on the wrong port
01:49.16*** join/#asterisk _DAW (n=bob@68-114-110-210.dhcp.slid.la.charter.com)
01:49.16dijungalmy clients are using port: 4569
01:49.21dijungalfirefly
01:49.27BoNaDsthe audio would be RTP
01:49.33BoNaDsnot iax
01:49.36dijungaloooh
01:49.50BoNaDsthink you need to open ports 10000-2000
01:49.59BoNaDser 10000-20000
01:50.01file[desk]IAX2 does not use RTP to transport audio
01:50.06BoNaDsno?
01:50.11dijungalnope
01:50.17dijungalfrom what i just read i don't think it does
01:50.25BoNaDsSIP does though right?
01:50.29file[desk]SIP yes
01:50.34file[desk]SIP is the signalling, RTP is the audio stream
01:50.39file[desk]IAX2 combines them all into the same
01:50.58*** join/#asterisk marv (n=ilovekim@pcp01529782pcs.huntsv01.al.comcast.net) [NETSPLIT VICTIM]
01:51.07*** join/#asterisk _hannes (n=hannes@port-212-202-55-197.dynamic.qsc.de) [NETSPLIT VICTIM]
01:51.25dijungalso what u think my issue is..?
01:51.58BoNaDsdunno
01:52.10BoNaDsi had the same issue using SIP accross nats
01:52.16fearnorproblem between chair and keyboard probably
01:52.20BoNaDshehe
01:52.26*** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw)
01:52.26*** mode/#asterisk [+o bkw_] by ChanServ
01:52.46dijungalmy sip word out great... but someone was like.. "IAX is beter" so now i'm trying IAX... it sucks.. :| it's not working for me
01:53.10bkw_Brain Surgery Sucks!
01:53.12fearnoriax is better in mark's head only.
01:53.16fearnorwell him and jerjer. :P)
01:53.18bkw_haha
01:53.25*** join/#asterisk MajestiK (n=MajestiK@S0106000ea6572b5f.ed.shawcable.net) [NETSPLIT VICTIM]
01:53.27bkw_you better don't get me started
01:53.32fearnorpersonally, iax is a waste of time
01:53.33fearnor:)
01:53.39bkw_file I have the res_perl and /etc/asterisk on the drones in CVS now
01:53.50bkw_so /usr/local/res_perl and all that jazz is in CVS w00t
01:53.55file[desk]bkw_: yay
01:54.06bkw_I have to smack tony around tommorow and ask him some questions about stuff
01:54.32Kattyfile[desk]: :<
01:54.38dijungalIAX is supposed to handle NAT traversals very well
01:54.44file[desk]:O
01:54.49bkw_in an ideal world sure
01:54.59dijungaland is supposed to be more eficient than sip
01:55.05bkw_it is not
01:55.08bkw_so not
01:55.09BoNaDsmy ZyXel wifi phones seem to hate NAT
01:55.14BoNaDshehe
01:55.17dijungallol'
01:55.28fearnora) it isn't more efficient, that's for sure
01:55.34Qwellhappiness ~= silly string
01:55.38fearnorb) SIP and NAT traversals is a solved issue
01:55.47Qwellfearnor: It is when you trunk
01:55.52fearnoryes, sip isn't all that nice to nat but it can be worked around.
01:56.06bkw_yes yes
01:56.10bkw_preach it brother fearnor
01:56.13fearnorqwell: whoever thinks that IAX saves bits does not know about IP header compression.
01:56.17bkw_tell it like it is
01:56.44fearnorin short, when you are doing headar compression, your ip/udp overhead is 8 bytes total.
01:56.48bkw_IAX will never scale as long as you try to reinvent the wheel of what the kernel can do.
01:56.53dijungalso mark does not know what he's talkin abut here: http://www.voip-info.org/wiki-IAX+versus+SIP
01:57.14fearnoropinions differ, dij.
01:57.17blitzrageNAT isn't a problem if you understand what NAT is and how it works
01:57.20dijungalindeed
01:57.29Qwellblitzrage: and/or choose not to use it ;]
01:57.34bkw_blitzrage, indeed
01:57.37blitzrageQwell: whatever works :)
01:57.44bkw_the rule is.. use what works
01:57.47bkw_the right tool for the job
01:57.48blitzragethose who have problems with NAT don't understand IP fundamentals
01:57.49*** join/#asterisk szi (n=rm@57-82.207-68.tampabay.res.rr.com)
01:57.50bkw_that sorrta thing
01:57.54fearnorhrm
01:57.56bkw_asterisk DOES NOT fit all places and installations
01:58.00bkw_it just doesn't
01:58.02*** join/#asterisk mr_bill (i=nobody@ohno.mrbill.net)
01:58.03fearnorhttp://www.voip-info.org/wiki-IAX+versus+SIP
01:58.04Qwellbkw_: LIES!
01:58.05Qwell:P
01:58.08fearnor#1 ignores header compression
01:58.13mr_billquick question - is a sound card required for asterisk anymore?
01:58.14Moc___bah
01:58.15fearnorwith header compression, IAX is only marginally efficient
01:58.20bkw_mr_bill, never had to
01:58.37dijungaland IAX2 :)
01:58.38mr_billif I can pull the sound card and save an IRQ, i will
01:58.41bkw_IAX will not scale past a handfull of calls
01:58.42blitzragemr_bill: it was never required
01:58.42fearnor#2 yes, sure. but then again, H323 is IE-encoded, not ascii. and that's not a good thing.
01:58.44mr_billcool
01:58.47mr_billthanks
01:59.00file[desk]it's easy to read SIP traces, one thing I like :P
01:59.04fearnor#3 is correct. that's why MGCP > SIP
01:59.10blitzragemr_bill: its only required if you want to use the CONSOLE channel (i.e., use Asterisk as a fancy phone with a headset)
01:59.18*** join/#asterisk Superman (n=yarick@h-68-167-104-244.mclnva23.covad.net)
01:59.19fearnor#4 yes, correct
01:59.28fearnor#5 is implemented in SIP too
01:59.35mr_billblitzrage: cool.  I've already got a BT101, two ATA188s, etc, and nothing will run locally on the machine, it wont even have a monitor
01:59.35bkw_MGCP aka Megaco aka h.248
01:59.40fearnor#6 well, remote-party-id is the answer
01:59.51file[desk]I like MGCP
01:59.56fearnor#8 can be done as OPTIONS
02:00.07fearnor#9 OOB DTMF is not necessarily a feature
02:00.11mr_billthanks for the quick help 8-)  I'll idle for a while and read while I swap cards in and out
02:00.13Kattyhmm.
02:00.13fearnor*required* oob that is
02:00.16file[desk]I should get an MGCP phone one of these days
02:00.30fearnorfor example, my customers would *hate* if i fuck with their rtp stream in any way or shape
02:00.35blitzragefile[desk]: 7960 with MGCP firmware?
02:00.38fearnoranyway.
02:00.43dijungalhmm...k
02:00.48file[desk]blitzrage: it was hell to get it to SIP, I'd prefer to leave it
02:00.52dijungalthanx... fearnor
02:00.56blitzragefile[desk]: :D
02:01.04file[desk]in fact it think it's a combo SCCP and SIP phone right now
02:01.15Qwellfile[desk]: Should've done the loader :p
02:01.28file[desk]and for a period of 3 hours it was a good brick
02:01.28fearnorand the most important thing from that comparison...mark seems to compare SIP as implemented in asterisk vs IAX
02:01.38fearnorobviously, SIP as implemented in asterisk is broken++
02:01.45dijungaloooh
02:01.54dijungalwhere is sip implemented properly..?
02:02.00dijungalfearnor...
02:02.02fearnorcause *cough* someone doesn't know what a proper FSM is
02:02.13bkw_SER does sip correctly
02:02.19file[desk]almost too correctly
02:02.22bkw_yep
02:02.29bkw_asterisk does it to try to be more compatible
02:02.30fearnorwell, lots of open-sores SIP stacks that are not fundamentally *BROKEN*
02:02.32dijungalSER...
02:02.33bkw_which causes issues
02:02.34dijungal?
02:02.41bkw_www.iptel.org
02:02.47fearnorwww.openser.org, bkw
02:03.08fearnornew folks taking over development from fraunhofer
02:03.24file[desk]whichever, they both work and are almost the same
02:03.38fearnoryes, sorta like zebra/quagga thang :)
02:03.52*** join/#asterisk val^bbs (n=nunya@71-8-97-108.dhcp.slid.la.charter.com)
02:03.57fearnorser is old and busted, cause authors trying to pimp out a commercial version
02:04.13file[desk]old and busted?
02:04.29fearnoropenser is the new hotness
02:04.30fearnor:)
02:05.52val^bbsAnyone dabble with Cisco 7960 phones before?
02:06.00bkw_yes
02:06.08bkw_thats like saying "do you guys have iron in your blood?"
02:06.13val^bbsMine is being weird
02:06.15fearnorhah
02:06.19bkw_:P
02:06.26bkw_anyway lets have a group hug
02:06.33bkw_because in the end we must all work together!
02:06.35val^bbsthe **# will not unlock it
02:06.45file[desk]yes sir!
02:06.46bkw_because you need to do settings 9
02:06.47brc_*back*
02:07.22val^bbscorrect and the password is hosed
02:07.33val^bbssigh*
02:07.44val^bbsthere is no way to reset the firmware on the phone is there?
02:07.46recis away
02:07.50bkw_the password is cisco by default
02:07.57val^bbsI tried that too LOL
02:08.03bkw_if it can find the tftp you can set the password
02:08.08bkw_in the .cnf file
02:08.26val^bbsit cant find the ftp LOL
02:08.38Qwellhard reset 'er
02:08.47val^bbshow do you do that?
02:08.52Qwellgot me
02:09.20BoNaDsif fxotune says "could not fill input buffer" before it starts "Dialing" does that mean it is effectivly not doing anything?
02:11.04mr_billI hope a P2/500 and 256M RAM is enough for decent performance.. I've only got one POTS line and only one IP phone will be active at any given time
02:11.48JerJermr_bill: just make sure asterisk runs with high priority
02:11.55JerJerand don't let the box to much else
02:12.33mr_billits dedicated to asterisk, and its going to run headless (no GUI, etc)
02:12.48*** join/#asterisk X-Rob (n=rob@dsl-202-173-151-24.qld.westnet.com.au)
02:13.08JerJergood
02:13.14JerJerenable dma on the hd
02:13.21JerJerdon't share irq
02:13.24*** join/#asterisk MikeJ[Laptop] (n=ircatjer@adsl-69-209-191-131.dsl.sfldmi.ameritech.net)
02:13.39JerJerand minimize your asterisk modules - for best performance
02:13.53mr_billso is digium not selling the X101P anymore?
02:13.57JerJerno
02:14.50_ioscanneranyone know where the docs are to convert sccp to sip on cisco 7912G phones?
02:14.54_ioscannerI can't seem to find it
02:14.56*** join/#asterisk SarahEmm (n=sarahemm@MTL-ppp-159032.qc.sympatico.ca)
02:15.18BoNaDsbbl
02:15.21*** part/#asterisk BoNaDs (i=THEPLOTE@pool-68-162-3-185.nwrk.east.verizon.net)
02:15.50SarahEmmhihi
02:16.15mr_billoh, youre supposed to buy a TDM400P + X100M instead I guess
02:16.35SarahEmmsivana: you around? *bugs you* ;)
02:16.39SarahEmmx100M?
02:17.20mr_billhttp://store.yahoo.com/asteriskpbx/tdtd1pofxsbu.html
02:17.24mr_bill$133 isnt bad at all
02:17.25szihey is anyone good at writing ivr systems using asterisk
02:17.33Sedoroxmr_bill, that $7... ebay?
02:17.35szii had some ivr specific questions
02:17.45JerJerSedorox:  you get what you pay for
02:17.47szii'm trying to figure out if something is going to be possible
02:17.51JerJerszi: ask them
02:17.54SarahEmmwhat's an x100m?
02:17.59sziok basically
02:18.04SedoroxI'm just wondering if its another one of thost motorola shit fxo's
02:18.08SarahEmmprobably
02:18.11SarahEmmthe ones that almost never work?
02:18.17mr_billsedorox: nope.  this is just for testing though.  if it works and I can get my wife to start using an ip phone, I'll get the real hardware from digium
02:18.17SarahEmmand have half the functionality stripped out if they ever do? ;)
02:18.17NetgeeksHey folks, I'm having all of the asterisk prompts (except for things like the demo messages) re-recorded by a female voice with a slight british accent (at the request of a client).  I have 9 words left in the price package and I'm done.  Any one got any suggestions (prepares for some awful ones)
02:18.30SedoroxSarahEmm, I think that its just the module.. either the FXS or FXO...
02:18.39SarahEmmahhhhh
02:18.45mr_billsarahemm: x100M is the FXO module for the TDM400P
02:18.47mr_billthe red module
02:18.48szii have to connect two parties, there is a customer and an agent and the agent must be dialed out (called back and presented with an ivr menu like "press one to accept, two to decline..." then i need to connect both the customer and agent together
02:18.48SarahEmmahh okay
02:18.53SarahEmmthat makes sense
02:18.59JerJerNetgeeks: Alison can do a british accent
02:19.01Sedoroxmr_bill, ok... the intel based clones seem to work fine.. there is one floating around called like X101P on ebay.. its based on motorola.. doesn't work...
02:19.04mr_billnetgeeks: "bollocks"
02:19.19mr_billsedorox: nope these are intel v.92 cards
02:19.21szii can't figure out how to connect the two ends of the calls when one end is dialed outbound
02:19.23Sedoroxkk
02:19.32Sedoroxis it the right chipset tho? :p
02:19.35NetgeeksBullocks added
02:19.38JerJerszi: use the manger interface
02:19.55sziwell i'm using the perl agi library to create the call flow
02:20.02sziand basically i have a call in progress
02:20.06mr_billyeah.  it gets recognized as an X101P
02:20.08sziand i need to create another outbound call
02:20.10*** join/#asterisk ericm (n=ericm@c-24-9-147-88.hsd1.co.comcast.net)
02:20.14szithen somehow get them together
02:20.15sziheh
02:20.23SarahEmmszi: uhh... Dial()?
02:20.30szithat's not enough
02:20.32NetgeeksJerJer: I personally don't like Allison's voice all that much, so I'm taking the option of someone else.
02:20.38SarahEmmbecause?
02:20.44sziunless you know some way to use Dial and play the callee an arbitrary ivr menu
02:20.53SarahEmmerr
02:20.58SarahEmmdial with the M option to use a macro
02:20.58mr_billnetgeeks: british slang and curse words
02:21.04SarahEmmthat'll let you play stuff to the callee
02:21.12SarahEmmor do whatever else you want with the callee
02:21.12szithis is all implemented in perl
02:21.15SarahEmmuhh
02:21.21SarahEmmwell this would be implemented in the dialplan itself..
02:21.27SarahEmmi dunno in perl
02:21.30*** join/#asterisk RCAM (n=rcammobi@232-35.26-24.tampabay.res.rr.com)
02:21.31sziso i mean i guess i could setup a macro that would call antoher agi script
02:21.33RCAMHowdy.
02:21.33Netgeeksmr_bill:  I've already had all prompts with 'ppund' double recorded , one with pound, one with 'hash'
02:21.33supaigtrAnyone have the new ecan card?
02:21.37SarahEmmyeah szi true
02:21.42Netgeekspound even
02:22.02sziwell it was either that or i was going to try to conference the parties together
02:22.14szii also saw that asterisk has ACD features but they aren't very dynamic
02:22.17sziat least not in 1.09
02:22.21*** part/#asterisk dijungal (n=ovr@host-216-226-255-89.interpacket.net)
02:22.24*** join/#asterisk tengulre (n=tengulre@61.185.238.166)
02:22.31szii've had shit for luck with the HEAD revision
02:22.35SarahEmmi like head
02:22.45Netgeeks...
02:22.48szimy agi scripts don't work with HEAD
02:22.53sziit's bizzare because they execute
02:22.54SarahEmmszi: ahh
02:23.00szidoes HEAD do any chroot()ing?
02:23.01NetgeeksI don't know if they will record that phrase for me Sarah
02:23.02SarahEmmyeah there are some differences
02:23.08SarahEmmNetgeeks: rofl
02:23.11szii thought of that as a possibility
02:23.14SarahEmmNetgeeks: *fwaps gently*
02:23.23*** join/#asterisk jaxxan (n=Snak@202.70.125.109)
02:23.26szii'ts crazy but i mean so far i'm enjoying *
02:23.36sziit's just little things i have problems with that don't seem obvious to me
02:23.41szior seem like they should be more obvious
02:23.56SarahEmmi like CVS HEAD
02:24.01sziheh
02:24.13sziwell girls can like head too :)
02:24.26mr_billCVS gives good HEAD
02:24.27mr_bill8-)
02:24.31szihah
02:24.33szitotally dude
02:24.36*** join/#asterisk Darwin35 (n=kvirc@ip70-179-214-245.dl.dl.cox.net)
02:24.43sziyeah so i've had problems with AGI and the HEAD revision
02:24.49szii've also on occasion had problems compiling it
02:24.54sziwhere it doesn't compile at all
02:24.55sziheh
02:24.57sziit's b0rked
02:25.01*** join/#asterisk Maticks (n=reece@whatda.whatthe.net)
02:25.07sziseems to depend on the day
02:25.10*** join/#asterisk toddf (n=toddf@net-66-210-111-62.theshop.net)
02:25.12Matickshi pplz
02:25.12SarahEmmhi lilo!
02:25.21szii've been mainly using 1.09 in debian apt
02:25.37mr_billare there any other free voice packages (recordings) for asterisk?
02:25.38szii'm a debian whore
02:25.48Maticksdoes anyone know what causes this i can find nothing on mailing lists about it
02:25.48MaticksAug 10 12:24:04 NOTICE[146]: chan_iax2.c:2455 iax2_read: I should never be called!
02:25.49szii saw some in various languages
02:25.52SarahEmm<-- gentoo grrl
02:25.52szilike french and german
02:25.54mr_billa british voice sounds nice
02:26.24szihrrm
02:26.33Netgeeksokay, I threw in "This call provided by Asterisk Open Source Telephony Server" to finish it off...
02:26.36lilohey SarahEmm
02:26.44szisarah i like gentoo, i just don't like waiting  8 hours for gnome to compile
02:26.44szihaha
02:26.54SarahEmmszi: i use distcc so all the machines in the house share the compile load
02:26.57sziactually nmow i'm not using gnome so i guess that doesn't matter much
02:27.06sziyeah i supposed you'd have to do that really
02:27.19szii have my own debian based distro
02:27.26sziso i have to say nice things about debian
02:27.28szihaha
02:27.33*** join/#asterisk shidan (n=shidan@CPE0013107d30c4-CM014280007905.cpe.net.cable.rogers.com)
02:27.38Netgeeksmr_bill: this is the voice I'll be using  http://www.shockwave-sound.com/previews/v/Voices%20Custom/kate-demo.mp3
02:27.42*** join/#asterisk ericm (n=ericm@c-24-9-147-88.hsd1.co.comcast.net)
02:28.05sziSarahEmm, well its' funny you mentioned the macro thing because that was my first option i was going to try that and failing that trying to build a 2 node bridge of some kind
02:28.14szifailing that i was going to see about using dynamic queueing
02:28.17sziwhich i thought would be overkill
02:28.51SarahEmmgnome-- anyway :)
02:28.53SarahEmm<-- vtwm grrl
02:29.05szii also considered parking but there wasn't really any way to do that programmatically to my satisfaction
02:29.19sziyyeah i mainly use ion3 these days
02:29.35sziit's alot more efficient when you write as much bloody code as i do
02:29.37szihaha
02:32.23szime and a friend of mine figured out a way to implement the entire AGI API in like 12 lines of ruby
02:32.29sziit's pretty crazy
02:33.15sziyou could only do that in ruby
02:33.16szihaha
02:34.20_ioscanneranyone upgrade 7912 to sip before
02:34.36_ioscannerit is not the same as 7940/7960
02:34.54_ioscannerI can't seem to find what needs to be in the XMLDefaults.cnf.xml file
02:35.11mr_billthanks for the help everyone
02:35.13*** part/#asterisk mr_bill (i=nobody@ohno.mrbill.net)
02:35.56*** join/#asterisk Beccara (n=tristram@222-152-13-41.jetstream.xtra.co.nz)
02:37.26ManxPowerOK.  $745.00 per night, two people.  I don't think so.
02:37.54ManxPowerWHY do these damn travel shows always showcase REALLY cool (and expensive) hotels?
02:38.14szino one would watch them if they showcased ghetto shit holes
02:38.33ManxPowerszi: perhaps showcasing something a tad more affordable
02:38.33*** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-90-1.tu.ok.cox.net)
02:39.01sziManxPower: like motel 6? hey at least they'll leave a light on for you
02:39.35ManxPowerszi: $300/night is plenty for a splurge weekend or vacation and it had better come with EVERYTHING
02:39.36Kattyfile[desk]: are you the very model of a modern major general?
02:39.50supaigtrAnybody have a minute to discuss echo?
02:39.52sziyeah i agree that 745/night is ridiculous
02:39.56SarahEmmkatty!
02:39.59SarahEmm*petpet*
02:40.03ManxPowerszi: and that's the mid range package.
02:40.10*** join/#asterisk mutilator (i=WebChat@i.think.napoleon.dynamiteblows.com)
02:40.11sziyou can rent a beach house for an entire week in clearwater for like another $200 on top of that
02:40.12Kattysarah!!!!!1111oneoneone
02:40.21SarahEmmlol
02:40.25Kattymine
02:40.28sziManxPower, stop watching crappy travel shows
02:40.40ManxPowerszi: but now I want to go there.
02:40.59ManxPowerhttp://www.konavillage.com/
02:40.59sziManxPower, yeah :~(
02:41.04szi*tear*
02:41.12szik
02:41.40sziaah well
02:41.44szihawaii what do you expect
02:42.09*** join/#asterisk Tili (n=Tili@219.136.217.55)
02:42.09sziwhat's going on this this channel
02:42.15szithere are chix purring and talking about distcc
02:42.17sziheh
02:42.18szii'ts crazy
02:42.24SarahEmmlol
02:42.26szipurrrrr distcc
02:42.27SarahEmmi like distcc
02:42.30SarahEmmand i like being petted
02:42.31SarahEmmmeow.
02:42.32szipurrrrr gentoo
02:42.34szihaha
02:42.43sziwr30rw
02:42.49szii love cats
02:42.53SarahEmm<-- is a kitrich
02:43.11mutilatorhas anyone here used a Cisco Catalyst 6000 24-Port FXS
02:43.42mutilatori was thinkin of gettin a TE110P and going pri to that.. then bunch of analog lines to apartments
02:44.02mutilatori assume that'de work?
02:44.19*** join/#asterisk nola-X (n=bdnola@ip68-11-88-171.no.no.cox.net)
02:46.27mutilatorusing?
02:46.36mutilatortype of hardware...
02:46.45supaigtrme?
02:46.51mutilatoryea
02:47.00Kattypretty kitrich.
02:47.12fearnoreliminate asstricks and echo will follow
02:47.19supaigtrOh.  I had a 410p and purchased a 411p and now I have more echo than before.
02:47.33fearnoror beg coppice to write a g168 echo canceller :P)
02:47.42mutilatortry messing with the gain on the pri
02:47.51fearnorsup: ya sure?
02:48.11fearnorwhats hanging off it?
02:48.13fearnorwhat channel bank etc
02:48.20supaigtrfearnor: I also get an occiosnonal popping sound.  A class 5 softswitch.
02:48.39fearnorno channel bank with analog phones?
02:48.48supaigtrIt  *->NFAS PRI - > softswitch
02:48.59fearnorumm
02:49.04fearnorthat's kinda strange
02:49.07supaigtrpolycom 600/500 -> * -> nFAS pRI -> software to be exact
02:49.16fearnorecho on which end?
02:49.20fearnorpolycoms hear echo?
02:49.22supaigtrpolycom only
02:49.33supaigtrPopping heard on both sides.
02:49.44fearnorwell popping is interrupt issues or timing issues
02:49.46fearnorfix that first.
02:49.50supaigtrI had a the non-echo 410 and it had less problems.
02:50.02fearnormake sure to time your card off your span
02:50.14fearnorcheck on your switch if there are any frame slips etc
02:50.23fearnorresolve interrupt issues (./zttest)
02:50.28supaigtrTiming is off span. I think its the ecan cause if I disable it disappears
02:50.48fearnorahh if *that* is the case, yeah
02:51.12fearnorimproperly designed echocan could do that
02:51.17supaigtrDoes the new ecan card just suck?
02:51.18fearnoron other hand
02:51.28opus_uhh, does realtime cache extension contextes?
02:51.31opus_Even after a shutdown?
02:51.32fearnoryou really *shouldnt need* echocan altogether
02:51.40opus_where does asterisk hold it 'cached' variables?
02:51.42fearnoropus: realtime does not cache anything at all.
02:51.47fearnoropus: asterisk database.
02:51.48supaigtrHow do I get there?  No ecan?
02:52.05fearnorsup: well...first, find where the issue is.
02:52.09opus_fearnor - i totally removed a context, then restarted asterisk, and it foudn it again
02:52.17opus_where is the asterisk database stored..
02:52.17fearnorgenerally if *you* hear echo, someone on remote end has bad echo can.
02:52.32fearnorsup: read cco.cisco.com there's a good explanation of it.
02:52.38supaigtrfearnor: local calls sound fine, some analog calls produce echo.
02:52.51fearnorsup: Yeap
02:52.58supaigtrI can't change the guy on the other end.
02:53.06supaigtrIt a customer sitting at home.
02:53.14fearnorask softswitch people to do echo cancellation
02:53.34fearnoror muck with tx/rxgain
02:53.50fearnoryes its kinda gay
02:54.04supaigtrWhat should tx/rx be???  I followed the directions on zttmonitor
02:54.07fearnorbut here's the thing, i think more and more people now will notice echo calling over very-long-distance
02:54.28fearnorsimply because latency will be >64ms and that's what standard echocans are designed to cope with
02:55.16fearnorif you hear echo from remote end, try rx 3 tx -3
02:55.43fearnorbut its kinda a hack and not proper way to fix it
02:56.45supaigtrI've used asterisk for about 6 months now and everyones about fed up with it.  Miss the old panasonics.
02:56.59fearnorHAH
02:57.24supaigtrecho, crashes, popping, no dss console...
02:57.41supaigtrMakes a nice VM system
02:57.43*** join/#asterisk hashu (n=root@64.252.54.126)
02:58.24supaigtrfearnor:  Is there a better way to do *?
02:58.29Sedoroxnight
02:58.33Kattynitenite
02:58.52*** join/#asterisk vuvie (n=vuvie@bb219-74-44-131.singnet.com.sg)
02:59.21_ioscannerAnyone know the format for XMLDefaults.cnf.xml to force default load of 7912.  I have it working for 7940/7960.  I can define by device but I would like a default method
03:03.26TerryMRi have cisco callmanager ... echo is still a PITA with some calls.
03:04.03Kattymmm, pita.
03:04.11TerryMRpain in the ass
03:04.16supaigtrTerryMR: What percentage?
03:04.45supaigtrI have echo on almost every call.
03:04.55supaigtrAccross the street even.
03:05.07supaigtrDidn't have it before.  
03:06.01X-Robmine does
03:06.07X-RobI have a wireless link across the street
03:06.12TerryMRany voip system will highlight echo problems because of the time delay ... not too many of our calls are affected but we still get some even with ciscos latest updates
03:06.12outtoluncnow if you said it's 'your network'
03:06.54*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
03:07.05outtoluncso obviously you tested this link before hand right?
03:07.23outtoluncsince it's just a wifi link 'across the street'
03:07.26TerryMRif it's on purely ip calls with no analog ... then it's a network problem
03:07.38outtolunchello
03:08.09outtoluncit has to cross the wifi what part of analog comes into play?
03:08.56outtolunci must be on the tail end of another convo, this one just doesn't make sense
03:09.06outtoluncafk
03:10.13TerryMRi was talking to supaigtr about his echo issue ... then x-rob jumped in with wireless links ... i think you thoguht i was talking wireless
03:10.34outtoluncnods
03:10.55outtoluncas i said, jumped in too late for it to make sense
03:11.12outtoluncso please disregard my comments
03:12.14outtoluncif you can sum it up in 1-2 lines i'd gladly listen
03:13.30*** join/#asterisk SkramX (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
03:13.51lilohi SkramX
03:13.58SkramXHrllo
03:14.07liloSkramX: there may be a few more netsplits.... we're working on stabilizing things
03:14.23liloSkramX: some packet loss on a main rotation hub earlier
03:14.34twistedyee haw
03:14.36twistednetsplits
03:14.39lilohey twisted
03:14.42lilohehe
03:14.42twistedhi lilo :)
03:15.03twisteddon't wince, i'm drugged
03:15.04TerryMRcould be hard to sum up ... I only jumped in when he said asterix sucks and is there a better way ... i was just offering my experience with cisco ... which can still have echo problems on pstn calls
03:15.08lilotwisted: oic 8)
03:15.10twistedpainkillers are working quite well :P
03:15.21lilotwisted: well, I hope you get better soon 8)
03:15.21supaigtrTerryMR: I was refering to TDM.
03:15.26twistedlilo, thanks
03:15.28liloouttolunc: I wish there was some way to do that 8)
03:15.32twistedmy hip thanks you too
03:15.43*** join/#asterisk shidan_ (n=shidan@CPE0013107d30c4-CM014280007905.cpe.net.cable.rogers.com)
03:16.01supaigtrShould I have echo from * -> PSTN via PRI -> pstn?
03:16.09twistedlol
03:17.17twistedlilo, just so you know, I'm digging the new server software
03:17.27TerryMRecho can is a fix for a pstn problem ... basically the echo can't come from your end becaue you're all digital ... its a problem with analog end points ... software/hardware should be able to filter it on most calls though
03:17.59outtoluncecho can 'still' happen with ip delay
03:18.22outtoluncor does the word 'sat' not register <G>
03:18.25supaigtrTerryMR: I had it down to around 10% then I purchased hardware echo which is up to 100%.  That 100% of PSTN -> analog is now echoing.
03:19.08TerryMRwhat are you using for phones, all analog or what ?
03:19.25ManxPowersupaigtr: what echo canceler did you buy?
03:19.36supaigtrThis is 30 phones on a VLAN with QOS going into a * box sitting right across from a class 5 switch via PRI with interconnects with carriers within 1200 feet of this building.
03:19.40supaigtrDigium
03:19.43supaigtr411p
03:19.58ManxPowersupaigtr: call digium in the morning.  They will support the card.
03:20.11supaigtrI called a few times.  No dice.
03:20.20outtoluncwhat other traffic is on that network?
03:20.20ManxPowersupaigtr: with a PRI, echo usually comes from the far end POTS line.
03:20.27TerryMRright ... ip phones can make the echo more noticable ... cisco  canceler did you buy?
03:20.27TerryMR<supaigtr> This is 30 phones on a VLAN with QOS going into a * box sitting right across from
03:20.27TerryMR<PROTECTED>
03:20.42ManxPowersupaigtr: then e-mail them.
03:20.43supaigtrWe upgraded to head and it got better but then popping started.
03:20.44TerryMRsorry about that ..hit the wrong buttin
03:20.54outtoluncnp
03:20.54TerryMRaggh button even
03:21.04AYANOtwisted:  you around?
03:21.10outtoluncwhat else is happening on that network
03:21.12twistedAYANO, somewhere
03:21.13TerryMRthanks ... need that
03:21.23supaigtremailed to.  Did it at 9:00am this morning got a response at 5:14pm.
03:21.35ManxPowersupaigtr: they are not exactly fast with their support.
03:21.57supaigtroutolunc: Its not the network.  Calls around are fine.  Currently there is almost 0 traffic.
03:22.19ManxPowersupaigtr: you recuced your txgain, I assume.
03:22.21outtoluncso drop the QOS
03:22.23supaigtrA local echo test sounds just fine.
03:22.24ManxPowerreduced, even
03:22.56supaigtrManxPower: I did that on the 410p which got me down to about 10% echo on calls.  Doesn't seem to affect the 411p.
03:23.40outtoluncbig Q: what else is running on that *box?
03:24.08supaigtrmysql
03:24.13supaigtrOh sorry
03:24.39outtoluncand what 'type' of interaction... how many queries per sec?
03:24.59outtoluncor how big is the table it's querying?
03:25.53outtolunci know that i have a 400k rec table that when it gets queried, asterisk moh does a hesitation
03:26.08outtoluncbut thats at multiple times a sec
03:26.35outtoluncso please offer whats really happening
03:26.53ManxPowersupaigtr: echo, if you have it, can be the hardest thing to solve.
03:27.02*** join/#asterisk mago2-cn (i=maxgluck@200.109.167.9)
03:27.24ariel_ECHO/echo ..e.c.h.ooooo
03:27.27mago2-cnhi, somebody know if it is possible to send calls to a callmanager 3.3(3)?
03:27.32supaigtrManxPower: I'm at the end. I'm just about to the point on install commerical pbx.
03:27.59TerryMRmago :  yes use h323 ... I have a range of extensions on callmanager that get sent to asterisk ... and the same going the other way
03:28.07lilotwisted: cool :)
03:28.16lilotwisted: what do you like about it? 8)
03:28.19*** join/#asterisk Newbie___ (i=me@211.24.146.12)
03:28.25mishehubah.
03:28.54Newbie___hi, with so many voip suppliers out there, dont know who to choose. any recommendation ?
03:29.02outtolunci like how the netsplits don't effect me <G>
03:29.02ManxPowersupaigtr: keep on Digium.
03:29.03mago2-cni'm being able to send calls from call manager to asterisk, but not to the call manager... is there any special config in the cm or asterisk? i have cm as a peer, phones in cm as users...
03:29.50ManxPowersupaigtr: keep dropping the gain until you can't go any lower without making it hard to hear people
03:29.53TerryMRmago - I just have a pattern  in my dial plan that gets sent to CM ... hang on I might be able to pull that part of the config
03:29.54mago2-cnto where should i dial from asterisk dialplan, to call manager or directly to users (sccp phones)
03:29.55supaigtrmanxpower: I plan to but there doesn't seem to be much out there on the new ecan cards.
03:30.18mago2-cnthanks terry...
03:30.20ManxPowersupaigtr: they are VERY VERY new, which is why you'll only really get help from Digium at this point.
03:30.36ManxPowersupaigtr: you may have just missed something really simple about enableing echocan on the card
03:31.14supaigtrManxpower: In theory its suppose to just work.  I see it attaching to channels in syslog and if using HEAD I hear very little echo. But the popping is a problem.
03:31.20outtolunche's running the new cards? i missed that part
03:31.21litagein high performance asterisk servers dealing with several hundred simultaneous connections, does having extra ram help, or is it not particularly needed for asterisk?
03:31.45outtolunc(cards/modules)
03:31.53Darwin35<===looking for a yob with asterisk
03:31.56supaigtrManxpower: They indicated if you see 8 lights and see things in syslog its working.
03:32.11twistedlilo, well, it SEEMS to be faster for starters ;)
03:32.25ManxPowerlitage: Asterisk is not a RAM hog.  Some versions (1.0.x) have memory leaks, however.  I put 512MB in a server and call it "overkill"
03:32.48jontowCPU power and disk space are more important, for sure.. imo anyway ;)
03:32.54litagethanks guys
03:32.57outtoluncmanx: he is running other stuff also
03:33.08TerryMRmago  : exten => _[2-6]XXX,1,Dial(oh323/${EXTEN:0}@10.1.40.1)    works for me ... any extensions 2000-6999 are sent to callmanager
03:33.09jontowCPU power only if you're trancoding.. responsiveness on the box is most important, and not running too many unrelated tasks helps..
03:33.13ManxPowerouttolunc: only a crazy person would do that.
03:33.21outtoluncwell <G>
03:33.23jontow(read: don't run X, apache/mysql get a little heavy at times)
03:33.37jontowand depending on the motherboard, SMP sometimes hurts more than it helps
03:33.42outtolunc[20:24] <supaigtr> mysql
03:33.43litageso would it be worth spending the money to on 3.6 rather than 3GHz CPUs?
03:33.46ManxPowerFor very small systems I usually share with other tasks, but nothing heavy
03:33.59outtoluncand that is just what he admitted too <G>
03:34.00jontowlitage; depends.. any transcoding involved? or all straight ulaw?
03:34.00TerryMRmago : make sure the calling search space is configured in callmanager for calls coming in from asterisk
03:34.04ManxPowerlitage: you going to use a server board or a desktop board?
03:34.07litagejontow: 'fraid i don't know at this time
03:34.19jontowthen get a faster CPU and save yourself some hassle/upgrading
03:34.31litageManxPower: i'm buying rackmountable servers, so server board i assume
03:34.35Darwin35res_sqlite
03:34.44Darwin35if you want a low load system
03:34.52ManxPowerlitage: in a server board I would try to get a dual CPU board and only buy 1 CPU.
03:35.00jontowmanxpower; good call.
03:35.05ManxPowerserver boards are expensive enough, a little more won't kill anyone.
03:35.34ManxPowerAnd don't worry too much about the CPU speed.  like 1 or 2 levels faster than "the fastest"
03:35.40ManxPower..er..
03:35.47ManxPowerAnd don't worry too much about the CPU speed.  like 1 or 2 levels SLOWER than "the fastest"
03:36.06ManxPowerbut the ACTUAL board you get IS important.
03:36.47litageManxPower: i'm going to be getting 2 Xeon MP CPUs in each machine, and thus have room for an additional 2 CPUs in each if they're needed
03:36.52jontowagreed, has anyone compiled a 'real' list of boards, graded in a sense on the way they work?
03:37.11ManxPowerlitage: Gads!  What are you doing, running telecom for a small country?
03:37.21jontowor is the wiki the most current, there?
03:37.22jontowhehe
03:37.33*** part/#asterisk Defraz_ (n=t0tal@tim.ibccom.net)
03:37.38litageManxPower: :)
03:37.48jontowi had a client once that wanted an asterisk system built.. hardware and all; when it came time to ask their budget they said "oh.. don't worry about it, we have plenty of money.. and we have a machine already"
03:37.48*** join/#asterisk Tili[Away] (n=Tili@219.137.40.89)
03:38.08jontowi immediately said 'oh god', to myself.. i was right, too.. was an ibm eserver x345 -- and it was crap
03:38.39jontowREALLY nice box, if there weren't hardware faults, it couldn't handle the interrupt load from a single T100P -- NMI issues and random reboots galore, plus some pretty badass echo
03:39.02jontowdual 3.2ghz xeon's with iirc 2MB L2 cache each, 4GB of RAM, 70GB RAID
03:39.17ManxPowerThe problem is related to the BIOS and/or chipset.  Many of the high end boards seem to lock interrupts for a very long time, causing HDLC Abort errors, screwing up faxes, distorting audio, and disconnecting calls.
03:39.24ManxPoweroh, and really screwing up echo.
03:39.47ManxPowerMostly Promise, some SATA, and some IDE controllers cause this.
03:39.49litageManxPower: 1+ machines with dual Xeon MPs are needed for several hundred simultaneous calls
03:41.38outtolunchell i'd take a dual xeon for just 2 t's
03:42.06outtoluncand 24 fxs
03:42.13Maxx4lifeouttolunc: thats serious over kill!
03:42.33outtolunchave you ran 2 t1's and 24 fxs (agents)
03:42.47ManxPowerwe run a 4-port board on a....Intel(R) Pentium(R) 4 CPU 1.80GHz stepping 03 with 512MB of RAM.
03:43.31Maxx4lifesrry just cam eback
03:43.43Maxx4lifeI've done the 2 T1's not the FXS side.
03:44.00outtoluncany moh? recording?
03:44.02litageMaxx4life: if you have a good # of customers, you need to be able to support several hundred simultaneous calls, so dual xeons are needed, don't you agree?
03:44.16ManxPowerThis card does a mixture of E&M/Wink, fxsks, fxoks, and DACS channels.
03:44.17Maxx4lifelitage: several hundred calls yes
03:44.25Maxx4lifelitage: I agree with that part
03:44.36litageMaxx4life: is there something you disagree with that i said?
03:44.49ManxPowerlitage: there's a big difference between corporate and service provider.
03:44.50Maxx4lifelitage: I don't agree needing it for 2 T1's...
03:44.57Maxx4lifeBut I've also never done the Channel Bank of things
03:44.57ManxPower<-- is a corporate geek.
03:45.27litageManxPower: how can one determine [before buying] whether their server board will provide interrupt evils?
03:45.44ManxPowerlitage: There is the great and mystical question.
03:45.54litageah
03:46.01ManxPowerlitage: you ask around for people running similar class hardware and get the EXACT model number of the board.
03:46.10ManxPoweryou read the mailing list archives too.
03:46.32*** part/#asterisk SarahEmm (n=sarahemm@MTL-ppp-159032.qc.sympatico.ca)
03:47.06litageManxPower: so if you buy a $5000 machine and it turns out to have these interrupt problems, you're borked?
03:47.38ManxPowerlitage: when that happens we "suddenly" need a non-telephony server somewhere ASAP.
03:47.48litagehahahah
03:48.08ManxPowerlitage: there's always a server somewhere we can justify upgrading.
03:48.27litageManxPower: so before buying, i need to find out the exact model # of the server's motherboard, and research that heavily? is there any other hardware that is sometimes finicky like this?
03:48.32*** join/#asterisk jdg (n=jdg@CA03F9CF.adsl.mana.pf)
03:48.51*** part/#asterisk jdg (n=jdg@CA03F9CF.adsl.mana.pf)
03:48.54*** join/#asterisk jpayne (n=jpayne@baconhouse.sackheads.org)
03:49.00ManxPowerlitage: you want my PERSONAL recommendations, which are heavly mixed with pain and heartbreak.
03:49.19ManxPowerIf I ever see another SuperMicro board in my life it will be too soon.
03:49.58litageboard == motherboard?
03:50.02ManxPowerlitage: you can prolly just avoid these issues by doing one of two things.  Use SCSI (Adaptec is what I'd use) card and a SCSI drive.
03:50.06litage(just want to make 100% sure)
03:50.10ManxPowerlitage: board == motherboard
03:50.18litagedanke
03:50.47litageManxPower: i was thinking of using 10,000rpm sata rather than 10,000rpm scsi
03:51.06ManxPowerAnd ANYTHING that says "high performance" or "server class" has got to be stealing time by locking interrupts for a long time to get lots done.
03:51.38litageManxPower: not sure what you meant by that last comment. would you mind explaining a bit more please?
03:51.45ManxPowerlitage: Again, my personal feelings is that SATA is too new and chipsets have not had time to iron out all the issues.
03:51.56litagehrmm
03:52.17ManxPowerlitage: people buy Promise RAID controllers for their Asterisk boxes and then get massive HDLC abort errors.
03:52.36ManxPowerthe controller is locking interrupts for long amount of time to "burst" data across the bus.
03:52.51litageah i see
03:53.25ManxPowerThe alternative to using SCSI would be to get a Sangoma card instead.
03:54.12ManxPowerNot all that many people have Sangoma cards w/asterisk so much less help on the net.
03:54.42mago2-cnterry: i'm dialing like this, please tell me if it something like you do: h323/1002001/ccm1
03:54.43ManxPowerThey are about the same cost.  Sangoma has not contributed much to the Asterisk codebase and so some people think they are just making money onDigium's coattails.
03:54.46litageManxPower: before production deployment, i'm going to test digium vs sangoma, but i've read an article or two that said sangomas result in significantly less load
03:55.27ManxPowerlitage: now, since I'm not a Sangoma expert, there very could be massive numbers of horrible problems with the Sangoma that I've just never heard about.
03:55.53ManxPowerlitage: ALWAYS have a backup plan.  ALWAYS.
03:56.18TerryMRmago2-cn :  my dial looks like  oh323/1000@10.1.40.1       ... i prefer to use the ip to save the dns lookup
03:56.28ManxPowerThe plan for when "oh god, the office opens in 2 weeks and the telco just told us it will be 6 weeks before the line is in" happens.
03:56.36litageManxPower: that's what the development and testing phases are for. if they don't go as planned, we'll go back and buy new, hopefully-supported hardware
03:57.02ManxPowerlitage: Ugh.  So many things crop up when you go into production that you never saw in testing.
03:57.45litageManxPower: yeah i know, so for the first couple of months, we're going to give our services away for free to X customers so that they can test for us, and in return, they save money on calls
03:58.55ManxPowerlitage: you might consider prototyping new services like that too.
03:59.08litageManxPower: what do you mean?
03:59.12litageoh right
03:59.19litagehahah brain fart
03:59.34ManxPowerWhat we do is talk to the top producing agents and anyone that leaves candy on their desk for MIS in the middle of the night gets asked if they want to participate in new services before we do a full rollout.
03:59.44litageManxPower: yup, cuz honestly, customers are the best way of testing your products and getting feedback
04:00.09litageMIS?
04:00.25ManxPowerManagement Information Systems aka IT.
04:00.31litageah
04:00.49litageManxPower: so these interrupt issues come from...the motherboard? scsi adapter? raid card? combination of these?
04:00.55ManxPowerif it's 10pm and I'm trying to deal with some issue, some a bowl of chocolate candy left on the desk really helps.
04:01.06litagehah touche
04:01.22ManxPowerlitage: who knows?  I have my "theories" but they are hardly scientific.
04:01.57litageManxPower: i would be interested in hearing your theories
04:02.09ManxPowermy theory is that it's the chipset.
04:02.28litagethe mobo's chipset?
04:03.09ManxPowerlitage: there was a thread on one of the asterisk mailing lists a while ago about the fact that many of the "known not good" motherboards for Asterisk are also "known not good motherboards" for any kind of high def sound recording.
04:03.21ManxPowerlitage: yup, the motherboard's chipset.
04:04.08ManxPowerHOWEVER, motherboard makers have a LOT of leeway in designing the rest of the board, so who really knows where it's coming from.
04:04.13litageManxPower: hrm, thanks for that tidbit about the sound recording. the parallels are obvious and make [uneducated] sense
04:04.32litages/uneducated/unresearched/
04:04.32ManxPowerlitage: *nod* seemed to make sense to me.
04:04.46*** join/#asterisk HellAgony (n=HellAgon@200.121.241.223)
04:05.21*** join/#asterisk The_Ball (n=alex@static-227.35.240.220.dsl.comindico.com.au)
04:05.59*** join/#asterisk Cresl1n (n=Cresl1n@user-24-236-124-147.knology.net)
04:07.27Cresl1nooh...
04:07.29Cresl1nquiet
04:07.49opus_whats the correct sytax for select into in sql
04:07.54litageManxPower: i'm strongly leaning towards a mainstream vendor for my rackservers (eg: dell, ibm, hp) because of their warranties and on-site hardware service. any comments about that?
04:08.33ManxPowerlitage: I don't do that, but there's nothing wrong with it, as long as you find someone with the exact same model with a working Asterisk setup
04:08.34litageopus_: SELECT X FROM theTable WHERE Y='something'
04:08.54ManxPowerNufone/JerJer uses Dells, for example.  And ONLY one specfic model, I think.
04:08.57opus_no, i mean select INTO theTable from secondTable
04:09.33litageManxPower: any reason why you choose to build your own servers rather than purchase them from a vendor?
04:10.06ManxPowerlitage: we have been using the same local reseller for *ages* and he does a good job.
04:11.09*** join/#asterisk _cleric_ (n=dacleric@p54829B09.dip0.t-ipconnect.de)
04:11.21litageManxPower: what made you decide to build your own at the very beginning?
04:12.09ManxPowerI think we just had crappy luck with name brand systems, expecially name brand server systems way back when they started putting servers in.
04:12.34litageManxPower: how long ago was that?
04:12.58ManxPowerWe tried using the Novell servers and turning them into Linux servers and all the name brands were a hassle to do so, and building our own worked better.
04:13.34ManxPowerlitage: I've been consulting for this company for around 10 years.
04:14.04ManxPowerAh!  No, at the time nobody thought IT was worth anything and so there was almost NO money for equipment (and they were MUCH smaller back then)
04:14.51litageheheh, how things change  =)
04:15.52ManxPowerlitage: It's a $550 mil/year company with 300 people and like 15 offices and their entire IT staff is a manager/tech, a tech, and a consultant.
04:17.48ManxPowerAnnoys me, but keeps me busy enough to make a living 8-)
04:18.18litagesounds good, ManxPower
04:18.31litagei'm the project leader for a voip startup
04:18.42litagecurrently in development phase
04:18.43*** join/#asterisk cgcorea (n=cgcorea@205.240.200.111)
04:21.04file[desk]yet another voip startup?
04:21.34ManxPowerlitage: if the company is not well funded, then run for your life.
04:23.40litageManxPower: luckily, it's well funded
04:24.07litageso ManxPower, you like...ass tricks?  =P
04:24.15ManxPowerBTW, MeetMe sucks.
04:24.24ManxPowerlitage: it's the quiet ones that are the strangest.
04:24.29litagemeetme?
04:24.41ManxPowerasterisk conferencing application
04:24.57litageah, just found it on voip-info.org
04:25.08*** join/#asterisk CunningPike (n=CunningP@S010600095b33697f.vc.shawcable.net)
04:25.31litageManxPower: so what's #asstricks for?
04:25.51ManxPowerlitage: gay social channel for asterisk people
04:25.54Corydon76-home~asstricks
04:25.54jbotwell, asstricks is #asstricks, the underground gay Asterisk channel. Be afraid, very afraid
04:26.13ManxPowerYeah, that too.  Thanks, Corydon76-home
04:26.44Corydon76-homeHeh
04:26.48litagehahah i see. does one HAVE to be gay (i assume homosexual) to participate in #asstricks?
04:26.58ManxPowerlitage: naw.
04:26.59litageor can one just be happy?
04:27.09file[desk]it helps
04:27.22Corydon76-homeNaw, a few of us are bi...
04:27.30Corydon76-homeThere's even a few token straights
04:27.42litagewhat about bi-curious?  =P
04:27.43ManxPowerWe make them wear funny hats.
04:28.18litageManxPower: it's funny you said that, cuz i've been joking with my boss that i'm going to get us all sombreros when we move into our new office
04:29.12Corydon76-homeThink "Cat in the Hat"
04:29.39litagehow about "hat in the cat"?
04:29.45litagehrmm actually that'd be too messy
04:32.51litageManxPower: so what hardware do you lean towards when building a server that'll be handling several hundred simultaneous connections?
04:33.11ManxPowerlitage: I don't.  None of our offices are even CLOSE to that big.
04:33.24litagephooey  =)
04:33.30*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
04:33.50*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1)
04:35.12*** join/#asterisk af_ (n=af@ip-142-5.sn1.eutelia.it)
04:36.32file[desk]B flat, D flat, and E walk into a bar - the bartender says, "you have to leave - we don't serve minors"
04:36.49*** join/#asterisk buddah (n=buddah@208.179.86.5)
04:36.55buddahhow do i restart a zap channel?
04:37.10litagefile[desk]: that was terrible
04:37.15Corydon76-homebah dum bum
04:37.26glm2klol
04:37.53litagefile[desk]: a mushroom walks into a bar and sits down. the bartender looks at him and says "we don't serve your kind in here". the mushroom says "oh come on! i'm a fun guy!"
04:38.00litagesounds better when it's said rather than read
04:38.22glm2kyou forgot to mention what _kind_ of shroom
04:39.07file[desk]The soprano, not being smart enough to use birth control, says to her saxophophonist lover,
04:39.13file[desk]"Honest, I think you better pull out now."
04:39.20file[desk]He replies, "Why? am I sharp?"
04:39.28file[desk]Honest = honey
04:39.35sskylesWelcome to Commedian Mail!
04:40.09glm2khmmm, does * have a jokes directory under sounds?
04:40.34glm2kshould be better than that sleepy MOH
04:40.49glm2kand would drive away waiting customers faster too.
04:42.17*** join/#asterisk tony__ (n=tony@66-191-69-132.dhcp.dlth.mn.charter.com)
04:44.12*** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM]
04:44.12*** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) [NETSPLIT VICTIM]
04:44.12*** join/#asterisk ecto (n=ectospas@69.85.202.242) [NETSPLIT VICTIM]
04:44.12*** join/#asterisk nwhit (n=nwhit@ns1.whittrio.com) [NETSPLIT VICTIM]
04:44.12*** join/#asterisk bjohnson (n=bjohnson@i216-58-17-181.igs.net) [NETSPLIT VICTIM]
04:44.12*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM]
04:44.12*** join/#asterisk Saaib (n=nabudoco@ns1.ensenada.gob.mx) [NETSPLIT VICTIM]
04:44.12*** join/#asterisk pnviking (N=pnlarsso@c83-248-2-153.bredband.comhem.se) [NETSPLIT VICTIM]
04:46.01litagedoes asterisk enjoy large L2 and L3 caches, or does it not really matter?
04:46.02*** join/#asterisk viLeR (i=1000@ip-47-252.telesat.com.co)
04:46.33dudeslitage - asterisk loves all it's hoes
04:47.22litagedudes: heh yeah, but asterisk benefits a lot more from faster CPUs than more RAM. so i was wondering if it makes much of a difference to have 1 or 2MB of L2 cache
04:47.35*** join/#asterisk _-Jon-_ (i=jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com)
04:47.49_-Jon-_Good evening everyone
04:47.58Cresl1nevening :-)
04:48.04jaxxanwhy do people chose skinny over sip ?
04:48.19jaxxanchoose even
04:48.23jaxxan!skinny
04:48.25file[laptop]because he's skinnier!
04:48.29Cresl1nskinny r0x0rs
04:48.30Cresl1n:-P
04:48.32file[laptop]less fattening
04:48.32litagecuz they like to sip their skinny coffee?
04:48.46sskylesTastes great! Less filling!
04:48.48Cresl1nor incessantly
04:48.48dudeslitage - a 2.8Ghz CeleronD 128k cache  are slower than a 2800XP Mobile /w 512k ... so I'd say it benefits from more cache.
04:48.54Cresl1nor something like that
04:49.14jaxxan!sip
04:49.17Cresl1nI finally found the ITU doc that has this message I'm trying to write
04:50.46*** join/#asterisk |Vulture| (n=V@76.233.204.68.cfl.res.rr.com)
04:51.26_-Jon-_I'm wondering if someone with a bit of free time could help me out with this configuration..  my dedicated server running asterisk answers the calls and then hands them off to an extention on a local asterisk box using IAX2..  Can someone help?  I've tried everything but the call just never gets to the 2nd box
04:52.19dudes_-Jon-_ - you couldn't have tried everything if it never gets tot he second box =0
04:52.28dudesbut do a iax2 debug
04:52.38dudesand then make the call and see why it's not being accepted
04:52.52_-Jon-_dudes, good point :P
04:53.13dudes_-Jon-_ - also do you have type=user type=peer in your iax.conf?
04:53.21*** join/#asterisk PakiPenguin (i=uppal@202.147.163.14)
04:53.24PakiPenguinhello everyone
04:55.06dudesthe computer being called should be the peer and the computer calling should be the user ... then call iax2/peer@user/${EXTEN}
04:55.06_-Jon-_dudes, i have type=friend
04:55.41_-Jon-_basically the entry in both iax.conf is the same except the [name] and host
04:55.50dudes_-Jon-_ - rather peer should the one calling and the user should be the one getting the call
04:56.16JerJerdudes:  that's bassackwards
04:56.30JerJerIAX2/user@peer/exten|timeout|options
04:56.41dudesI know read my last message
04:56.47_-Jon-_hmmm
04:57.06_-Jon-_can I replace user@peer with the [name] in iax.conf?
04:57.24JerJer[peer]
04:57.24dudesyea use the name in the context
04:57.26JerJertype=peer
04:57.35JerJeror type=friend if you are lazy
04:58.16glm2kthis is the perfect job for you then...too much time and too little money to buy a cisco >:)
04:58.43_-Jon-_haha
04:58.46dudesI'm poor but not lazy
04:59.24glm2kmy ex would disagree with you
05:00.06dudesthat's good
05:00.14glm2kshe once told me to get a loan from the bank and buy those bloody expensive pbx's that you plug in and start using...not spends nights debugging :)
05:00.37dudesnah * works just fine
05:01.14glm2kjust what i said...
05:03.03sskylesThose expensive PBX's require a phone guy to come in and set-up.
05:03.03dudesmy ex was easy ... good bless her soul
05:03.13dudesgod bless rather
05:03.24dudeswas is
05:04.03dudesI'm freakin tired tonight
05:04.24dudesWhere's the speed hiding
05:04.44dudesOh that's right the kid didn't have any ... j/k  
05:04.47*** join/#asterisk mago2-cn (i=maxgluck@200.109.167.9)
05:05.00*** join/#asterisk tessier_ (n=treed@ip-66-181-5-46.cust.i2bnetworks.com)
05:05.25mago2-cnbilling systems, recommendations please?
05:05.34JakBeatZAnyone know wha the best current method is for getting a zaptel dummy driver working for timing?  is it still zaprtc in the bristuff package or ztdummy or what?  I'm using 2.6.10
05:05.56glm2kegad, voip-info page for FXO's has been redirected to Voice Gateway: GB-16
05:06.41dudesJakBeatZ - gnudialer.org/releases --- we have the zconfig.h file setup to compile for ztdummy
05:07.08dudesso as long as you have everything else setup it will compile
05:07.43supaigtrmago2-cn: What type of billing?
05:09.22JerJeris the mailing lists dead again?  it seems to be awfully dead
05:09.32JerJermago2-cn:  write one
05:09.38dudesFucking windows 98 | get bill gates in here
05:10.09JerJermago2-cn: in the long run a canned solution written by someone else will never do exactly what you need it to do
05:10.19JakBeatZdudes: so I replace the existing one with yours and then make linux26 as per normal?
05:10.31dudesYes
05:10.45dudesand as long as you have the kernel stuff done correctly it will compile
05:10.52brc_meow
05:11.13*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
05:11.29dudesI used that config on our test servers which run 2.6.8 and 10
05:11.48supaigtrAnyone working on provisioning or have any magic echo pill.  You know the purple ones.
05:12.09JakBeatZdudes:  I had the kernel working with zaprtc in 1.0.9 then I went to CVS-HEAD And everything went to hell in a handbag so I assumed I needed to do something different so 1.0.9 implied rtc being disabled from the kernel which was the only requirment, iirc, no?
05:14.56dudesI don't kno w
05:15.25mago2-cnIn-house then Jer Jer?
05:15.34JakBeatZOk..  well, I tried your header and I keep getting this error when I startup *:  "Aug 10 01:02:05 WARNING[9246]: chan_iax2.c:9328 load_module: Unable to open IAX timing interface: Permission denied"
05:15.54JakBeatZafter I modprobe ztdummy
05:16.16JakBeatZthat's the problem I've been having ever since I went to cvs-head from 1.0.9
05:16.52JerJermago2-cn:  that's my own opinion, yes
05:17.27dudesJakBeatZ - Permission denied ...
05:17.35JerJeradd some permission
05:18.04dudesare you running asterisk as a user?
05:18.05JakBeatZdo what?  the module?  the /dev/zap/pseudo?
05:18.13JakBeatZya, running as user/group asterisk
05:18.22dudesJakBeatZ - well
05:18.31JakBeatZs/do what/to what/
05:18.48JerJersee asterisk.conf.sample
05:19.00_-Jon-_okay so I got my 2 boxes linked, i enable debugging and it shows nothing at all when I try and call
05:19.46dudes_-Jon-_  - post your config @ pastebin.ca
05:20.04JerJer_-Jon-_:  then a peer entry is wrong
05:20.18_-Jon-_JerJer, I have 2 friend entries
05:20.25_-Jon-_dudes, hold on, let me go there
05:20.41dudes_-Jon-_ - also you don't need to register or use usernames; passwords are a good idea though
05:21.38JakBeatZJerJer:  I
05:22.00JakBeatZJerJer:  I've checked all the dirs from asterisk.conf and they are all owned by the asterisk user
05:22.11JakBeatZnot sure if I'm understanding your comment properly
05:22.17SkramXanyone know of a ipkall.com like place... for IAX2??!
05:22.47_-Jon-_http://pastebin.ca/19673
05:22.58glm2kSkramX: FWD?
05:23.10SkramXipkall gives a free PSTN number
05:23.16SkramX~ipkall
05:23.21SkramX~ipkall.com
05:23.22SkramXhmm
05:23.29SkramXwell they do http://ipkall.com/
05:23.32glm2koh
05:23.40SkramXthey connect to any sip server.. i want one that can connect via iax.
05:23.41glm2kand htye only do sip?
05:23.48glm2ki see.
05:23.51SkramXglm2k: yes/
05:24.25dudes_-Jon-_ - try netxero@netxero ...
05:26.38_-Jon-_dudes, still nothing :(
05:27.01dudeshost=net-xero.net ... ??? put the IP's
05:27.02_-Jon-_wait..  what port does iax2 use?
05:27.13file[laptop]4569 UDP
05:28.46_-Jon-_kay I have solved the mystery
05:29.03_-Jon-_4569 wasn't being allowed in :P
05:29.12dudes_-Jon-_ - way2go dipshit =0
05:29.14dudeshehe
05:29.28JerJersupaigtr:  yes
05:29.41_-Jon-_hehe, my bad :P  I had the same problem with net-snmp, i should have known
05:29.57dudesDon't let it happen again or your're fucking tired
05:29.59supaigtrJerJer: U gotta solution for the studder in head with the 411p?
05:30.04dudesfired not tired ...
05:30.22JerJerwe don't have studder
05:30.40_-Jon-_hah
05:31.02supaigtrI get a studder with HEAD and the 411p card.  Take it out and put a 410 w head or 1.0.9x and it goes away.
05:31.29supaigtrI have a sound clip.  Nothing in the log and no errored secs on the span.
05:31.43_-Jon-_thanks for all yrou help help
05:31.45_-Jon-_*though
05:32.04supaigtrJerJer: we = digium?
05:32.40JerJerno
05:32.47JerJerwe = my company
05:32.48supaigtrwe = ?
05:33.07supaigtrSo you have the ecan versions with no problems?
05:33.22supaigtrIAX term?
05:34.25JerJerTE411P is what we run
05:34.29JerJeriax and sip
05:34.47supaigtrTE411P to a PRI with NFAS?
05:35.01supaigtrWhats the secret?
05:35.14JerJerno NFAS
05:35.22JerJerwe don't
05:36.10supaigtrAll sorts of wierd stuff here.  It seems to require head and head currently seems to have issues.
05:36.30dudessupaigtr - head is awesome!
05:36.41JerJerwe haven't had any issues with head
05:37.16dudesit's awesome
05:37.17supaigtrHad a working box for months and wanted to get rid of more echo.  Installed a 411p and crashola.  
05:37.28JerJerdebug the crash then
05:37.48dudesYou're running windows 98 ... Yup that's the problem
05:38.07supaigtrHow.  Nothing in the log no errored secs.  It just stops sending audio for a few secs then starts back up again working normally.
05:38.12JakBeatZdudes:  seems to work if I chown -R asterisk:asterisk /dev/zap
05:38.35JakBeatZis that the fix or is that just a workaround?
05:38.43supaigtrdudes: me
05:39.11dudessupaigtr - yes ... but I was only joking
05:39.42dudesI remembered a infomerical /w the Cleo and how she always say crap like "Yup, he's the daddy'>
05:39.44supaigtrdudes: no its on a coco3 running os-9. in color :)
05:40.09dudesYup that's your problem
05:40.16supaigtrThought so.
05:40.18dudesos-9 ...
05:40.56supaigtrback in the day. AKA joke.  coco  = Tandy Color Computer 3 512k mem (1 meg with hacks) running 6809 processor.
05:41.00*** join/#asterisk PhreeStyle (n=PhreeSty@cpe-24-221-52-165.az.sprintbbd.net)
05:41.03dudesyea
05:41.14supaigtros-9..  the only aftermarket os that was any good for it.
05:41.29dudesMy old man when I was (3) bought a tandy (and that was kick ass back then)
05:42.25supaigtrI got a coco 1 when I was around 5.  Rainbow magizine i believe was one of the groups for it.
05:43.03supaigtrHmm.  No popping.  something major change in head in last few hours I guess.
05:43.23dudesI got head like 7 days ago
05:43.35dudesthat's a good head ... I've had no issues with it
05:43.58supaigtrI've been battling this for weeks.  Either the echo or the popping.  People screaming everywhere.
05:44.14supaigtrGot PBX vendors coming outta the woodwork trying to get us to change.
05:44.25supaigtrdudes: what is your setup?
05:44.51dudesOne of our clients has 4 TE410P's in one box and no echo issues
05:45.13dudesBut I've never used a 411
05:45.14*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
05:45.17supaigtrMust not be calling ananlog PSTN much.
05:45.57dudessometimes
05:46.28dudesBut most the time they are connected to a dialer and allow them to use a SIP carrier to terminate calls on a dialogic dialer
05:46.47supaigtrI dont' get any echo on digital to digital. Not much even on internet termination.  But make a call to the backwoods guy on that 20,000 foot span of unbalanced copper and I have issues.
05:47.29*** join/#asterisk drumkilla_ (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
05:47.29*** mode/#asterisk [+o drumkilla_] by ChanServ
05:47.39dudesHave you tried some of the echo cancel shit that you can compile in from the zaptel drivers
05:47.57*** join/#asterisk drumkilla (n=russell@asterisk/developer-and-stable-maintainer/drumkilla)
05:47.57*** mode/#asterisk [+o drumkilla] by ChanServ
05:48.10*** join/#asterisk PhreeStyle (n=PhreeSty@cpe-24-221-52-165.az.sprintbbd.net)
05:49.13supaigtrYea. It does work turning on aggressive but it makes some noise itself on the connects on regular calls.  The hardware ecan seems to work well but something using it introduces almost like a flashook while talking sometimes.
05:49.42supaigtrI can't get it to do it now after getting updated head but it seems to resurface over time.
05:50.31dudesSo it only happens sometimes ... when calling certain numbers?
05:50.47dudessupaigtr - is it an echo or a crack
05:51.29supaigtrIts like a crack.  No audio for a sec or so afterwards then its back to normal.
05:51.50dudeshave you tried adjusting the gain in zapata config
05:53.02supaigtrI've tried manually doing it and doing it off a milliwat test. with the 411p it doesn't change. stays at the 14xxx marks.
05:53.29NetgeeksI didn't think gain was applicable to pri circuits
05:54.02Netgeekswhat motherboard/proc are you using?  Have you swapped out the power supply in the system?
05:55.03*** join/#asterisk HellAgony (n=HellAgon@200.121.241.223)
05:55.16supaigtrI have 6 machines all identical.  The most run 410p with no problems.  They are used for applications that are digital. I have two systems that need the ecan card.
05:55.37*** join/#asterisk Snake-Eyes (n=blog@ws01.5749.dsl.winshop.com.au)
05:56.11*** join/#asterisk HellAgony (n=HellAgon@200.121.241.223)
05:56.35*** join/#asterisk archimedes (i=archimed@adsl-70-247-240-218.dsl.ltrkar.swbell.net)
05:56.45Netgeeksand they all have the echo and cracking?
05:57.20archimedesanybody here care to help a novice with a RH9 install problem?
05:57.21supaigtrno just machines that connect to PSTN and term to analog.
05:58.21Netgeeksand the machines that connect to the PSTN are TE410Ps and they are connecting using PRI, right?
05:58.25*** join/#asterisk recursion (n=as@204.246.139.68)
05:59.38*** join/#asterisk recursion (n=as@204.246.139.68)
05:59.44supaigtrWe are tring to use 411p to connect PSTN via PRI.
06:00.16archimedeswell anoy way this is probably an easy one, I can get all but asterisk to compile because of the app_rxfax.c: In function `phase_e_handler':
06:00.16archimedesapp_rxfax.c:92: structure has no member named `cid'
06:00.16archimedesmake[1]: *** [app_rxfax.o] Error 1
06:00.16archimedesmake[1]: Leaving directory `/usr/src/asterisk/apps'
06:00.18archimedesmake: *** [subdirs] Error 1
06:00.21archimedesproblem
06:00.42*** join/#asterisk recursion (n=as@204.246.139.68)
06:00.43Netgeeks411s are the new cards with the echo cancellation daughter boards?
06:00.52Netgeeks~pastebin
06:00.52jbotmethinks pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
06:01.07archimedeswoops soryy.
06:01.21*** join/#asterisk recursion (n=as@204.246.139.68)
06:01.35Netgeeksit looks like your rxfax vversion is incompatable with the version of asterisk
06:01.45Netgeekscallerid structure was changed a while back
06:01.54drumkillaa long while back
06:01.54Netgeeksyou need to match versions
06:02.04drumkillabut is different between 1.0 and head
06:02.11*** join/#asterisk bonez41 (n=aint@c-67-166-77-14.hsd1.ut.comcast.net)
06:02.33Netgeekshey drumkilla, I got money in my paypal acocunt that wants to be in yours instead..... ;)
06:02.45Netgeeksgod I type like shit
06:03.05twistedi'll take it
06:03.20drumkillatwisted: ha
06:03.24drumkillawant to write some code?
06:03.28Netgeeksas long as you do the work!
06:03.32twisteddrumkilla, for what?
06:03.44drumkillaNetgeeks: seriously, if you don't find someone by next week, then I can do it
06:03.56SkramXNetGeeks what do you  need??
06:04.02drumkillabut I'm about to have to pack up everything I own and move it 1500 miles by Saturday
06:04.37drumkillait wouldn't take very long at all ...
06:04.41Netgeeksadd setvar= to zapata and add a directive somewhwere to set a prefix for uniqueid
06:05.19NetgeeksI tossed a $100 on the table the other day at drumkilla to get it done
06:05.30drumkilladangit, I'll do it right now
06:05.34drumkillaI need the money :/
06:05.37Netgeeks:)
06:06.04Qwelldrumkilla: where you moving?
06:06.17drumkillaQwell: back to South Carolina
06:06.21Qwelleww :p
06:06.25archimedesgotcha, I see it now... dang it...
06:06.33Qwelldrumkilla: Why going back so soon?
06:06.38drumkillaschool
06:06.40Qwellahh
06:06.54twistedoh
06:07.00twistedi could do that in like...  less than an hour
06:07.11drumkillatwisted: too late
06:07.13drumkillai'm doing it
06:07.21twistedhehe ok
06:07.23twistedenjoy
06:07.27*** join/#asterisk kks (n=kks@202.73.8.130)
06:07.36twistedyou need the money more than me anyway
06:08.01NetgeeksI'll have some money shortly for some other work if anyone is interested
06:08.19*** join/#asterisk clive- (n=pirch@rndf-146-28-144.telkomadsl.co.za)
06:08.19drumkillaI might be :)
06:08.21archimedesI may have some for some tutoring...
06:08.23archimedes:)
06:08.34twistedNetgeeks, keep me posted too
06:08.59Netgeeksthis one is more involved....  two projects.  One, fix the stuff martin wrote for me in mantis (twisted wanted to see the config file format changed)
06:09.00twistedalthough with me it all depends on time :
06:09.02twisted:/
06:09.27Netgeeksand 2. I need a way to inject an audio stream into a call as background and I don't want to use meetme
06:09.32twistedI what?
06:09.54twistedoh, #2 can be done using slinfactory ;)
06:09.56Netgeeksyour comments to the patch was (I don't like the way this is configured, can you make it less complex)
06:10.01twistedooooh ok
06:10.16twistedOH... THAT one
06:10.21NetgeeksI'll have a batch of cash to toss at both those later this week
06:10.50supaigtrnite ppl.  thanks
06:11.06twistedbtw, drumkilla, was your laptop released?
06:11.22drumkillayeah, supposed to be here tomorrow
06:11.28twistedword
06:11.32archimedesbut seriously, I nees a crash course in asterisk, and am wiling to pay for it. Idealy it would be via phone and some kind collaboration software.
06:11.32Qwelldrumkilla: LA customs?
06:11.40twistedassuming i'm all good, I'll have to come by and see it ;)
06:11.40Qwellrather...US customs in LA
06:11.51Netgeeksarchimedes where are you located?
06:12.01archimedesLittle Rock, Arkansas
06:12.06twistedQwell, it was stuck in china, IIRC
06:12.10Netgeekshrm
06:12.10Qwelloh
06:12.19Qwelldrumkilla: Well, don't feel bad.  I had a PC case stuck in customs for 6 weeks.
06:12.27drumkillaQwell: yeah, Chinese customs
06:12.27Netgeeksmight want to get on asterisk-biz and ask for the help, might find someone local
06:12.40drumkillait's in the US now
06:12.50twistedword
06:12.50Qwellsome jackass decided that since the box had "XP" on the front, that it was pirated software
06:12.54twistedwhere at in the US/
06:12.54NetgeeksChina is BAAAAD
06:13.03archimedesyeah thought about that but since you were tossing around cash, figured I might join the fray
06:13.06twistedWHOA... next round of painkillers is hitting HARD
06:13.17twisted*(guess that's becaues i w as alredy tired)*
06:13.25drumkillait left Memphis at 2:22 pm today
06:13.29twistedahh
06:14.06NetgeeksAren't there some asterisk starter courses at Astricon?  or is that too far away
06:14.12twistedyeh
06:14.15twisted"meat asterisk"
06:14.18twistederr wait
06:14.20twistedMEET asterisk
06:14.24drumkillameat, ha
06:14.29Netgeekslol
06:14.33QwellThats at the afterparty
06:15.29NetgeeksThats nearly as good as SarahEmm's comment... I was asking if anyone had any prompts they wanted recorded in a female britt voice (I'm having the entire asteirsk sound pack recorded by her) and SarahEmm was making a comment about CVS.. She said: I like Head.
06:15.37NetgeeksI said I'm not shure they would record that for us
06:15.44Netgeekssure
06:15.46twistedI like head too
06:15.51twisted(getting it)
06:16.01drumkilla~thwack twisted
06:16.01jbotACTION hits twisted on the ear with a UNIX Manual
06:16.05twistedoww
06:16.06NetgeeksBut I added it to the list, so we'll see
06:16.20SkramXnetgeeks: this is free and unlimited?
06:16.21twistedNetgeeks, you haven't heard my IVR have you
06:16.27Netgeeksnope
06:16.50twistedAT&T natural voices
06:17.02Netgeeksthis message mean anything to anyone
06:17.07NetgeeksAug 10 01:45:33 WARNING[49158]: chan_iax2.c:4896 socket_read: Error: Resource temporarily unavailable
06:17.13twistedoof.
06:17.17twistedNetgeeks, gentoo?
06:17.39Netgeeksnope, RedHat 9.0
06:17.41twistedoh
06:17.50NetgeeksCVS Head from 2004.09.09
06:17.53*** join/#asterisk gdsm (n=gdsm@e1-1.ns500-1.ts.milt.as9105.net)
06:17.55twistedi've seen it on gentoo when running in the hardened kernel
06:18.00Netgeekswas working for 6 months
06:18.00twistedOMFG
06:18.02twistedUPGRADE
06:18.05Netgeeksthen today it went poof
06:18.12twistedhehe
06:18.38twistedokay, i must go lay down, as my head is starting to feel up in the clouds
06:19.04*** join/#asterisk Beccara (n=tristram@222-152-13-41.jetstream.xtra.co.nz)
06:19.54PakiPenguinanyone's in uk right now? check a did for me please?
06:21.07SkramXPaki: what DID?
06:21.30archimedeshmmm the next astricon training is in December...
06:21.56outtoluncwhich town?
06:22.17archimedesorlando florida
06:22.30outtoluncdamn, gonna miss that one also <G>
06:22.48archimedeswhy is that?
06:23.17archimedesholy price tag batman..... 3G's
06:23.24outtoluncbecause i'm SJ area and can't make $$$ just to hang in FL
06:23.40PakiPenguin+442070994869
06:23.57archimedeswell if your a cute blond female I have a place there... :)
06:24.02outtoluncat least he didn't put up an 800 number <G>
06:24.09PakiPenguin:)
06:24.18outtolunci'm not cute
06:24.24outtolunci'm not blonde
06:24.29outtolunci'm not femake
06:24.33outtoluncer female
06:24.42archimedes:)
06:25.02X-RobI am!
06:25.04X-RobOh hang on.
06:25.06X-RobNo I'm not.
06:25.06*** join/#asterisk toddf (n=toddf@net-66-210-111-62.theshop.net)
06:25.10archimedesthat's ok for $3000 I can get allot tutoring from other people...
06:25.10X-RobD'oh.
06:25.10outtoluncso basically, i'm a dweeb in SJ with no way to the 'ball'
06:25.23outtolunchahha
06:25.30archimedeseverybody join me.... awwwwww
06:25.33Netgeekswhat's 3k?
06:25.38Netgeeksastricon?
06:25.39dudesNope I'm a guy ... wait let me check ... Yup I got a Penis
06:25.49vuviehahha
06:25.52archimedesastricon training in orlando florida... 1 week's worth
06:25.54QwellNetgeeks: no, astricon is like 550 for a full pass
06:25.54outtoluncfor $3k i'll show you how to make ... never mind <G>
06:25.58X-RobI asked my wife if I was a lesbian.
06:26.06X-RobShe didn't think I was.
06:26.48drumkillaAstricon is going to be awesome
06:26.50drumkillaI can't wait
06:26.52outtolunci'm letting qwell to my travel plans... and we all know me and his don't mix <G>
06:27.03outtoluncer his/him
06:27.05Qwellouttolunc: eh?
06:27.16QwellI'm doing your travel plans?
06:27.31archimedesyeah crap I'll be out there on vacation sept 4-24th would have to fly out there again 2 weeks later...
06:27.46outtoluncdid he read alot into that, yes i think he did <G>
06:27.47*** join/#asterisk Tili (i=Tili@219.137.40.89)
06:28.00outtoluncit was a statement
06:28.02Qwelldrumkilla: yeah it is...
06:28.12archimedescrap recompiled and it's still giving me that crap...
06:28.26drumkillaI hope they are taking up IRC names for nametags :)
06:28.28outtoluncone ment to well, history will tell
06:28.37Qwelldrumkilla: I was going to request it on mine, heh
06:28.48drumkillathat's how a lot of us know each other
06:28.49QwellMy badge is gonna have to be oversized...
06:29.05Qwellreal name, alias 1, alias 2, employer
06:29.38drumkillai'm going to sleep now
06:29.53drumkillaNetgeeks: i'm too tired to think about programming right now ... sorry
06:30.05Netgeeksno problem drum
06:30.07outtoluncover sized badges is a *big* thing, i'm wondering if i even need to continue
06:30.18Netgeeksstill gonna try and get it done, or should I find someone else?
06:30.19outtolunchehe
06:30.41Netgeeksbefore you move, I should say
06:30.53archimedeshmmm I can't do this anymore either.. time for beds... hmm MUST be getting old... used to be able to do this until 6 in the morning.
06:31.20outtoluncwell i was about to post
06:31.35archimedes???
06:31.59outtoluncstraight white boy w/gf looking for transcontinental passage
06:32.13archimedeshehe
06:32.29outtoluncbut i thought that *just might be a bit harsh
06:33.41archimedesnah.... oh well gnight I'll be back on it in the morning....
06:33.47outtoluncgiven that *some* thought it was ok to 'traverse' #asterisk-drinkers with bs the last few days... i thought i'd return the favor
06:34.44outtoluncseems kind of silly eh?
06:35.38outtoluncwell?
06:35.52*** join/#asterisk MGSsancho (N=sancho@68.120.83.236)
06:35.54Netgeeksarch: you were trying to install rxfax?  or you just want a general primer on asterisk?
06:37.03*** join/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net)
06:39.45*** join/#asterisk ptblank (n=MURDER1@ca-yorbalnd-cuda2-c1a-124.anhmca.adelphia.net)
06:40.49*** join/#asterisk Maksim (N=max@213.142.207.20)
06:43.10UberbotHi all.
06:43.49Maxx4lifehola
06:45.05MaksimHeh, JerJer's H323 uses atxfer well, OH323 have problems with atxfer... :)
06:46.23*** join/#asterisk erickj_az (n=erickj_a@ip68-228-209-118.ph.ph.cox.net)
06:49.04erickj_azhas anyone tried to set up two different ip networks and sip?
06:49.40erickj_azI seam to only be able to get one gateway to allow sip registrations
06:51.02*** join/#asterisk stkn_ (i=nobody@gentoo/developer/pdpc.active.stkn)
06:57.40*** join/#asterisk recursion (n=as@204.246.139.68)
06:57.54*** join/#asterisk heath__ (n=root@12-215-34-84.client.mchsi.com)
06:59.31*** join/#asterisk rec (n=rec@c-67-182-205-227.hsd1.ut.comcast.net)
07:00.13*** join/#asterisk el_basto (n=basto@adsl-69-232-57-184.dsl.irvnca.pacbell.net)
07:03.23*** join/#asterisk knielsen (n=knielsen@stone.sifira.dk)
07:04.04*** part/#asterisk rec (n=rec@c-67-182-205-227.hsd1.ut.comcast.net)
07:05.38*** join/#asterisk rec (n=rec@c-67-182-205-227.hsd1.ut.comcast.net)
07:05.48*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-145.igs.net)
07:06.19*** part/#asterisk el_basto (n=basto@adsl-69-232-57-184.dsl.irvnca.pacbell.net)
07:11.02*** join/#asterisk djin (n=djin@213-132-172-4.multikabel.nl)
07:16.33*** join/#asterisk PhreeStyle (n=PhreeSty@cpe-24-221-52-165.az.sprintbbd.net)
07:21.09*** join/#asterisk RoyK (n=roy@142.80-203-78.nextgentel.com)
07:23.51*** part/#asterisk bonez41 (n=aint@c-67-166-77-14.hsd1.ut.comcast.net)
07:24.46*** join/#asterisk rkvarala (n=rkvarala@59.93.52.232)
07:25.45rkvaralahello friends how to know that i have installed the zaptel correctly or not ?
07:26.34Qwellrkvarala: if it works, there is a good chance it's installed right
07:27.38Qwellbed
07:28.10*** join/#asterisk mago2-cn (i=maxgluck@200.109.167.9)
07:29.26mago2-cnhello everyone, i'm using lcdial and it selects the routs accordingly to rates, but it doesn't replace ${EXTEN} from the dialstr in the provider table... please take a look and comment, thanks:   http://pastebin.ca/19676
07:29.31rkvaralano it is not workig
07:29.50mago2-cnlcdial?
07:29.53rkvaralamy doubt is that it is not installed properly
07:30.24*** join/#asterisk A-Tuin (N=a-tuin@steves.ip.v4.me.uk)
07:30.38mago2-cnrkvarala: are you referring to lcdial?
07:31.01*** part/#asterisk PakiPenguin (i=uppal@202.147.163.14)
07:32.33*** part/#asterisk knielsen (n=knielsen@stone.sifira.dk)
07:32.40*** join/#asterisk razu (n=razu@61kontor.ewn.ee)
07:33.09mago2-cnany ideas please?
07:33.18rkvaralano
07:33.47*** join/#asterisk malandro (n=awefawe@201.240.216.189)
07:37.41*** join/#asterisk knielsen (n=knielsen@stone.sifira.dk)
07:39.42djinmago2-cn, what's your extensions.conf line?
07:39.59malandrohi does anybody knows where can i host cheap a voip pbx in asterisk?
07:40.34*** join/#asterisk EECore (n=EECore@CPE-144-136-192-218.sa.bigpond.net.au)
07:42.17EECorehey
07:45.21*** join/#asterisk skram| (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
07:48.39*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
07:51.42*** join/#asterisk tobiasWolf (n=konversa@195.162.255.10)
07:51.44*** join/#asterisk lele (n=lele@rivendell.windmill.it)
07:52.02*** part/#asterisk EECore (n=EECore@CPE-144-136-192-218.sa.bigpond.net.au)
07:52.07*** join/#asterisk lele (n=lele@rivendell.windmill.it)
07:52.18tobiasWolfhi all
07:52.24*** part/#asterisk lele (n=lele@rivendell.windmill.it)
07:53.44tobiasWolfanybody successful using app_sms from asterisk 1.0.9 in germany ??
07:53.54*** join/#asterisk luke-jr_ (n=luke-jr@user-0c938q3.cable.mindspring.com)
07:54.06djinmago2-cn, I think your problem is ${ EXTEN} instead op ${EXTEN}, btw
07:54.17djinop=of
07:57.22*** join/#asterisk FlungDung (n=wayne@rndf-146-63-145.telkomadsl.co.za)
08:03.35*** join/#asterisk leandro_it (n=leandro@ip-2-173.sn1.eutelia.it)
08:08.23*** join/#asterisk Tili (n=Tili@219.137.40.89)
08:10.10*** join/#asterisk rkvarala (n=rkvarala@59.93.52.232)
08:10.57rkvaralai have some problems with zaptel + sangoma
08:13.07rkvaralawhen i say pri show span 1
08:13.24rkvaralain status it shows as down and In Alarm
08:13.33rkvaraladoes any body have any idea on this
08:14.05NetgeeksI don't think too many are awake here
08:14.31Netgeeksare you using the latest beta drivers for sangoma and have you updated the card's firmware?
08:15.34rkvaralayes latest drivers only iam using
08:16.53leandro_ithave you connected the wire in the correct order?
08:17.20rkvarala<PROTECTED>
08:17.53leandro_ityes, maybe a PRI crossover cable instead of straight
08:18.09rkvaralaso what is the correct order may i know
08:19.38leandro_itI don't recall, look on voip-info
08:19.42rkvaralaok fine
08:20.02rkvaralathe problem is i have remote administration on that one
08:20.12Netgeeks1-4, 2-5, 4-1, 5-2
08:20.41*** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it)
08:21.45*** join/#asterisk LapTop006 (n=laptop00@sparc006.chriskaine.com.au)
08:23.24*** join/#asterisk luke-jr_ (n=luke-jr@user-0c938q3.cable.mindspring.com)
08:23.24*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-145.igs.net) [NETSPLIT VICTIM]
08:23.24*** join/#asterisk Maksim (N=max@213.142.207.20) [NETSPLIT VICTIM]
08:23.24*** join/#asterisk gdsm (n=gdsm@e1-1.ns500-1.ts.milt.as9105.net) [NETSPLIT VICTIM]
08:23.24*** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM]
08:23.24*** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) [NETSPLIT VICTIM]
08:23.24*** join/#asterisk ecto (n=ectospas@69.85.202.242) [NETSPLIT VICTIM]
08:23.24*** join/#asterisk nwhit (n=nwhit@ns1.whittrio.com) [NETSPLIT VICTIM]
08:23.24*** join/#asterisk _mwoodj_ (n=mwoodj@pdpc/sponsor/digium/hyper-eye) [NETSPLIT VICTIM]
08:23.24*** join/#asterisk Saaib (n=nabudoco@ns1.ensenada.gob.mx) [NETSPLIT VICTIM]
08:23.24*** join/#asterisk pnviking (N=pnlarsso@c83-248-2-153.bredband.comhem.se) [NETSPLIT VICTIM]
08:27.00*** join/#asterisk Maksim (n=max@213.142.207.20) [NETSPLIT VICTIM]
08:27.01*** join/#asterisk gdsm (n=gdsm@e1-1.ns500-1.ts.milt.as9105.net) [NETSPLIT VICTIM]
08:27.36*** part/#asterisk Maksim (n=max@213.142.207.20)
08:28.04*** join/#asterisk Maksim (n=max@213.142.207.20)
08:28.36*** join/#asterisk pnviking (n=pnlarsso@c83-248-2-153.bredband.comhem.se) [NETSPLIT VICTIM]
08:29.30*** join/#asterisk RoyK (n=roy@80.239.107.80)
08:29.36*** join/#asterisk Guggemand (i=irc@tester2.har-tabt.dk)
08:30.09*** join/#asterisk _mwoodj_ (n=mwoodj@24.96.145.218) [NETSPLIT VICTIM]
08:30.15*** join/#asterisk luke-jr_ (n=luke-jr@user-0c938q3.cable.mindspring.com) [NETSPLIT VICTIM]
08:30.25*** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com) [NETSPLIT VICTIM]
08:30.43*** join/#asterisk ecto (n=ectospas@69.85.202.242) [NETSPLIT VICTIM]
08:30.45*** join/#asterisk Saaib (n=nabudoco@200.76.231.14) [NETSPLIT VICTIM]
08:33.18*** join/#asterisk nwhit (n=nwhit@ns1.whittrio.com) [NETSPLIT VICTIM]
08:33.22*** join/#asterisk bjohnson_ (n=bjohnson@i216-58-62-145.igs.net) [NETSPLIT VICTIM]
08:34.40*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
08:36.05*** part/#asterisk xylome (N=asterisk@hg-msq-hol.levigo.de)
08:37.15*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
08:38.39*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
08:39.18*** join/#asterisk FlungDung (n=wayne@rndf-146-14-62.telkomadsl.co.za)
08:41.13*** join/#asterisk tuxinator_linuxM (n=tuxinato@ip68-109-146-168.ph.ph.cox.net)
08:41.24Tiliwhat is this for AST_FRAME_CNG
08:41.29djinIs it possible to write incoming sms (app_sms) to a file?
08:41.34TiliComfort noise. Howdoes it wrk
08:44.13*** join/#asterisk kaldemar (n=kalde@vipunen.hut.fi)
08:45.07*** join/#asterisk kore (i=kore@mindwipe.org) [NETSPLIT VICTIM]
08:45.18tobiasWolfdjin: if i understand it correctly then the command SMS(555,a) will write the incoming sms to /var/spool/asterisk/sms/sc-me.555. After that you can make with this file whatever you want,e.g. deliver it to an local phone. but for myself i havened even manged to let app_sms create this file. if i send an sms from my mobile to the * i see app_sms to some things but no file will be created
08:46.31djintobiasWolf, thanks. I'll give it a try.
08:47.44tobiasWolfif you have success i would be if you give me a hint of what exactly you have done
08:47.50tobiasWolf:)
08:47.59djinsure :)
08:48.08tobiasWolfthx
08:59.09djinmmm, strange. SMS was received, but no file indeed
09:01.35*** join/#asterisk shony (n=shony@nant.sumix.com)
09:02.25shonyHello
09:02.48shonyCan any body help me wtih asterisk + Solaris 10 + MGCP
09:04.43tobiasWolfdjin: my first thought was wrong permission on directorys but in my case the asterisk runs as user 'asterisk' and he owns all the files and dirs for *, like /var/spool/asterisk/*
09:04.57tobiasWolfdjin: btw what version of * do you run ?
09:05.52djinthis machine is running 1.0.5. I'll try 1.0.9 on a different machine to be sure.
09:10.47tobiasWolfi am running currently 1.0.9
09:12.10djinmmm, that provider doesn't seem to accept sms
09:12.42djinLuckily there are two more servers I can try :)
09:16.53*** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net)
09:16.56moverhellp
09:17.07movererr hello =)
09:17.15tobiasWolfdjin: can you tell me what do you mean with 'provider' ? do you mean the 1.0.9 box or do you have an extern provider for landline sms?
09:17.56djinI manage several boxes connected to several telecom providers.
09:19.40moveranybody has expeiences with HEAD and callgroup eg. pickupgroups?
09:19.54moveri need to increase this vom 63 to nnnnnnn
09:20.06movervom=from
09:20.36moveris this possible ?
09:20.51*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
09:21.17moveror can i manage the same function to pickup a call in another way?
09:21.42tobiasWolfin germany or in england? in wiki it says that app_sms is for BT, do you think this is important if i want to be able to send an sms from an landline phone to the *. in the first shot the sms should be stored there locally
09:21.44moveri have read many docs but no hint
09:22.00*** join/#asterisk secure75 (n=mic@ppp-62-245-210-171.mnet-online.de)
09:22.07djintobiasWolf, in The Netherlands.
09:22.53*** part/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
09:23.13*** join/#asterisk sambal (n=sambal@213.148.236.189)
09:23.21djinmm, found a working example. I'll try that.
09:23.35tobiasWolfgood luck :)
09:27.02*** join/#asterisk SkramX (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
09:28.21rkvaralawhat is pri cross over table
09:28.22rkvarala?
09:28.55rkvaralawhat is the order like 1-4,2-5,4-1,5-2
09:31.08djinrkvarala, http://www.google.com/search?hl=en&lr=&client=safari&rls=en&q=pri+crossover+cable&btnG=Search
09:32.13rkvaralaok
09:32.40*** join/#asterisk oob (n=oob@203-173-152-25.bliink.ihug.co.nz)
09:35.09djintobiasWolf, apparently KPN SMSC is failing :(
09:39.27*** join/#asterisk Netrunz (i=Netrunne@195.137.145.50)
09:39.34NetrunzHello everybody.
09:40.04Netrunzwhat does this error mean? WARNING[10947]: indications.c:77 playtones_alloc: Unable to set 'SIP/200-6d95' to signed linear format (write)
09:41.51tobiasWolfdijn: am i correct that you have tried to send an sms from a landline phone to an mobile ?
09:42.14*** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net)
09:42.35djinno, from a mobile to a landline (for now)
09:43.36tobiasWolfdijn: and the sms doesn't reaches you * because the smsc of your mobile provider is down ?
09:43.45tobiasWolfyou -> your
09:44.54NetrunzAnybody wanting to help?
09:44.54Netrunzwhat does this error mean? WARNING[10947]: indications.c:77 playtones_alloc: Unable to set 'SIP/200-6d95' to signed linear format (write)
09:48.38lterswell, it is a warning...
09:49.16ltersis your sip working?
09:49.22djintobiasWolf, outbound was accepted by the SMSC, but no sms received on my mobile (yet)
09:49.40djinand the inbound sms is nog coming in (after >15 min)
09:50.42djinmm, it make a directory in /var/spool/asterisk/sms and leaves it empty.
09:50.48*** join/#asterisk skram| (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
09:51.29*** join/#asterisk joelsolanki (i=Joel@202.160.161.93)
09:51.34*** join/#asterisk IronHelix (n=irc@ool-45785cfe.dyn.optonline.net)
09:52.05*** join/#asterisk bsdfreak (I=ninja@enterthebass.com)
09:53.22joelsolankiHello group
09:53.35joelsolankiI need some help for setting up asterisk and ser voip
09:57.26shonyDoes anybody can help me with installation of asterisk on solaris?
09:57.26*** join/#asterisk ixx_ (i=foobar@cpe-24-27-44-163.austin.res.rr.com)
09:58.37Netrunzlters yes the sip is working but it still gives me that warning
09:59.37bsdfreakdiddly bop
10:00.54Netrunzbesides I keep not on understanding why whenever I hangup any call (inbound or outbound) I get a spawn extension exited non-zero error
10:01.18FlungDungHi all, I have a wierd problem... When I boot asterisks it starts fine but if I stop if manually it dies with no real error -that I can see-. Any ideas? I must be missing something.
10:02.47*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net)
10:09.38FlungDungCould this be considered a fatal error? http://pastebin.ca/19680
10:10.17Netrunzwell it isn't loading the application for receiving faxes
10:10.18*** join/#asterisk iguy-duex (n=iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
10:10.35NetrunzYou don't have all files installed
10:10.46*** join/#asterisk SkramX (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
10:10.57Netrunzdid u copy the APP_TX and APP_RX to the app dir before recompiling asterisk?
10:11.28*** join/#asterisk Romik (n=romik_@1.fix.netvision.net.il)
10:11.29Netrunzand patched the makefile by the way
10:12.33FlungDungI just used asterisk at home. I can run asterisk using amportal, but asterisk -vvvvvvvcg dies.
10:13.07joelsolankihello
10:13.08NetrunzOh, I see. did u install from the full cd or id u install it over a fresh linux distro
10:13.12Netrunzhello
10:13.53*** join/#asterisk grimse (n=grimse@p5481E699.dip.t-dialin.net)
10:14.38Chotaireoh my
10:14.42tobiasWolfdjin: when i send an sms from an mobile to the * box i get the following on console output
10:14.59tobiasWolfdjin: -- SMS TX 93 00 6D 00 00 00...                                                                    
10:15.08joelsolankiHello room
10:15.15tobiasWolfdjin: -- CAPI Answering for MSN 1087997                                                                
10:15.28tobiasWolfdjin: -- SMS TX 92 01 FF 6E 00 00...                                                                    
10:15.40FlungDungfull cd
10:15.41tobiasWolfdjin: after that a lot of the last line i pasted
10:16.08tobiasWolfdjin: any clue ?
10:16.23*** join/#asterisk jeanmi_ (n=besnard@front.tekkno.net)
10:16.24Netrunzflung pvt me
10:16.35FlungDungNetrunz, fresh install this morning by accident, restored from a backup and everythings bee pearshaped since
10:17.22*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
10:17.24Chotairethen your backup is incomplete ;)
10:17.38FlungDungall the files seem to be there
10:17.53jeanmi_what hardware do I need if I want to use asterisk as an IVR (aimed at running scripts on the asterisk host) ?
10:18.12Chotaireif you had also backed up your kernel and all kernel modules such as your dev files/links, it should run.
10:18.28Chotaireyou better check your backup script for missing dirs.
10:18.33jeanmi_for instance I'd like to call my phone number, talk to the IVR and press one to run a given script...
10:19.22FlungDungthats the thing, it does run, pretty well if I run using amportal
10:19.24Chotaireflungdung: if you hit "lsmod", do you have all the asterisk modules up?
10:19.32Chotairehmm...
10:20.01Chotaireso when you start asterisk -vvvvvvvvvvvvvvvc where does it die?
10:20.32FlungDunghttp://pastebin.ca/19680
10:20.58Chotairedo you use any fax options with asterisk then?
10:21.16FlungDungnot that I know of
10:21.38NetrunzWell asterisk@home installs spandsp by default
10:22.13Chotairewe want him to try noload?
10:22.19Chotaireif he doesn't use faxes anyway?
10:22.21NetrunzYup
10:22.30Chotaireok go to /etc/asterisk/modules.conf
10:22.38Chotaireinsert the following line:
10:22.42Netrunzbut still, that shouldn't stop asterisk from loading anyway
10:23.06Chotairenoload => app_rxfax.so
10:23.15Chotairethen restart asterisk and see what happens.
10:23.24Chotairethere must be more errors.
10:24.31Chotairenetrunz: same opinion here but let's see.
10:24.35NetrunzChot by the way maybe you could help me.
10:24.59Netrunzbesides I keep not on understanding why whenever I hangup any call (inbound or outbound) I get a spawn extension exited non-zero error
10:25.27jeanmi_what hardware do I need to be able to call my asterisk with an analog line ?
10:25.49jeanmi_I have a quicknet card at home but I don't know if that's enough ...
10:25.53Chotairenetrunz: you should be talking to a dev then.. I get the same error and I can live with it, as it probably only means that this error code is being forwarded to any script/app that used this feature.
10:25.58NetrunzNo hardware... just a voip account
10:26.07Netrunzat least :)
10:26.16Chotairejeanmi: so you want analog PSTN termination?
10:26.27FlungDunglol, no change?
10:26.42Netrunzthought so
10:27.05Chotairejeanmi: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=S101I
10:27.12FlungDungno, even the errors still in the log.
10:27.18Chotairejeanmi: browse that one and see if it is cool for you.
10:27.21FlungDungthat should be gone...
10:27.30jeanmi_Chotaire I would like to be able to do the following: from a public telephone, I call a number to reach my IVR which would run scripts
10:27.36Netrunzflung why don't you use a normal box and not AAH
10:27.38Netrunz?
10:27.52Chotairejeanmi: do you have an analog phone line or ISDN?
10:27.55NetrunzAAH is really a pain in the butt when it decides not to work
10:27.57jeanmi_analog
10:28.06Chotairehttp://www.digium.com/index.php?menu=product_detail&category=hardware&product=S101I
10:28.08Chotairethen check this one.
10:28.20Netrunzchotaire he can do that with a voip account
10:28.27Netrunzhe doesn't need hardware
10:28.36Chotaireoops
10:28.38jeanmi_Netrunz so where would I create that voip account ?
10:28.48Chotairenetrunz: yes but the quality is not the same.
10:28.56Netrunzdepends
10:29.05Netrunzjeanmi where u from?
10:29.08Chotairejeanmi: if you wanna do that, all you need is e.g. a SIP provider that offers you a real phone number with your SIP account.
10:29.10jeanmi_france
10:29.11FlungDungNetrunz, Spose the only reason is that the installs quicker
10:29.30NetrunzFlung if u want to i can help u with a clean install
10:29.38Netrunzjean
10:29.43Chotaireflungdung: I think you should take netrunz' offer and do that.
10:29.54NetrunzI will look for a provider in your country hold on
10:30.03jeanmi_so asterisk needs to know about that SIP account... I call my SIP number and I'll get to talk with my IVR ?
10:30.12Chotairejeanmi: if you want to use your own analog phone line, all you need is a compatible voice modem.
10:30.17*** part/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw)
10:30.23Chotairejeanmi: yes, asterisk acts as your SIP phone in that case.
10:30.47NetrunzYupù
10:30.49NetrunzYup
10:30.55jeanmi_how do I find out whether my modem is compatible ?
10:31.04Netrunzthere is a very good explanation on voip-info.org
10:31.12jeanmi_Netrunz what do u mean ? where are u located ?
10:31.15Netrunz"Asterisk as a sip client"
10:31.23Netrunzjean I am in Italy
10:31.26jeanmi_ok
10:31.28Chotairejeanmi: if you want to use your analog phone line for this (so you dont have a new phone number), try one of these hardware devices:
10:31.29ChotaireDiscontinued Zaptel Hardware
10:31.29ChotaireWildcard X100P A single port FXO PCI interface card for interfacing with a standard analog phone line.
10:31.31Netrunzbut I havea list of voip providers
10:31.35ChotaireGeneric
10:31.35ChotaireGeneric X100P Intel IA92 WinModem compatible with X100P. This is the so-called "clone" X100P. No technical support is available for this product.
10:31.36jeanmi_ok
10:31.51Netrunzchotaire that thing has quality problems
10:31.55jeanmi_does the auicknet cqrd help in so;e zqy M
10:31.58Netrunzi got two of them
10:32.01jeanmi_oops
10:32.14jeanmi_does the quicknet card help in some way ?
10:32.36Chotairejeanmi: http://www.asterisk.org/index.php?menu=hardware
10:32.39Chotairethat's all the supported hardware.
10:33.12*** join/#asterisk xeax (n=x@adsl-215-41-46.mia.bellsouth.net)
10:36.25*** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it)
10:37.35Mimmushi, all on the beach?
10:40.57Netrunzwell I got a new warning I can't understand...NOTICE[11217]: channel.c:1691 ast_set_write_format: Unable to find a path from slin to g729
10:41.14NetrunzI only have g729 allowed everywhere!
10:41.24Netrunzwhy does it go in slin?
10:48.17*** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca)
10:53.12*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
10:59.00joelsolankiheo
10:59.05joelsolankihelo
10:59.33*** join/#asterisk mutil (i=WebChat@i.think.napoleon.dynamiteblows.com)
11:01.46*** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
11:04.42*** join/#asterisk seong (n=seong@218.111.210.171)
11:14.11*** join/#asterisk tld (n=tld@cD9088681.inet.catch.no)
11:14.20*** join/#asterisk Praktikant01 (n=root@dsl-084-059-128-166.arcor-ip.net)
11:14.52Praktikant01I need help, my asterisk cuts of the 0 from incomming calls, whats going wrong ??
11:15.10*** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl)
11:15.38puzzledmorning
11:16.59*** join/#asterisk ham4fun (n=ham4fun@pool-138-88-28-85.res.east.verizon.net)
11:17.30*** join/#asterisk gordonjcp (N=gordonjc@cpc3-broo2-5-0-cust232.renf.cable.ntl.com)
11:17.50jalsothi
11:18.13djintobiasWolf, that's the same I get. Strangely enough there is no output of the 'hangup', following the 'sms'-command.
11:18.21*** join/#asterisk liversmudge (n=liversmu@217-14-176-201.as25582.net)
11:18.25jalsothow can I used fetures from features.conf? I have allowed them, getting DTMF tones from phones, but nothing happens :(
11:18.32liversmudgeafternoon
11:18.37*** join/#asterisk SkramX (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
11:19.26liversmudgeanyone know anything about "flash operators panel" .. specifically trunks?
11:20.01liversmudgeI have multiple buttons but using a , seperated list of positions
11:20.22liversmudgeand the buttins resopnd when I dial out of the runk , but nothing shows on the inbound??
11:20.25liversmudgeany ideas?
11:22.05RoyKhmmmm
11:22.16RoyKanyone that knows how i can convert mp3 to alaw wav?
11:23.03*** join/#asterisk GMsoft (N=gmsoft@gentoo/developer/gmsoft)
11:23.06GMsofthi
11:23.35GMsoftare there a commands to enable/disabled MWI in extension.conf ?
11:23.41*** join/#asterisk quabil (i=gst@eris.sysfrog.org)
11:24.09RoyKMWI?
11:24.18GMsoftMessage Waiting Indicator
11:24.41*** join/#asterisk Akelavlk (n=jansun@82.119.239.141)
11:25.01RoyKah
11:25.11GMsoftI'd like to turn it on/of myself and not rely on the buildin voicemail to do that
11:25.19RoyKGMsoft: dunno. google or look it up on the wiki
11:25.32RoyKGMsoft: i beleive people are doing it......
11:25.42GMsoftRoyK: if I'm here that means that I've already been looking :)
11:26.44AkelavlkHello, does anybody know if it's possible has dynamic MeetMe rooms? I mean without configure meetme.conf. That caller just click to # and he will create new room.
11:26.53dwildesGMsoft - in sip.conf, just don't specify the 'mailbox=<exten>' option
11:27.17RoyKGMsoft: sorry. didn't mean to be rude
11:27.27GMsoftdwildes: that's for the built-in voicemail. I want to turn it on and off myself ! :)
11:27.28RoyKGMsoft: or at least, not very rude :P
11:27.33GMsoftRoyK: np :)
11:27.51GMsoftdwildes: I'm recoding the voicemail in a AGI in fact
11:28.29dwildesgmsoft:  I guess the reason is - why would you need to turn it on/off yourself?
11:28.49*** join/#asterisk riksta (n=rick@84.93.243.170.broadband.plus.dyn.plus.net)
11:28.49GMsoftdwildes: because I recoded the voicemail myself and mwi is the only stuff I can't get working
11:28.55dwildesyou could use the sip_notify.conf file and issue 'sip notify' cli command
11:29.24dwildeswhat type of phone are you using?
11:29.27GMsoftmhh let me look that up
11:30.08*** join/#asterisk mutil (i=WebChat@i.think.napoleon.dynamiteblows.com)
11:30.27dwildesyou can register multiple mailboxes in sip.conf for the extension as well:   mailbox=1000,1001,1002  and if any of 1000,1001,1002 receives a voicemail - most sip phones will light the MWI
11:30.39dwildesif that helps anyway  ;)
11:31.09GMsoftmhh no because I don't use the built-in voicemail so the mailbox parameter is useless
11:31.31dwildesoh - I gotcha now
11:31.39GMsoftand sip_notify.conf seems to be in the cvs version and I'm using the stable one atm
11:31.42quabilhmm.. how can i read channel variables in initreqprep? i tried pbx_builtin_getvar_helper(p->owner, "SIPDOMAIN") but this always returns NULL...
11:32.27dwildesgmsoft - if you are using an AGI, you could use the 'sipsak' program to send a 'NOTIFY' message manually to the phone
11:32.49quabil(in chan_sip.c)
11:34.19GMsoftdwildes: mhh ok but then I need to parse myself the sip.conf and gather ip and so on. I would have liked a asterisk command like turnMWI("on", "SIP/foo") :)
11:37.17dwildesgmsoft:  Yeah, that would have been nice :)   but, that's the purpose of the newer 'sip notify' command so you could pretty much send any type of notify to any sip peer
11:38.12*** join/#asterisk syle (n=blah@wnpgmb06dc1-44-164.dynamic.mts.net)
11:45.14*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
11:50.08*** part/#asterisk knielsen (n=knielsen@stone.sifira.dk)
11:50.21*** part/#asterisk vuvie (n=vuvie@bb219-74-44-131.singnet.com.sg)
11:52.40*** join/#asterisk loick (n=loick@per92-7-82-236-197-96.fbx.proxad.net)
11:55.00*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
11:58.15*** join/#asterisk DannyF (n=dannyf@h194n1fls32o865.telia.com)
12:02.38*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
12:03.15*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
12:09.09*** join/#asterisk Jas_Williams (n=Jason@host86-130-0-82.range86-130.btcentralplus.com)
12:12.28*** join/#asterisk lathos42 (n=lathos42@65-42-27-66.dowdingindustries.com)
12:17.16*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
12:17.21shadebobooany zaptel developper here? I have a question on interrupt initialisation...
12:22.02jalsothow can I used fetures from features.conf? I have allowed them, getting DTMF tones from phones, but nothing happens :( It seems, the called party is getting the DTMF digits and asterisk doesn't process them. Any suggestion?
12:22.38sylereading some security stuff
12:22.59syleif your kernel is less than 2.6.11 i;d silently upgrade :)
12:23.41*** join/#asterisk zotz (n=zotz@24.231.36.100)
12:24.09*** part/#asterisk Akelavlk (n=jansun@82.119.239.141)
12:25.15*** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com)
12:28.33*** part/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it)
12:29.26*** join/#asterisk nextime (n=nextime@213-140-6-96.ip.fastwebnet.it)
12:31.56*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
12:32.10brimstonesyle: what are you reading that makes you think this? link?
12:32.45djinanyone experience with receiving sms on *?
12:32.58djinKeep getting:
12:32.58djin<PROTECTED>
12:32.58djin<PROTECTED>
12:33.16djinlast two line repeated about 15 times
12:33.44djinon Asterisk 1.0.7, btw
12:33.45*** join/#asterisk fenlander (n=neils@82.152.81.57)
12:33.54shadeboboosomeone have implemented a polarity reversal in wcfxs.c?
12:34.20gambolputtyanyone work with devstate?
12:37.09*** join/#asterisk DruidFyr (n=druidfyr@rrcs-24-73-79-30.se.biz.rr.com)
12:37.30*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
12:37.56dwildesgambolputty - spent most of yesterday playing with it , submitted a 'Devstate' patch to implement in HEAD yesterday based of the 'bristuff' patch
12:38.31gambolputtyI have a snom 190 ip phone
12:38.35dwildesI needed something to toggle a led button on my snom phone for meetme conferences, DB entries, etc..
12:38.36gambolputtyI have head from yesterday
12:38.38*** join/#asterisk shony_ (n=shony@nant.sumix.com)
12:38.51gambolputtyI have put the patches from bug 3644 in
12:38.55gambolputtyand app devstate
12:39.24gambolputtyhttp://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20BristuffDevstate
12:39.29gambolputtyI put in statements like that
12:39.30dwildesgambol:  sounds like you are ready to try it out
12:39.41gambolputtyI type in at the the cli devstate 335 2
12:39.46gambolputtyand the led doesn't light up
12:40.08gambolputtywhat am I doing wrong?
12:40.15dwildesyou have to setup a 'hint' for that extension as well, and make sure your snom button is set as a 'dest' for 335
12:40.34gambolputtythat is done
12:40.42dwildesare you hinting on a SIP extension - or a DB variable?
12:40.51gambolputty<PROTECTED>
12:40.59gambolputtywhat is it supposed to be?
12:41.15dwildesthat looks right
12:41.32gambolputtythen why doesn't the led light up then?
12:41.32dwildestry 'devstate 335 3'
12:41.36gambolputtyhold on
12:41.45dwildesdo a 'show hints' and see if the state number changes
12:41.48gambolputtydoesn't work
12:41.49gambolputtyhold on
12:42.01gambolputtyThere are no registered dialplan hints
12:42.09gambolputtyhow do I change that?
12:42.17dwildespaste your dialplan entry for 335
12:42.23rkvaralai am facing a problem with music on hold where it says its playing but no   instance of mpg123 is starting
12:42.32*** join/#asterisk shony_ (n=shony@nant.sumix.com)
12:42.32gambolputty<PROTECTED>
12:42.32gambolputtyEdit Delete 1045 from-sip 335 hint DS/335  
12:42.32gambolputtyEdit Delete 1054 from-sip 335 1 DevState 335|6
12:42.41rkvaralai have ztdummy in lsmod
12:43.13gambolputtyI even put app devstate in modules.conf
12:43.22dwildesgambol - sorry, I meant from your extensions.conf
12:43.30gambolputtythat is from there
12:43.32tobiasWolfdijn: back again :) before lunch i have started reviewing the app_sms.c. from what i can tell from now is the the funtion which writes the file isn't called. can you tell me how do you executes the sms app? i have made such an extension extension => 555,1,SMS(123,as). my hope was that i would be able to send an sms to 555 and that * stores the sms in queue 123.
12:43.33gambolputtyI use realtime
12:43.48dwildesah - hadn't tried the realtime stuff
12:44.00gambolputtyit should still work despite that
12:44.01dwildesbut it should be essentually the same principle
12:44.22dwildesdo a 'show channeltypes' and see if devstate shows
12:44.41rikstawho is "oej" on mantis please?
12:44.48gambolputtyDS          Devstate                       yes          no           no
12:44.56gambolputtythats what comes back
12:45.19tobiasWolfdjin: have you found the description under http://www.automated.it/asterisk/sms.html ?
12:45.27dwildesthat's correct - so the channel tech/type is registered.
12:45.37dwildesand a 'show hints' doesn't give you anything?
12:45.45gambolputtycorrect
12:45.49gambolputtyis it supposed to?
12:46.16dwildesyes - that's where the state information for you LEDs will come from
12:46.20h3x.0
12:46.30gambolputtythen how do I get a hint to show up there then?
12:46.50dwildesyou should have at least one entry, such as:
12:47.05dwildes<PROTECTED>
12:47.05dwildes<PROTECTED>
12:47.23gambolputty<PROTECTED>
12:47.26dwildestry putting in a statement not using realtime like:
12:47.28gambolputtystate 0?
12:47.32dwildesexten => 335, hint, DS/335
12:47.41gambolputtyI have to use realtime
12:47.45dwildessee if that makes it register with the hints
12:47.48gambolputtyelse I don't have database integration
12:48.06dwildeswell - this will at least know where the 'hints' problem is coming from
12:48.15gambolputtylet me change files
12:48.16gambolputtyhold on
12:48.30djintobiasWolf, yes. I found that info.
12:48.36djinI now try:
12:48.36djin<PROTECTED>
12:48.37djin<PROTECTED>
12:48.37djin<PROTECTED>
12:48.37djin<PROTECTED>
12:48.37djin<PROTECTED>
12:48.42djinstill no luck :(
12:49.00dwildesgambol:  if it works for static entries, but not for realtime - then we have a bug(or a different way to register a hint)   I personally haven't used realtime, so I'm unsure
12:49.31gambolputtyhint registers
12:49.39gambolputtywhen that entry is directly in extensions.conf
12:50.03gambolputtydevstate 335 6
12:50.10gambolputtythat doesn't make the led light up
12:50.20gambolputtynor does anything else
12:50.52dwildesgambol:  just found this in the maillists about hints & realtime:
12:50.54dwildeshttp://lists.digium.com/pipermail/asterisk-users/2005-April/102951.html
12:51.04dwildesthey said it didn't work for them either using realtime
12:51.14gambolputtygreat
12:51.16dwildesand HEAD
12:51.28gambolputtywell, we have a long way to go
12:51.47dwildeshow many hints are we talking about?
12:51.48gambolputtyrealtime is where its at
12:52.00gambolputtyI am just trying to monitor one extension at first
12:52.06gambolputtyas a test the waters kind of thing
12:52.10gambolputtythats it
12:52.28dwildesgotcha
12:52.46Kattymorning
12:53.11dwildesmy biggest problem with the 'hints' thing is the fact you lose all your subscribe info when you reload or restart.  I wonder if that is related, since realtime is dynamic
12:53.20*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
12:53.41gambolputtyI have the snom phone (333) monitoring 335
12:53.58gambolputtyI just plain expect all calls from 335 that go to 333 to show up on that function key
12:54.08dwildesreboot your snom, it probably lost its subscription when you reloaded asterisk
12:54.08gambolputtythat is doing the "monitoring"
12:54.16gambolputtyeven that doesn't happen
12:54.19dwildesdo a 'sip show subscriptions' to make sure it's there
12:54.45dwildesalso - now make sure your devstate is changing for 335
12:54.52dwildes> devstate 335 3
12:54.55dwildes> show hints
12:55.02Jas_Williamsgambolputty: this is completely off the wall does it have any affect adding rtcachefriends=yes  to sip.conf as this makes MWI work
12:55.03dwildessee if the state is now changed
12:55.23Jas_Williamsfor realtime
12:55.32shadebobooanyone known structure of "alpha  indirect_regs[]" in wcfxs.c?
12:55.40gambolputty192.168.3.8      333         3c26700c5cc6-1l93kt0o  335                   Down
12:55.47gambolputtymwi works for me
12:56.07gambolputtyand so does custom ring tones
12:56.12gambolputty3.60i firmware
12:56.16*** join/#asterisk Meaty (n=cp_simbu@office.abi.ca)
12:56.37gambolputty;these statements were added to get MWI working on my Snom 190 phone
12:56.37gambolputtycallerid=asterisk
12:56.37gambolputtyrtcachefriends=yes
12:56.37gambolputty;notifymimetype=text/plain
12:56.37gambolputtynotifymimetype=application/simple-message-summary
12:57.13*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:57.18gambolputtythis function key thing is the last hurdle
12:57.46*** join/#asterisk bkw_ (n=brian@asterisk/friend-and-developer/bkw)
12:57.46*** mode/#asterisk [+o bkw_] by ChanServ
12:58.29Jas_Williamsgambolputty: is this the phome your getting hints for as well ?
12:58.47gambolputtymy snom phone is 333
12:58.54gambolputty335 is a polycom ip 500 POS
12:59.20gambolputtyone of the 333 keys is set to destination for 335
12:59.22Jas_Williamsit might be worth moving the rtcachefriends=yes to the global section
12:59.34gambolputtywhy?
13:00.03Jas_Williamsso it applies to all phones, other wise you are only caching the infor for the snom phone
13:00.04gambolputtyits already in the general section
13:00.11Jas_Williamsk :)
13:00.21gambolputtyisnt that the global section?
13:00.44Jas_WilliamsThen no idea why hints don't work then have you checked mantis ?
13:00.46*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
13:01.32gambolputtybug 3644     www.junghanns.net
13:01.35gambolputtythats what I find
13:01.39gambolputtythat is related
13:02.08gambolputtyit seems like that there are differing code trees about snom lights
13:02.33gambolputtyand those are the two code trees
13:02.33gambolputty!
13:02.35*** join/#asterisk coppice (n=chatzill@253.198.17.210.dyn.pacific.net.hk)
13:02.46RoyKevening, coppice
13:03.03coppiceevening
13:03.14fearnoror morning ;)
13:03.20RoyKor night
13:03.23RoyKor whatever
13:03.28RoyKgood localtime()
13:03.34gambolputtyI ultimately would like to use the extra buttons on a snom phone that can attach to the right hand side
13:03.41coppicedepends how you classify 9PM
13:03.51RoyKmorning, definetely
13:04.14Jas_Williams~seen oej
13:04.17jbotoej <~oej@apollo.webway.se> was last seen on IRC in channel #asterisk, 7d 19h 35m 13s ago, saying: 'cvs is still dead'.
13:04.52*** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca)
13:04.54gambolputtyI thought this app devstate would just plain make a light turn on and off
13:05.05gambolputtyit doesn't do this
13:05.21RoyKapp_cansandstring
13:05.37RoyKs/app/chan/
13:05.48dwildesgambol:  it does - I'm just waiting on your output from the 'show hints'
13:05.57jontowchan_packetradio :P
13:06.10gambolputty*CLI> show hints
13:06.10gambolputty<PROTECTED>
13:06.10gambolputty<PROTECTED>
13:06.10gambolputty----------------
13:06.10gambolputty- 1 hints registred
13:06.23jontowthat'd be funny stuff.. VoIPoPR :P
13:06.23gambolputtywith that 335 stuff directly in extensions.conf
13:06.24dwildesgambol:  perfect
13:06.29gambolputtythats what happens
13:06.35gambolputtybut the light still doesnt go on
13:06.41coppiceI think someone did a chan_packetradio
13:06.46jontowreally.. :o
13:06.46dwildesbut your 'sip show subscriptions' shows you are subscribed to  '333' not '335'
13:07.00coppicechan_andshtring
13:07.06gambolputty*CLI> sip show subscriptions
13:07.06gambolputtyPeer             User        Call ID                Extension       Last state
13:07.06gambolputty0 active SIP subscriptions(s)
13:07.32tzangercoppice: we were talking about slinear<-->ulaw/alaw last night...  is it accurate to say that conversions between slinear and ulaw/alaw are more of a companding/expanding function than any real kind of data mangling?
13:07.41dwildesmust have done another reload on asterisk.   this is the annoying aspect of 'hints' I was mentioning earlier  ;)
13:07.55dwildesreboot that phone again, to make sure it's re-subscribed
13:08.21gambolputtyshouldnt the phone just plain stay subscribed?
13:08.41coppicetzanger: I'd call it a lossy compresssion, but its a free country (or are you in the US?) so call it what you  like
13:09.09gambolputtyPeer             User        Call ID                Extension       Last state
13:09.09gambolputty192.168.3.8      333         3c26700c46cd-8qlxrysl  335                   Down
13:09.10gambolputtyok
13:09.12dwildesit does as long as asterisk isn't reloaded/restarted.  the phone will then eventually re-subscribe
13:09.24gambolputty"eventually"
13:09.25dwildesbut, to make the process quicker, we reboot  :)
13:09.34gambolputtyhow often does it resubscribe?
13:09.39gambolputtyonce in a blue moon?
13:10.07dwildesI'm unsure - I've tested telling my phones to re-register every min, but sometimes the subscription doesn't come back for a few minutes
13:10.07tzangercoppice: is it lossy though?  I thought you could convert between slinear and ulaw or alaw all day long and get the exact same bit patterns after every conversion
13:10.21tzangercoppice: :-)  no I'm in Canada but it's not far off
13:10.43gambolputtydevstate 335 (whatever) now doesn't make the lights do anyting
13:10.48dwildesI personally haven't debugged where that problem lies.   It's definitely an annoyance
13:11.03coppiceyep, but you can't get back to the original. what you describe is also try of G.726, and that is far more lossy
13:11.12gambolputtyI expect the lights to instantly turn on
13:11.15coppices/try/true
13:11.31mutmornin everyone
13:11.40*** join/#asterisk Hmmhesays (i=negative@66.173.103.108)
13:11.41gambolputtythe subscription thing went away
13:11.52tzangercoppice: ahhh okay so slinear -> ulaw is lossy, but ulaw <--> slinear incurrs no further loss
13:12.06dwildesgambol:  like I said - your devstate is on DS/335, but you are monitoring 333
13:12.16dwildeseither choose to monitor 335 or change your devstate to 333
13:12.57coppicelinear->ulaw->linear->ulaw->linear->ulaw->linear->ulaw->linear->ulaw->linear should give the same as
13:12.59coppicelinear->ulaw->linear
13:14.22gambolputtyexten => 335,hint,DS/335
13:14.22gambolputtyexten => 335,1,dial,sip/335
13:14.22gambolputtyexten => h,1,DevState(333,0)
13:14.26gambolputtythat look right?
13:14.33tzangercoppice: that's what I said (wihtout all that copy/paste)  :-)
13:14.37gambolputtyfor extensions.conf?
13:16.30*** join/#asterisk secure75 (n=mic@ppp-82-135-14-79.mnet-online.de)
13:19.01*** join/#asterisk pdugas (n=pdugas@h117.72.40.69.ip.alltel.net)
13:20.08pdugasCan anybody tell me the names of the files Polycom files try to upload to the boot server at various times?
13:20.13gambolputtydevstate now changes the led lights
13:20.14dwildesgambol:  if you are just going to monitor the extension - you won't need 'devstate' at all.  Just do:  exten => 335,hint,SIP/335
13:20.15*** join/#asterisk skram| (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
13:20.28gambolputtyas long as the phone is subscribing
13:20.40dwildesdevstate is only used for applications that aren't 'state' friendly, like meetme, parking, astdb, etc...
13:20.55gambolputtyjust as long as I can control the state of the lights
13:21.00gambolputtythats all I care about
13:21.08*** join/#asterisk Weezey (i=Weezey@vaio2.tsnetworks.ca)
13:21.08dwildesyes, if your phone is subscribed - and you have your hints in there, all will be well
13:21.23gambolputtywhat prevents the phone from subscribing all the time everytime
13:21.26gambolputty?
13:21.28dwildesjust be sure to check it all by doing 'show hints' and 'show sip subscriptions'
13:21.38gambolputtythanks for that info
13:21.42*** join/#asterisk Craziman2 (n=Craziman@boromir.apid.com)
13:21.45gambolputtybut the subscribing thing goes away
13:22.11gambolputtywhen it does, I have no led light control via devstate anymore
13:22.16dwildeswhenever asterisk is reloaded, all the hint states are purged from memory.  As long as asterisk stays up/not reloaded - the subscriptions/hints won't go away
13:22.31gambolputtyhow can that thing be fixed?
13:22.38gambolputtyhints should never go away
13:22.44gambolputtysame with subscriptions
13:22.46dwildesif you do reload/restart, the phone will re-subscribe - but again, I'm not 100% sure of the frequency on the snom phones.  I thought it was during re-registration
13:23.04dwildesbut that hasn't always been the case for me
13:23.10gambolputtygreat
13:23.13RoyKnice. full opera browser now works on virtuall all gprs+ phones :)
13:23.18gambolputtyI reregister once a minute
13:23.26dwildesI agree - I don't think they should go away as well.
13:23.46dwildestry reloading asterisk, and see how long it takes the snom to re-subscribe to the subscription
13:24.14dwildesit will 'eventually' come back, for me - re-register every 60sec it would range from 1min->5min before it came back
13:24.36gambolputtynot good
13:24.46gambolputtyit should resubscribe every second
13:24.51*** join/#asterisk Jason357 (n=m00@cpe-65-26-177-31.indy.res.rr.com)
13:24.55Jason357hello
13:25.09gambolputtywhen I want the lights to on or off, I want them to go on or off
13:25.12gambolputtyplain and simple
13:25.16dwildestoo much traffic for a large network to re-sub every second
13:25.34gambolputtyanything to let an operator know if her boss is busy on the phone
13:25.41*** part/#asterisk HeadachesAbound (N=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net)
13:25.52dwildessince it's all in memory - I'm not sure what solution they are coming up with for the 'hints'
13:26.28dwildesif you are using realtime, maybe you could add another field like 'laststate' in the database
13:26.29*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
13:26.57*** join/#asterisk HeadachesAbound (n=Administ@adsl-70-244-228-14.dsl.tulsok.swbell.net)
13:27.05*** join/#asterisk DarthClue (N=DarthClu@adsl-70-244-228-14.dsl.tulsok.swbell.net)
13:27.10dwildesthen on a 'safe_asterisk' restart, it would run through the database for the laststate, and send the corresponding 'notifys'
13:28.09dwildesotherwise, unless it saved it someone on the HD or DB - I'm not sure how asterisk will 'remember' the states whenever it reloads/restarts
13:28.35dwildeswell, somewhere instead someone  :)
13:28.36*** join/#asterisk NirS (n=NirS@62.90.49.77)
13:28.47NirShey all
13:28.54NirSanyone has experience with app_changrab ?
13:29.15Jason357I have a broadvoice Sipura 2100 which has QoS... Is there any way to use it so that I can prioritize VoIP traffic over data when I switch to asterisk instead of just using the Sipura?  If not, how do I prioritize the packets?
13:31.31jontowQoS on the linux machine?
13:31.36*** join/#asterisk likwid-- (n=likwid@nc-67-76-40-104.dyn.sprint-hsd.net)
13:32.41SpaceBassdoesnt the switch have to have QoS too?
13:33.18*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
13:34.24jontownot if the linux machine is the router, but yes.. it technically does, if not.
13:34.59*** part/#asterisk Maksim (n=max@213.142.207.20)
13:35.04SpaceBassI thought about playing with QoS here at home... haven't gotten around to it
13:35.27Jason357so, you would suggest that I add a second network card and use the asterisk linux box as my router too?
13:35.27SpaceBasswhat I really need is a vlan but I'm too lazy to go identify ports on my switch
13:35.48SpaceBassJason357 how big is this network? what is your existing router?
13:36.41Jason357spacebass, home network
13:37.05Jason357existing routers are sipura and a linksys wifi routerr
13:37.22Jason357i take that back, it is a d-link wifi router
13:37.25SpaceBassJason357 probably no real need for QoS traffic-wise, but I'm all for playing... if its a wrtg54 or what ever, hack it and you can get QoS on it
13:37.38SpaceBassahh scrap the hacking the router part
13:38.46*** join/#asterisk b0ef (n=b0ef@062016141085.customer.alfanett.no)
13:40.02*** part/#asterisk secure75 (n=mic@ppp-82-135-14-79.mnet-online.de)
13:41.30Jason357I had a problem with my old ATA on packet8, since there was no QoS, any big data transfers would cause problems with the VoIP
13:41.58Jason357I have not yet had that problem with with the sipura, which has QOS
13:41.58SpaceBassyou might consider throwing a 2nd nic in the * box and getting a wrt54g linksys, hacking it and turning on qos
13:42.05SpaceBassnow that I think about that, i might consider that too
13:42.22SpaceBassalthough I've always heard that those linksys dont do well with too many clients
13:42.54Jason357so, why do I need to add the second nic and get a QoS router?
13:43.25Jason357shouldn't a QoS router be enough?
13:44.27SpaceBassyeah, if your going to take the * off the other router, or you have to put the other downstream
13:44.53Jason357right now, i am not using *
13:45.03SpaceBassgotcha
13:45.03Jason357just broadvoice directly with their Sipura
13:45.47SpaceBassah
13:45.55Jason357it goes Sipura connected direct to cable modem.  telephone connected to Sipura and D-Link connected to Sipura
13:46.25SpaceBassgotcha
13:46.27Jason357the Sipura prioritized all the VoIP traffic, but that is easy to do because all the traffic passes through the Sipura
13:46.31SpaceBassand you have traffic issues now?
13:46.36SpaceBassgotcha
13:46.38Jason357no, no problems now
13:46.39*** join/#asterisk kerspoon (n=kerspoon@213-232-83-17.dsl.prodigynet.co.uk)
13:46.46Weezeyopen source rules.
13:46.50*** join/#asterisk ccfiel (n=ccfiel@210.5.72.36)
13:46.57ccfielhello ppl..
13:47.08ccfielis it possible to connect..2 * box?
13:47.09Jason357I did have problems when I was on packet8 configued D-link to cable, and ata into d-link
13:47.18Weezeyccfiel: sire
13:47.21Weezeyccfiel: sure too
13:47.31*** join/#asterisk Arnaud (n=StorM@alf94-3-82-66-251-138.fbx.proxad.net)
13:47.54ccfielWeezey: how..? we have 2 * box... different location...i want to connect both of them..:)
13:47.58Jason357so, I fear if I connect the * box to my network in the same manor that the ata was connected that I will have the same problems
13:48.04bkw_a dingo ate my baby!!!
13:48.21*** join/#asterisk JakBeatZ (n=JakBeatZ@216.7.194.254)
13:48.30*** join/#asterisk aminorex (n=tony@66-191-69-132.dhcp.dlth.mn.charter.com)
13:48.46Weezeyccfiel: you can use IAX2 trunking, or SIP contexts, there's lots of ways.
13:49.02*** join/#asterisk critch (N=critch@steven.basesys.com)
13:49.11*** join/#asterisk Brumle (n=brumle@brumle.com) [NETSPLIT VICTIM]
13:49.38ccfielWeezey: what you mean IAX2 trunking?
13:49.41SpaceBassJason357 so really your bottle neck is at your gateway... sounds like you need traffic shaping on your router
13:49.48*** join/#asterisk twisted (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted)
13:49.48*** mode/#asterisk [+o twisted] by ChanServ
13:50.20ManxPowerJason357: There is only so much you can do about QoS issues on a consumer internet connection.
13:50.20*** join/#asterisk Dobaj (n=root@avonstreet.plus.com)
13:50.22Weezeyccfiel: http://www.voip-info.org/wiki-IAX
13:50.39JakBeatZFolks, is there a way to get another sequence number in a dialplan to run after the channel hangs up?  What I'm trying to do is a System() call to email a recorded meetme conference that happens in the sequence above, but the sequence number following the meetme() never gets run after a hangup is detected.  Any way around this or is there another method where I can achieve this?
13:50.43Jason357SpaceBass, just get a router that will let me give priority to one port and plug the * box into the port with priority?
13:51.06Dobajanyone installed Asterisk on a RaQ server with a connection to an external phone line?
13:51.46critchwhat is a blue alarm, and why would I see it on a otherwise stable pri?
13:51.52*** join/#asterisk nwhit (n=nwhit@ns1.whittrio.com) [NETSPLIT VICTIM]
13:52.01*** join/#asterisk grimse_ (n=grimse@p5481FB5F.dip.t-dialin.net)
13:52.19coppiceblue alarm == telephone sex
13:52.54coppicea blue alarm indicated a failure on a remote link
13:53.00critchfun
13:53.09ManxPowercritch: I've seen a blue alarm a couple of times.
13:53.24ManxPowerI called our teleco rep at home and he had it fixed in an hour 8-)
13:53.34SpaceBassJason357 basically, but in this case port means TCP/UDP port but ManxPower is right, sometimes there just isnt anything you can do... I suspect you have enough bandwitdh for VoIP (at least!) so a little prioritization would go a long way
13:53.48*** join/#asterisk jeffik (n=Jeff@toronto-HSE-ppp3943426.sympatico.ca)
13:53.53SpaceBassJason357 www.ipcop.org is a great router if you have an older box laying around as is m0n0wall
13:53.53coppiceif you have a blue alarm the telco should alreadty know about it
13:53.58critchCool, means I need to get my boss to call the owner of the line and get them to fix the link.
13:53.59ManxPowerTECHNICALLY QoS can only happen on SENT data.
13:54.07ccfielWeezey: im a little confuse on how to implemet this..
13:54.32*** join/#asterisk pstew (n=paul@ip68-4-20-95.oc.oc.cox.net)
13:54.47coppiceManxPower: TECHNICALLY QoS can only happen on data just before it is sent
13:55.06*** join/#asterisk nesys (n=nesys@81-174-12-111.f5.ngi.it)
13:55.07Weezeyccfiel: create an IAX2 context on each box for the other, register them to one another, then pass calls from box A to box B.
13:55.10Jason357I sell refurbished computers and buy P2 and P3 machines by the pallet, so... I can use an infinate number of 300-800mhz machines
13:55.41coppiceI just threw an 800MHz machine in the trash
13:56.05SpaceBassreally?
13:56.21WeezeyI still have a 900MHz in production.
13:56.26SpaceBassmy county is about to sell 1000 iBooks for $50 next week
13:56.32SpaceBassmy * is a 350mhz
13:56.47Jason357i now throw away anything less then 200Mhz
13:56.53SpaceBassam planning on getting two ibooks... going to move * onto one of them
13:57.24Jason357i find 800mhz machines to be very usefull
13:57.43SpaceBassyeah, I have a 250mhz laptop that I just cannot find a use for...
13:57.43greg_workget as many as you can, sell on ebay for even like $100, still a good deal
13:57.50SpaceBassthinking about a digital picture frame or something
13:58.01WeezeySpaceBass: that's a cool idea.
13:58.28ccfielWeezey: ok..do i need only one registration..for each server? if yes..would it not cause a busy line if an existing call is pass thru box a to b?
13:58.48*** join/#asterisk persia (n=persia@p3101-ipbf412marunouchi.tokyo.ocn.ne.jp)
13:58.51ManxPowerccfiel: Asterisl will happily send 1,000 calls at the same time between servers.
13:58.52*** part/#asterisk persia (n=persia@p3101-ipbf412marunouchi.tokyo.ocn.ne.jp)
13:58.58ManxPowerYour problem will be limiting that.
13:59.13*** join/#asterisk szi (n=rm@66.193.229.254)
13:59.31szi:)
13:59.34SpaceBassif I do use an ibook for a 2nd * box, I'll have to figure out how to pass calls b/t two *'
13:59.35Kattyhmm.
13:59.36SpaceBassss
13:59.38mmlj4does anyone have an installation checklist, i mean a comprehensive list of what's needed to do an * install (a box for *, phones, switches, UPS, etc. etc.)? I need to spec out a job
13:59.51ccfielManxPower: ic..even i just register one for each * box?
13:59.55coppiceI have a 750MHz vaio that is fine for speed, but the case has turned into a festering mass of corroded alloy, and a new battery is 50% of the price of a new machine
13:59.56Weezeyccfiel: no, they it's designed to allow multiple calls simultaneously.  
14:00.08*** join/#asterisk erickj_az (n=erickj_a@ip68-228-209-118.ph.ph.cox.net)
14:00.32SpaceBassi feel like there has to be some good use for this old pII laptop
14:00.45MikeJ[Laptop]SpaceBass. where is that??
14:00.45Jason357SpaceBass, did you suggest IPCop as a router?
14:00.52ccfielWeezey: so just need to set trunk=yes
14:00.54*** join/#asterisk mkrufky (n=mk@68.160.103.77)
14:01.07coppicenever never never ever under any circumstances buy a vaio
14:01.15*** part/#asterisk critch (N=critch@steven.basesys.com)
14:01.38Weezeyccfiel: if you don't, it would still work, it would just take more bandwidth between the boxes.
14:01.40Darwin35why is this coppiece
14:01.49SpaceBassJason357 I sure did
14:01.58*** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:01.58*** mode/#asterisk [+o anthm] by ChanServ
14:01.58Weezeycoppice: PC or notebook?
14:02.04SpaceBassMikeJ[Laptop] where is what?
14:02.15*** join/#asterisk szi (n=rm@66.193.229.254)
14:02.16MikeJ[Laptop]that they are selling those?
14:02.22Jason357SpaceBass, will that give me an option to give priority to all traffic from the * box?
14:02.25coppicesony rip you off for everything
14:02.25SpaceBassMikeJ[Laptop] Henrico Co. Virgina
14:02.32SpaceBassMikeJ[Laptop] Richmond basically...
14:02.43MikeJ[Laptop]yes, I have people down there.. .can you pm me info..
14:02.51MikeJ[Laptop]I may have somone go get me a few
14:02.57SpaceBassMikeJ[Laptop] they have limited the sale to henrico co. residents only...but I have two friends who live in the county willing to go with me for a free breakfast and some beer
14:03.15MikeJ[Laptop]yes, I have co-workers in ricmond
14:03.21JakBeatZAnyone know of a method to do something after a hangup is detected on an extension?  ie:  exten => 100,1,Meetme(1234,r) ; exten => 100,2,System(cat conf.wav | uuencode conf.wav | mail nobody@home.com)?  Sequence 2 will not run after a hangup from the meetme.  Anyone know a way around that?
14:04.00coppicea sony battery supplied by Dell - $60, with a year's warranty
14:04.01WeezeyJakBeatZ: h,1
14:04.02coppicea sony battery supplied by sony - $350 and 7 days exchange for faulty good
14:04.04coppicethe sony battery fails much more often than the dell, even though sony make both
14:04.11*** join/#asterisk hhrp (i=hhrp@nsc66.147.83-189.newsouth.net)
14:04.24JakBeatZWeezey: Thx.
14:04.35hhrphi guys
14:04.42ManxPowerWe have a bunch of users that have not checked their voicemail in months.
14:04.56SpaceBassJason357 yes, Ipcop has some great traffic shaping features... for instance I get top priority to my VPN connection to work and then to VoIP
14:04.57WeezeyManxPower: because they get them via email?
14:05.06ManxPower"I also considered just calling them up and asking them when they stopped working for the company.  I figure that would get their attention but doubt management would like this method." --me in an e-mail
14:05.07coppiceManxPower: sounds like the philippinnes :-)
14:05.11SpaceBassJason357 works very well... or course I'm also on a 8mbs/1.5mbs connection
14:05.22hhrpcould you advise if asterisk can register on a SIP gw as a client and then incoming calls come in it would switch calls to H323 GW?
14:05.28ManxPowerWeezey: I don't know if any of these people even have an e-mail account with the company.
14:05.42erickj_azDoes anyone have any experience with two IP gatewats in an asterisk machine?  The first works great, the second will not register sip clients.  Both are on seperate NICS and I can ping and SSH to either.
14:06.00ManxPowererickj_az: 1.0.x or CVS-HEAD?
14:06.06erickj_azCVS-HEAD
14:06.21ManxPowererickj_az: less than a week old CVS-HEAD?
14:06.36erickj_azSomewhere about a week old
14:07.04ccfielWeezey: how about my dial plan..example ... in my box A to call a SIP(local) name phone1 i would used this..in my dial plan in box A...exten => s,1,Dial(SIP/phone1,20,t) what would be my dial plan in box B..to call phone1?
14:07.06ManxPowerupgrade (since everyone will tell you to do that) and then report the problem to the mailing lists, if no help/response, report it on bugs.digium.com
14:07.11Jason357SpaceBass, you recommend IPCop over XORP or MyBSD?
14:07.27*** join/#asterisk _-Jon-_ (i=jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com)
14:07.37*** join/#asterisk rmonteux (n=Administ@193.120.146.179)
14:07.37Darwin35MYBSD not heard of that one
14:07.47ManxPowerccfiel: Dial(IAX2/userid@otherbox/phone1,20)
14:07.53rmonteuxhowya
14:08.04ManxPowerthere [otherbox] is defined in iax.conf
14:08.10ManxPowerthere == where
14:08.11erickj_azOK can I just upgrade asterisk or do I need to do zap and prilib as well?
14:08.17SpaceBassJason357 I haven't used XORP (nor heard of it)... its simple, has a nice web interface...  I am thinking of moving to m0n0wall... but I like the proxy plugin with ipcop
14:08.47ccfielok..thanks!
14:08.50ccfielbye all
14:08.57_-Jon-_hey everyone..  I'm having an odd problem with BroadVoice and wondering if someone can take a look at something..  Asterisk registers fine, but when I try and call my number it doesn't ring and I get my BroadVoice VM right away..  Can someone look at what 'sip debug' spit out and see if they can pinpoint the problem?
14:08.57Jason357why move to m0n0wall?
14:08.59ManxPowererickj_az: always make sure you have the same version of asterisk, libpri, and zaptel
14:09.23*** join/#asterisk PakiPenguin (i=uppal@unaffiliated/pakipenguin)
14:09.25_-Jon-_http://pastebin.ca/19704 if you want to take a look
14:09.31SpaceBassJason357 b/c I've been thinking about building an embedded device and m0n0 is great for that
14:09.34erickj_azOK.  Is there an easy way to move my pastches for rxfax and my sangoma card?
14:09.44djinDid someone with sms knowledge log in recently?
14:09.45erickj_azor do I need to re-patch the files?
14:09.49SpaceBass_-Jon-_ pastbin.ca the config in here
14:09.57puzzledanyone noticed that stable cvs does not compile?
14:10.04ManxPowerHiya, djin
14:10.25djinHi, ManxPower. How are you?
14:10.38ManxPowerdjin: waking up.  mornings should be outlawed.
14:10.41_-Jon-_http://pastebin.ca/19705
14:10.45tzangerm0n0wall's nice I'll admit
14:10.49djinChecking my inbox for your mail every day ;)
14:11.20Jason357well, I best get to work on my net network configuration :)
14:11.40Jason357s/net/new/
14:11.54hhrpcould you advise if asterisk can register on a SIP gw as a client and then incoming calls come in it would switch calls to H323 GW?
14:11.55gsthmmm... is there a possibility to keep the username & domain of the From field of an incoming SIP call if the user isn't a peer but a guest?
14:12.04ManxPowerdjin: I almost never see people here that use SMS w/Asterisk. 8-(  Did you try the mailing lists and/or mailing list archives?
14:12.10gstcurrently asterisk seems to rewrite this to asterisk@ip-address
14:12.17ManxPowerhhrp: YES!
14:12.22hhrpthanks
14:12.33ManxPowergst: no.
14:12.57djinManxPower, yes. And suggested solutions didn't work for me.
14:13.04Jas_Williams_-Jon-_: you have no s extension in from-bv context
14:13.05ManxPowerdjin: that sucks. 8-(
14:13.08hhrpi did make samples when i installed it, does that mean it created bunch of unnecessary configs that i would need to fix?
14:13.08*** join/#asterisk Chickens (n=NNSCRIPT@c-24-19-96-100.hsd1.wa.comcast.net)
14:13.47Jason357* behave on an IPv6 network?
14:13.52djinManxPower, indeed. Some Dutch say 'I've got it working' without providing the code :(  (typical Dutch ;)
14:13.56ManxPowerdjin: there are a couple of commercial SMS/Asterisk gateways out there.  For USA users they are MUCH more expensive than regular text messsages, but for EuroPeople it should be a good deal.
14:14.03gstManxPower: hmm.. ok - than i'll continue modifiying the src :)
14:14.10_-Jon-_Jas_williams, I have 2 numbers with BV and I want each to ring different extentions though
14:14.22djinManxPower, true. But I'm trying just inbound for now.
14:14.29gstManxPower: currently i have the problem that i'm not able to read channel vars in initreqprep in chan_sip.c
14:14.35ManxPowerdjin: Inbound?  eek!
14:14.36Jas_Williams_-Jon-_: What do you have as the register line in sip.conf
14:14.56gstManxPower: i tried to add some code which uses the var SIPFROMUSER and SIPFROMDOMAIN if present
14:15.08_-Jon-_Jas_williams, just register => username:pass@broadvoice..  I have no /extention set
14:15.14djinManxPower: /me was up for a challenge ;)
14:15.27gstsomehow pbx_builtin_getvar_helper(p->owner, "SIPDOMAIN") just returns NULL :(
14:15.35gst(when called in initreqprep)
14:15.49hhrpwhat's the minimum configuration that I need to do just to register on a SIP gw by proxy and password? add register => ?
14:16.08Jas_Williams_-Jon-_: Then set an extension one for each number add that extension into the from-bv context and handle each one as you want
14:16.14*** join/#asterisk Chickens (n=NNSCRIPT@c-24-19-96-100.hsd1.wa.comcast.net)
14:16.42SpaceBassJason357 If the distro supports IPv6 then probably, but doubt many phones support it
14:17.10_-Jon-_Jas_Williams, what I have for each number is: exten => <number>,1,Answer()...  Is that what you mean?  Because when I used to have LiveVoip that's how it was set up and it was fine
14:18.42Jas_Williams_-Jon-_: on sip that will not work, add to the end of your  register => username:pass@broadvoice/number with a register for each one then it may work as you whant
14:19.03_-Jon-_Jas_williams, alright I'll give that a try.  Thanks!
14:19.41Jason357If I use an ATA instead of a SIP phone, is it still possible to utalize all the * features?  i.e., how can I put someone on hold from a standard phone hooked into an ATA?
14:19.51hhrpwhy would astman not connect to my server?
14:20.00hhrpi have them on the same machine
14:20.05Hmmhesayswhat do y'all think about audiocodes?
14:20.26Jas_WilliamsJason357: press the R button or flash the hook dependant on the ATA
14:20.31Darwin35I need a job
14:20.38Darwin35I need money
14:20.42hhrp:)
14:20.46Darwin35I need love
14:20.50Darwin35I need honey
14:20.56coppiceI need tea
14:21.00KattyHmmhesays: if i knew exactly what you were talking about, i could reply.
14:21.25hhrpcould anyone help me a little with started config on newly installed asterisk 109? plz pm me
14:21.56Jas_Williamshhrp read the manuals
14:21.57DarthClueHmmhesays: haven't ever touched one, but have heard both slightly good and really bad about them.
14:22.01Jas_Williams~docs
14:22.02jboti 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
14:22.04hhrpim reading
14:22.08hhrpstill dont work
14:22.30_-Jon-_Jas_Williams, I just tried and no such luck still :(
14:22.40*** join/#asterisk Netrunz (i=Netrunne@195.137.145.50)
14:22.45Netrunzhelloù
14:22.46Jas_Williams_-Jon-_: what is the sip debug now
14:22.52NetrunzI got a question...
14:23.03*** join/#asterisk santiago (n=santiago@63.245.86.153)
14:23.07Netrunzhow do i change
14:23.13Netrunzthe default country?
14:23.23Netrunzin asterisk 1.0.9?
14:23.25Darwin35in wich file
14:23.25*** join/#asterisk cpm (n=Chip@orthanc.avwashington.com)
14:23.37*** join/#asterisk jeanmi_ (n=besnard@front.tekkno.net)
14:23.38Darwin35for what codec ?
14:23.46_-Jon-_Jas_williams, actually get this..  I added the 2 register lines, and when I call the alt # sip debug shows nothing
14:23.54hhrpwhat is this error ug 10 10:23:25 NOTICE[13806]: indications.c:397 ast_unregister_indication_country: Removed default indication country 'us'?
14:24.01ManxPowerNetrunz: in the correct config file for the device.
14:24.18ManxPowerhhrp: that's harmless message.  ignore it.
14:24.21NetrunzManxpower it gives me the error hhrp just reported
14:24.22hhrpok
14:24.27Netrunzoh ok
14:24.31Netrunzand after tha
14:24.32Netrunzt
14:24.35Netrunzit says
14:24.40Jason357can I run IPcop on the * hardware?
14:24.45Netrunz- Setting default indication country to 'us'
14:24.58Hmmhesaysd
14:25.01HmmhesaysDarth same here
14:25.12HmmhesaysI guess I get to become an audiocode expert now
14:25.20hhrpi did make samples after install is there a way to roll it back?
14:25.23Jas_Williamshhrp: Netrunz try looking in indications.conf
14:25.35Netrunzok
14:25.41*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfnja.dialup.mindspring.com)
14:25.55coppiceHmmhesays: experts should get the name right. you missed off the "s" :-)
14:26.21Hmmhesayscoppice, you lost me
14:26.25Hmmhesayslol, what are you talking about
14:26.35coppiceits audiocodes
14:26.42*** join/#asterisk wizhippo (n=wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca)
14:26.42_-Jon-_Has anyone actually managed to route an alternate BV number to a different extention?
14:26.48Hmmhesaysoh? interesting, thanks for the info
14:26.57NetrunzJas is this the general indication for country? I mean does it affect all asterisk?
14:26.58Hmmhesaysso "audiocodes" is not plural it is singular
14:27.11*** join/#asterisk zoo (i=nobody@ip-161-16.travedsl.de)
14:27.19coppicethis appears to be their name. I guess they have more than one code :-)
14:27.47*** join/#asterisk PakiPenguin_ (i=uppal@unaffiliated/pakipenguin)
14:27.58Hmmhesaysi'm actually kind of interested to work with them, i grow bored with the current product line we normally sell
14:28.30Jas_WilliamsNetrunz: yes unless it is overridden by another config file read after this one
14:28.41hhrpis there a way to roll back 'make samples'?
14:28.50ManxPowerAsterisk's local support is scattered around.
14:28.57*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
14:29.15ManxPowerThere is indications.conf that ONLY controls tones (ringtone, busy tone, etc) and then there is the specific LANGUAGE setting for each device/port.
14:29.21Jas_Williamshhrp: rm /etc/asterisk -r then  make samples again
14:29.27Hmmhesayslook in /etc/asterisk debian doesn't overwrite the config files
14:29.50*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
14:29.58Netrunzbeside that i do not understand why anytime i place a call i get this message (although it works correctly): NOTICE[27211]: channel.c:1703 ast_set_write_format: Unable to find a path from slin to g729
14:29.58NetrunzAug 10 18:28:16 WARNING[27211]: indications.c:77 playtones_alloc: Unable to set 'SIP/6250000402-5db1' to signed linear format (write)
14:30.25NetrunzI have forced g729 EVERYWHERE!
14:30.38Netrunzwhy does it try slin
14:30.39ManxPowerNetrunz: that means you don't have indications.conf or it's wrong
14:30.44*** part/#asterisk PakiPenguin_ (i=uppal@unaffiliated/pakipenguin)
14:30.49Jas_WilliamsNetrunz: Turn off g729 on your sip device or buy a license for *
14:30.51Hmmhesaysindeed
14:30.52ManxPoweroh! no, that's not indications.conf
14:31.03*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-70-18-115-90.buff.east.verizon.net)
14:31.06ManxPowerNetrunz: What options are you giving to Dial?
14:31.15NetrunzJas I have licence on the phone and my remote provider does. I just need to do passthough
14:31.18SuPrSluGhello all
14:31.30NetrunzManx only timeout
14:31.43Craziman2Jas_Williams:  why do you have to have a license for * to use g729?
14:31.43hhrpif i have a SIP gw that im registering at should it be PEER or USER?
14:31.44Jas_WilliamsNetrunz: you dial command is trying to provide ringing from asterisk
14:31.46*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
14:31.47ManxPowerNetgeeks: you're not trying to answer the channel, are you?
14:32.06shadebobooanyone can explain me the resample algo in tonezone.c?
14:32.19ManxPowerNetgeeks: i.e. if you don't have a G729 license, playback, background, saydigits, voicemail, etc will NOT work if you are using G729 and don't have a license.
14:32.32NetrunzJas my dial command is exten => _3X.,1,Dial(SIP/tzm1/${EXTEN})
14:32.53Jas_WilliamsNetrunz: I thing Manx is talking to you
14:33.06ManxPowerNetrunz: paste the cli output of a failed call to pastebin.ca
14:33.07NetrunzI thought he was talking to netgeek
14:33.29Netrunzok Manx i will
14:33.31Netrunzhold on a sec
14:33.32ManxPowerIt's amazing how much time people will waste in order to save a couple of dollars on G729 licenses.
14:33.42mutreally
14:33.53toddfsome people don't have that option. Iaxy's for example don't have the codec.
14:33.58Jas_WilliamsManxPower: or a few pounds on bandwidth
14:34.10hhrphow can i uninstall asterisk?
14:34.16ManxPowerhhrp: you can't.
14:34.18NetrunzJas I will buy the licence but my provider MAKES me use g729
14:34.27ManxPowerhhrp: unless you want to track down every file it installs.
14:34.40mutit installs one?
14:35.13hhrpok
14:35.26Jas_WilliamsNetrunz: Bah lets see what your cli output is then use pastebin
14:35.27mutmake your own quick uninstall script
14:35.35Netrunzhttp://pastebin.ca/19710
14:35.46jeanmi_MySQL query failed: Table 'cdr.egress' doesn't exist
14:36.02jeanmi_where can I find the structure for the different mysql tables used by asterisk ?
14:36.28*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
14:36.37ManxPowerNetrunz: we need more stuff that happens before the stuff you pasted.
14:36.40skram|phpmyadmin?
14:37.24skram|or 'mysql' from the unix #/$
14:37.24Netrunzok let me do another call
14:37.24ManxPoweri.e. we need to know what's LEADING UP TO the error
14:37.24mutmake install | grep [c]p | sed -e "s/cp /rm /" > uninstall.sh
14:37.24Netrunzand i'll stick all ok?
14:37.24mut:P
14:37.24skram|jeanmi_ ^&^^^
14:37.28hhrpanyone uses asterisk with stanaphone?
14:37.29mutlike magic..
14:37.42jeanmi_skram| ????
14:38.00skram|hhrp: I have.
14:38.02Netrunzhttp://pastebin.ca/19712
14:38.28ManxPowerNetgeeks: and that dial is at priority 1?
14:38.45Netrunzyup
14:39.00tzangerEYEAM TEH SHIZNAT
14:39.04*** join/#asterisk MichaelSaunders (n=mick@staff.tsn.cc)
14:39.07Netrunzexten => _3X.,1,Dial(SIP/tzm1/${EXTEN})
14:39.08Netrunzexten => _3X.,2,Hangup
14:39.08ManxPowerNetgeeks: then we need your sip.conf entry for [200]
14:39.15Netrunzok
14:39.15sivanawhat a P mean in cvs update?
14:39.18tzanger(MAP_FAILED is ffffffff) mmap /dev/mem...(ffffffff) and file...(00d2b000)
14:39.18tzangerconfiguration area @ 0x80000fc0-0x80000fff
14:39.18tzanger80000fc0:  01 01 02 00 00 ff 21 62 00 04 00 01 00 65 01 03
14:39.18*** join/#asterisk grimse (n=grimse@p5481FB5F.dip.t-dialin.net)
14:39.23tzangerPH33R
14:39.32tzangersivana: prune empty directories/files
14:39.35ManxPoweroh and for [tzml]
14:39.40sivanaok
14:39.40Jas_WilliamsNetrunz: remember to blank secrets
14:39.41WeezeyIf asterisk dumps a core, what do I need to do so report it on bugs.digium usefully?
14:39.51ManxPower..er...[tzm1]
14:39.56sivanatzanger: P utils.c
14:40.03tzangeroh
14:40.05tzangerpatched (updated)
14:40.09Netrunzhttp://pastebin.ca/19713
14:40.15sivanaok
14:40.29MichaelSaunderstzanger: do you know how to you sipsak to test users on asterisk
14:40.38ManxPowerNetrunz: I have no idea why that is happening.  Go buy a license or 10
14:40.50Netrunzok where can i buy it from?
14:40.54*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
14:40.54Netrunz:)
14:41.07DarthClueNetrunz: digium
14:41.16Netrunzu mean if i install a license i shouldn't have this anymore?
14:41.17ManxPowerNetgeeks: Digium.  $10 per simul channel
14:41.31_-Jon-_does anyone here have an alternate number with BroadVoice?
14:41.32ManxPowerNetgeeks: asterisk think it needs to transcode.
14:41.49Netrunzmanx that i don't understand why though
14:41.55SpaceBass_-Jon-_ I have a few
14:41.56*** join/#asterisk secure75 (n=mic@ppp-82-135-14-79.mnet-online.de)
14:41.58ManxPowerNetgeeks: try this.  replace the Dial with a Wait(120), then do a "sip show channels".
14:42.03ManxPowerlook at the codec listed.
14:42.10Netrunzok hold on
14:42.12_-Jon-_SpaceBass, are you able to make each number do different things in your dialplan?
14:42.26ManxPowerNetrunz: what you are trying to do should work, but what happens when the SIP device wants to call voicemail?  Then you'll still need a license.
14:42.29SpaceBass_-Jon-_ with alternate numbers, no, b/c bv doesnt sent DID info
14:42.38SpaceBass_-Jon-_ with 2 seperate accounts, you can do that
14:42.56tzangerMichaelSaunders: no idea.. I avoid SIP
14:43.10_-Jon-_SpaceBass, crap, well this just sucks
14:43.17muti could really got for a footlong chili cheese dog right now
14:43.18muto man
14:43.34mutsome reason that popped in my head and my mouth is watering
14:43.35ManxPower_-Jon-_: you tried adding /1234 to your BV register line?
14:43.38SpaceBass_-Jon-_ yeah, I ran into that problem... so I just got a 2nd byod lite plan
14:43.46*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
14:43.53SpaceBassManxPower that works for seperate lines but not for alternate numbers
14:44.06ManxPowerSpaceBass: ah.
14:44.07_-Jon-_ManxPower, Yup, tried that
14:44.12Netrunzmanx 192.168.2.29     200         6cbc93303b1  00101/05622   g729
14:44.28ManxPowerNetrunz: I have no more suggestions
14:44.38Netrunzok i will just go for the license :)
14:44.40Netrunzthanks
14:44.40_-Jon-_Hmmm, I suppose what I could do is direct it to an extention based on the number calling me
14:44.51*** join/#asterisk riaan (n=riaan@196.36.161.235)
14:44.58_-Jon-_Anyone calling my PA number will obviously live in PA
14:45.07MichaelSaunderstzanger: What do you use
14:45.27tzangerIAX2
14:46.42*** join/#asterisk ccfiel (n=ccfiel@210.5.72.36)
14:46.47SpaceBass_-Jon-_ the byod line plan is $5 and includes unlimited incoming... thats the best workaround I've found
14:46.54ccfielWeezey: still there? :)
14:47.34ccfielWeezey: is there a way... to do..this  box A <---- iax -----> box B ..
14:47.52ccfielboth are * box..
14:48.05ccfieli want to connect 2 * box..
14:48.12_-Jon-_SpaceBass, hmmmm I just might look into that!
14:48.27*** part/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com)
14:48.37Hmmhesaysyou say Hmmmmm a lot
14:48.41SpaceBass_-Jon-_ it does get expensive... but seems to work well... i just wish I had known that before I bought the 800 number on the wrong account!
14:49.02*** join/#asterisk klasstek (n=nunyobiz@sta-206-168-218-206.rockynet.com)
14:49.41_-Jon-_SpaceBass, I just checked..  It says $5.95 and is only 100 minutes
14:50.21*** join/#asterisk santiago (n=santiago@63.245.86.153)
14:50.37ccfielhello..ppl...can somebody help me with my problem..:(
14:50.49ccfieli want to connect my 2 * box..
14:51.20DarthClueccfiel: take a look at the wiki
14:52.08ccfielDarthClue: yes i have.. its just not clear to me..
14:52.56ccfielDartClue: i know i will have a problem with NAT..
14:53.03ccfielwith SIP..
14:53.11SpaceBassccfiel use IAX2
14:54.07ccfielSpaceBass.. this is the ..situation... in my BOX A... all local connection..is SIP..
14:54.35ccfielSpaceBass...in my BOX B...all local connection is SIP..
14:54.40DarthClueccfiel: http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers
14:55.24ccfielspaceBass..: what i want is connect them..with IAX2...but retain..the local SIP..in each box.
14:55.25*** join/#asterisk Ishii (n=nox@igate.bellsimons.com)
14:55.26jeanmi_I have installed asterisk using the debian packages but while staring asterisk I can see errors related to missing mysql db/tables. Where am I supposed to find the db/tables creation scripts ?
14:55.32ccfielis it possible?
14:55.32*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
14:55.49DarthClueccfiel: http://www.voip-info.org/tiki-index.php?page=Asterisk+-+dual+servers
14:56.04*** join/#asterisk docelm0 (n=docelm0@67.106.194.90.ptr.us.xo.net)
14:56.18ccfielDarhClue: thanks..i think ...it would answer..my question.
14:56.21docelm0anyone in here have experience with asterisk and TNT's?
14:57.06*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
14:58.15_-Jon-_Hmm okay would this work..  have exten => s,1,Answer().... and then exten => s/_610XXXXXXX,1,Answer()?
14:58.34erickj_azI have built the new CVS and still have the same issue with two IP gateways.  The first works great, the second will not register sip clients.  Both are on seperate NICS and I can ping and SSH to either.
14:58.35shadebobooanyone can explain me the zt_tone_def in zonetone.h?
14:58.58erickj_azAny other suggestions from anyone?
15:01.16docelm0eric what are your configs?
15:01.32fenlanderSigh. Somebody deleted a load of consultants from the wiki when they moved pages around
15:01.32docelm0Have you tried debugging it to see if its a registration problem?
15:01.57fenlanderIf you thought you were listed on the wiki you may wish to check again.
15:02.43erickj_azIt seems that asterisk might be sending regerstration using the first GW when trying to register on the second GW
15:02.51erickj_azBut I'm not sure about that.
15:03.08docelm0What protocol are you using?
15:03.10*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
15:03.15docelm0SIP/IAX/H323/???
15:03.24erickj_azSIP
15:03.35docelm0Do you know how to debug SIP packets?  
15:03.41erickj_azNope
15:03.50erickj_azI'm a beginner
15:04.16*** join/#asterisk wunderkin (i=kev@12-215-218-160.client.mchsi.com)
15:04.17docelm0Have fun..   You need to be able to debug to figure out this problem.  I am having a issue with a TNT/Asterisk and am having a time with that
15:04.34docelm0If you can debug it will tell you the raw errors and help you find your solution
15:04.47erickj_azHow do I learn to debug SIP?
15:05.03hhrpso asterisk doesnt have G729 support?
15:05.16puzzledit has if you buy the licenses from digium.com
15:05.25hhrphow much is it ~?
15:05.32puzzled$10 per channel
15:05.36docelm0It does but you have to pay 10 per channel
15:05.45hhrpthanks
15:05.45tobiasWolfdjin: still there ?
15:06.15*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
15:06.33*** part/#asterisk Netrunz (i=Netrunne@195.137.145.50)
15:07.44blitzrageanyone from Digium around?
15:08.41*** join/#asterisk Netrunz (i=Netrunne@195.137.145.50)
15:08.50sivanaany idea why I would get dropped calls with Normal Clearing dialstatus
15:09.12Netrunz:
15:09.17Jas_Williamssivana: Carrier Far end cleared the call, or you cleared the call
15:09.17Hmmhesaysbecause one of the endpoints is fxo and it thinks that the end user has hung up?
15:09.18docelm0did the call connect?
15:09.42hhrphow do i reload * from console? so my newly configured SIP gw engage?
15:09.53Netrunzhhrp reload
15:09.58hhrpthats all?
15:10.02Jas_Williamssivana: do you have busy dectect on in zapata.conf
15:10.03Netrunzyup
15:10.05hhrpthanks
15:10.10sivanaJas_Williams: let me check
15:10.12hhrpi wasnt sure if it does it
15:10.33Netrunzyes. you can also do reload extensions for extension.conf
15:10.35sivanaJas_Williams: I have no entry
15:10.46Netrunzor sip reload
15:10.57Jas_Williamssivana: good it's not that then :)
15:11.05*** join/#asterisk Defraz (n=t0tal@tim.ibccom.net)
15:11.35sivanaJas_Williams: http://pastebin.ca/19717
15:13.35*** join/#asterisk archimedes (i=archimed@adsl-70-247-240-218.dsl.ltrkar.swbell.net)
15:14.16jalsotfor a saturated adsl link which codec can fit better? ilbc or speex? [through IAX2]
15:14.32hhrpi dont quite understand, where exactly do i configure my extention? I need to configure an extention that is H323 GW authorization by IP?
15:14.38*** join/#asterisk mithro (n=tim@c213-100-42-188.swipnet.se)
15:15.25*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
15:15.47erickj_azCould it be that CVS is not compiling correctly?
15:15.54docelm0Any TNT * people in here?   I have a small question
15:16.09erickj_azI missed that part of the chat
15:17.18*** join/#asterisk sigmounte (n=sigmount@www.sighq.net)
15:17.37*** join/#asterisk stevej (n=stevej@67.97.36.238)
15:17.57puzzleddocelm0: not me but if you search the mailinglist you will find a number of people who use TNT's
15:18.33docelm0k..  ya Im having a small RTP issue
15:18.53*** join/#asterisk gerwinin (n=gerwin@84.87.179.14)
15:19.50erickj_azHow does one tell what CVS date they ar running?
15:19.59*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
15:20.08tzangererickj_az: show version at the * CLI
15:20.19DarthClueshow version at the cli or show file version if you are running cvs-head
15:21.40erickj_azshow version: Asterisk CVS-HEAD built by root@pbx1.intrepidvoice.net on a i686 running Linux on 2005-08-10 13:54:56 UTC
15:22.08erickj_azBut show file file version is an error
15:23.01DarthClueerickj_az: 'show version files', my bad
15:23.16lathos42Katty: ooh, shiny
15:24.25Uther_PO_o
15:24.51brettnemdocelm0: I'm turing up my first TNT right now actually
15:24.59cpatryshow version files file.c to get a rev for a specific file too.
15:25.06archimedesanybody here work with the softfax patch addon for asterisk? asterisk keeps giving me compile errors when the patch is applied.
15:25.10erickj_azOK now wehre do I find what the latest revision numbers should be?
15:25.54Uther_PKatty: is it running well on your laptop? (or have you not gotten that far yet)
15:26.14KattyUther_P: i don't have my laptop yet
15:26.18Uther_Poic
15:26.21Katty(=
15:26.30Hmmhesaysits the same, only the names have changed, everday it seems we're wasting away
15:26.53Hmmhesays:D
15:27.08_-Jon-_hey i got one more question for any BroadVoice users..  Anyone ever get an intermittent "We're sorry, this number has not been assigned" message before when calling your BV number?
15:27.18Katty:<
15:27.29Hmmhesaysman up nancy
15:28.01stevejCan anyone recommend a good (cheap!) soundcard to use with *? I have an onboard Via 8237 AC97 soundchip, running under Whitebox 4.0 (RHEL-4), and it's just plain garbage.
15:28.23cpatryerickj_az: cvs log file.c , on the head: line
15:28.43cpatryasterisk:/usr/src/asterisk# cvs log file.c |grep head:
15:28.43cpatryhead: 1.72
15:28.55Uther_Pone would think that since any soundcard can do a bandwidth FAR greater than a telephone line that any soundcard would do
15:29.10*** join/#asterisk cgcorea (n=cgcorea@205.240.200.107)
15:29.23stevejBut not all soundcards are supported well under Linux.
15:29.30tzangerUther_P: ?
15:29.33Hmmhesayseasy fix? get an ata
15:29.53Uther_Pah true... can't help ya there.. I use BSD
15:30.05Hmmhesayshaha, I have no response to that DarthClue
15:30.12Hmmhesaysyou got me, you got me good
15:30.37mutTE110P is 5v right?
15:30.55Hmmhesaysspeaking of t1 cards, cylogistics has not gotten back to me
15:31.02*** part/#asterisk Uberbot (n=Uberbot@pcp01880954pcs.sandia01.nm.comcast.net)
15:32.02*** join/#asterisk SkramX (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
15:32.23hhrpI need to configure an extention to where my calls gonna drop from a gw i configured in SIP.CONF do i have to configure that other gw in the same file?
15:32.52Hmmhesayswhich other gateway?
15:33.16*** join/#asterisk Romik (n=romik_@212.143.5.146)
15:33.16hhrpi need to configure a H323 gw
15:33.29Hmmhesaystheoretically no
15:33.32hhrpi successfully configured SIP gw in sip.conf
15:33.35*** join/#asterisk archi (i=archimed@adsl-70-247-240-218.dsl.ltrkar.swbell.net)
15:33.40*** part/#asterisk archi (i=archimed@adsl-70-247-240-218.dsl.ltrkar.swbell.net)
15:33.43hhrpi do sip show peers and see it
15:33.48MichaelSaundersanyone use sipsak with asterisk
15:34.11hhrpall i need now is to configure H323 gw and tell * to router all calls from sip gw to h323 gw
15:34.19kerspoonHi, just a quick question. What BRI (ISDN2) cards do you reccomend. I'm in the UK and having problems finding any.
15:34.25Hmmhesayswhich h323 channel have you chosen?
15:34.39hhrpi have not chosen anything yet
15:34.59hhrpim trying to understand where in which config i need to put my h323 gw
15:34.59Hmmhesayscheck out asterisk h323 channels on the wiki
15:35.15hhrpso i need a channel configured riht?
15:35.16puzzledkerspoon: active avm or eicon cards if you have the money or a cheap isdn card with a hfc-s chip on it
15:35.37Hmmhesayshhrp: you have to figure out which h323 chan you want to use
15:35.41kerspooncheers puzzled
15:35.46Hmmhesaysthere is oh323, h323 ooh323, woomera
15:35.55hhrph323
15:36.18Hmmhesaysthey all do h323 though
15:36.40Hmmhesayshttp://www.voip-info.org/wiki-Asterisk+H323+Channels
15:36.43hhrpi dont understand tho from a configuration stand point how would it know to route incoming calls to that gw
15:36.43Hmmhesaystake a looksee there
15:36.50hhrpthanks im looking now
15:37.01Hmmhesaysonce you have an h323 channel installed, then you just route it in the dialplan
15:37.22hhrpso i dont need any [phonex] in sip.conf
15:37.28Hmmhesaysdial($H323/<number>@<host>)
15:37.47Hmmhesaysno, sip.conf is for sip endpoints
15:38.05hhrpok thanks let me read i ll be back
15:38.16Hmmhesayspersonally I use oh323
15:38.25Hmmhesaysbut your results may vary
15:38.34brettnemhey, gpl question here.. if I design a custom app or channel that is really particualr to my business, do I need to disclose, share it or ?? what? like chan_tuna.so :)
15:38.37erickj_azI don't see how to compare to see if what I am running is the latest version.
15:38.52erickj_azCVS does not seem to be compiling
15:39.00*** join/#asterisk mago2-cn (i=maxgluck@200.109.167.9)
15:39.13Hmmhesayslook there its mago2-cn
15:39.29kerspoondamn. the active avm isdn cards is £450.
15:39.40*** join/#asterisk Malthus (n=herman@67-43-156-39.loudpacket.net)
15:39.42coppicemaybe GCC would compile better than CVS :-)
15:39.44*** part/#asterisk Malthus (n=herman@67-43-156-39.loudpacket.net)
15:39.53brettnemhar har
15:40.02erickj_aznot sure what GCC is?
15:40.11brettnemmaybe you should ask around in #cvs
15:40.22Hmmhesaysthey are messing with you erickj_az
15:40.28erickj_azI thought so.
15:40.32erickj_azThank guys
15:40.36erickj_azThat was fun.
15:40.42Hmmhesaysyour statement made it look like you are trying to compile using cvs
15:40.47Hmmhesaysas the compiler
15:41.03erickj_azand you didn't kno what I ment?
15:41.06*** join/#asterisk Temon1900 (n=Temon@dsl62-93-329.saudi.net.sa)
15:41.13Hmmhesaysof course we all knew what you meant
15:41.22Hmmhesaysbut whats the point in being serious 24/7
15:41.32Hmmhesaysthis isn't paid support
15:41.34brettnemwe're all a bunch of sarcastic fools
15:41.51erickj_azI understand, I'm just frustrated
15:42.00archimedesjoin the club...
15:42.04Hmmhesayserickj_az grab the latest head off of cvs and try again
15:42.37erickj_azOK I just tried that a few minuted ago and got the error:cvs checkout: cannot chdir to asterisk: Not a directory
15:42.39Hmmhesaysor get stable, and see if that compiles... if it doesn't you have other issues
15:43.12*** join/#asterisk Uther_P (n=uther_p@66.180.120.82)
15:43.15hhrphow can i see if * is connected to a sip gw at the moment thru console?
15:43.22*** join/#asterisk toddf (n=toddf@adsl-66-141-65-235.dsl.okcyok.swbell.net)
15:43.34Hmmhesaysregistered? of if it has a channel up
15:43.42*** join/#asterisk docelm0 (n=docelm0@67.106.194.90.ptr.us.xo.net)
15:43.46erickj_azI have an older version of CVS running and I can compile that.
15:43.54hhrpregistered
15:43.59Hmmhesayssip show peers
15:44.03hhrpyes, if its succesfully registered
15:44.15hhrpso that shows if it's hooked up or just confed in config?
15:44.21Hmmhesaysboth
15:44.22mago2-cnmisheu, mashlomja?
15:44.35erickj_azHooked up or recently hooked up
15:44.39erickj_azit shows
15:44.49Hmmhesaysuntil the sip registry expires
15:44.57hhrpi see
15:44.58hhrpthanks
15:45.41erickj_azcould there be somthing wrong with cvs where the directory is not available?  Does that happen often?
15:45.50Hmmhesayserickj_az: take a break, have a beer, eat some pizza, come back download cvs and try
15:45.55Hmmhesays*cvs-head
15:46.56Hmmhesaysif you're not into pizza, maybe a nice sandwich and a salad
15:47.06erickj_azSounds better
15:47.09archimedeshave a kitkat
15:47.14Katty:<
15:47.52archimedesnot you katty, the candy bar... although you might be sweet, who knows
15:47.58erickj_azHow long do they uasualy take to get CVS back up and running?
15:48.09erickj_azWhoever "They" are
15:48.21hhrptheres one version of h323 is already in * distro right?
15:48.36Katty:>
15:48.40Hmmhesayshhrp: kind of
15:48.46hhrpkind of?
15:48.47Kattyarchimedes: i'm not sweet
15:48.47Hmmhesayscouple hours erickj_az
15:48.53*** join/#asterisk Nukemizer (n=Nuke@67.137.28.163)
15:48.53Kattyarchimedes: i'm cranky, sarcastic, and a smart ass
15:48.59hhrpi have installed 1.0.9 should i have it?
15:49.06archimedesyeah sweet... seee :)
15:49.16Hmmhesaysooh323 is in asterisk addons, and chan_h323 is in the tree, but you need openh323 and pwlib
15:49.22erickj_azHumm..neeto!
15:49.27hhrpok
15:50.02*** join/#asterisk d3mize (n=verbalco@cpe-68-173-29-212.nyc.res.rr.com)
15:50.03erickj_azIs there a know issue with two or three routes for SIP registration?
15:50.10erickj_azknown issue?
15:50.20erickj_azHow would I go about finding that out?
15:50.23Hmmhesaysdefine "routes"
15:50.36Hmmhesaysmantis contains known bugs
15:50.42erickj_aztwo different IP's on two different gateways
15:50.42malandrohi does anybody knows where can i host cheap a voip pbx in asterisk?
15:50.47d3mizei have what's probably a really dumb question, but i've googled/asked a bunch of people/etc. and i can't find the answer anywhere
15:50.53Hmmhesays42
15:51.13erickj_azand two different NICs
15:51.19*** join/#asterisk generalhan (i=general_@63.133.146.82)
15:51.38d3mizecan i use an asterisk pbx box or something like that with callvantage or some other residential voip provider?
15:51.44Hmmhesaysas in voip gateways? or ip gateways
15:51.56erickj_azIp gateways
15:51.59Hmmhesaysd3mize: ask them
15:51.59*** join/#asterisk IPD (n=asterisk@vpn.expetel.com)
15:52.18hhrpso i need openh323 installed
15:52.24DarthClueKatty is very sweet, from a distance.  Get too close and expect to be hurt.
15:52.26erickj_azOne seems to register great while the other does not.
15:52.33KattyDarthClue: exactly
15:52.37erickj_azI can ping and ssh and mysql to either
15:52.44Hmmhesayshhrp: is that what you read?
15:52.44*** join/#asterisk BoDePlOt (i=user@pool-68-162-3-185.nwrk.east.verizon.net)
15:52.56hhrpyea seems so
15:53.04Uther_PDarthClue: like racoons and badgers?
15:53.09erickj_azbut only register sip clients with one.
15:53.13Kattymushroom mushroom
15:53.14Hmmhesayshhrp: ;)
15:53.21hhrpi jusrt dont understand why wouldnt they include it in install
15:53.27d3mizei really don't want to call callvantage, they're the biggest tools and they don't know anything
15:53.34Hmmhesayshhrp: once you dive deeper in you will realize
15:53.38d3mizeplus i'd have to hold for a few hours
15:53.39Uther_Pcute to look at, but very angry and rabid amist close proximity
15:53.59hhrp:)
15:54.21erickj_azAny ideas of where I should start looking?
15:54.24*** join/#asterisk tld (n=tld@195.84-48-78.nextgentel.com)
15:54.30DarthClueUther_P: haven't actually been close to a 'coon or badger, but I'm not sure I would want to find out.
15:54.35*** part/#asterisk zoo (i=nobody@ip-161-16.travedsl.de)
15:54.47Hmmhesaysreminds me of the family guy episode 2 nights ago, peter keeps getting attacked
15:55.08BoDePlOtdoes fxotune need to be run locally from the console (i.e. not work remotely over an ssh session?) - it complains "cannot fill input buffer" when I run it from ssh
15:55.26*** part/#asterisk d3mize (n=verbalco@cpe-68-173-29-212.nyc.res.rr.com)
15:55.43IPDanyone here know about h323 errors when firing up *
15:56.11Hmmhesaysdepends on which h323 channel you are using
15:56.14IPDI got a error - Loading module chan_h323.so failed! ..... broken pipe.
15:56.26Hmmhesaysusing JerJer's chan_h323?
15:56.30Uther_PDarthClue: my point exactly
15:56.35IPDyes chan_h323
15:56.44Hmmhesayscheck the addy it binds to
15:56.47*** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk)
15:56.54IPDas recommended by Jeremy McNamara..
15:56.57Hmmhesaysin h323.conf, be default it is set to something like 1.2.3.4
15:57.20Hmmhesaysand it crashes faster than continental dc-10's
15:57.21DarthClueUther_P: but I did spend a week with Katty and I'm still living...
15:57.32Uther_Pcongradulations
15:57.37Uther_Pyou're ready for the badgers now
15:57.38archimedescrap now that asterisk compiled with out softfax I'm sure, the add-ons won't... [cdr_addon_mysql.so]
15:57.40SkramXhaha
15:58.01Uther_P"D
15:58.02Uther_P:D
15:58.08SkramXis there way to do faxing with asterisk, with just a virtual phone line.. and no digium cards?
15:58.23Hmmhesaysspandsp
15:58.26Hmmhesayshylafax
15:58.32KattyDarthClue: :>
15:58.34*** join/#asterisk greg_work (n=greg@d221-73-237.commercial.cgocable.net)
15:58.34Hmmhesaystwo tin cans and a string
15:58.50IPDhmmhesays: if defaul tis 1.2.3.4. then I need to map is to the * ip correct?
15:58.56Hmmhesaysor just 0.0.0.0
15:59.00Uther_Phey!
15:59.06Hmmhesaysthat should bind to all addresses
15:59.08Kattymine :>
15:59.11Uther_Pyou can't do that
15:59.20Kattywatch me
15:59.21Uther_Pwell alright... you can cuddle my badger :D
15:59.26Hmmhesaysum.
15:59.26IPDwhy is is binding? and why not be specific..
15:59.27Katty...
15:59.30Uther_Phahah
15:59.57IPDlord I hate being a noob
15:59.59archimedesyeah Spansd is what I'm working on and it's kicking my A$$
16:00.04HmmhesaysIPD: 0.0.0.0 will listen on all network interfaces if IIRC
16:00.07archimedesditto on that IPD
16:00.19HmmhesaysIE: bind
16:00.27JerJeri think there might be an issue with 0.0.0.0 in the version of head i'm running
16:00.32IPDhmmhesays: let me go figure out how to change that.. BRB =) and thank
16:00.35JerJer7-30-05
16:00.37erickj_azIntresting..If I use the outbound proxy as the address that work the device registers
16:00.47HmmhesaysJerJer: honestly why did you leave 1234 in the config file?
16:00.59JerJereh?
16:00.59Hmmhesaysit took me like 3 minutes to figure that out
16:01.13Hmmhesays3 minutes I'll NEVER HAVE BACK
16:01.29erickj_azJerJer:  I'm having the same issue.
16:01.32DarthClueHmmhesays: you spend more than that on your hair i think.
16:01.40Hmmhesays37 seconds on my hair
16:01.42cpatryMOUAHAHAHA
16:01.46Hmmhesaysno more
16:02.02erickj_azOnly one IP is allowing SIP registrations
16:02.02muthey.. anyone ever used an adtran total access 850? I'm looking to go from a * -> TE110P -> t1 -> Total Access 850 -> 24 fxs ports
16:02.11mutand then people will just have regular analog phones to use
16:02.12JerJermut: sure
16:02.25mutthat will work like i think it will, correct?
16:02.29HmmhesaysJerJer: bindaddr was set to something like 1.2.3.4 in the sample and it failed to load because of it
16:02.40JerJererickj_az: i've seen that with IAX registrations
16:02.48mutthen once i hit * i can send the call however i want, another pri or voip or whatever
16:03.21erickj_azJerJer do you think it's just the CVS-HEAD or somthing else?
16:03.22hhrpi dont need a gatekeeper tho
16:03.41mutJerJer: will that work how i think it will?
16:03.45hhrpso i dont need to do anything extra h323 to the one thats already in *
16:04.08Hmmhesayschan_h323 requires openh323 and pwlib per the documentation
16:04.15brookshiremut: i have :)
16:04.23Hmmhesaysooh323 is a seperate h323 stack
16:04.25*** join/#asterisk robotalk (n=joefoo@63-110-40-82.client.cypresscom.net)
16:04.44IPDhmmhesays: I have the following chan_h323.c
16:04.48archimedeswell I'm going to post this here as well on asterisk-biz list, I'm looking for a crash course in asterisk and am willing to pay for it.
16:04.48IPDnot chan_h323
16:04.53IPDis the .c the same?
16:05.02erickj_azIt is supposed to be able to use more than one IP address, right?
16:05.23*** join/#asterisk outtolunc (i=outtolun@adsl-66-218-53-170.dslextreme.com)
16:05.28HmmhesaysIPD: not sure what you are asking
16:05.32IPDor am I looking in the wrong place.. under the asterisk/channels directory
16:06.12Hmmhesaysjerjer's chan_h323 is in there, it requires openh323 and pwlib libraries
16:06.24IPDhmmhesays: you wanted me to change the chan-h323 default ip from 1.2.3.4 to 0.0.0.0 - well I can't find chan_h323... but I can find chan_h323.c
16:06.35Hmmhesaysoh...
16:06.45Hmmhesays"/etc/asterisk/h323.conf"
16:06.58Hmmhesaysi'm getting convo's mixed up here, normally there aren't so many people talking about h323
16:07.03IPDthanks.
16:07.08IPDlol
16:07.20IPDya.. I got MGCP working.. just h323 is a pain so far
16:07.27archimedesok, can soebody tell me what's going on here? http://pastebin.ca/19721
16:07.29IPDall the dependancies etc..
16:07.40IPDand with me being a newbie... =)
16:08.01Hmmhesayswe hate newbies in here
16:08.15hhrpi dont even have h323.conf
16:08.29erickj_azin sip.conf can I have * bind to several different specific addresses?
16:08.30*** join/#asterisk SkramX (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
16:08.36Hmmhesayserickj_az: no
16:08.59Hmmhesays0.0.0.0 will bind to all network interfaces
16:09.12IPDhmmhesays: Well Im a newbie.. guess I need to bale?  I could bring 10 years voice switching to the table if you let me stay =)
16:09.43IPDss7 routing.. pri signaling.. q921 a931 etc
16:09.46*** join/#asterisk Romik (n=romik_@212.143.5.146)
16:09.54IPDWOOHOO!! IM in!
16:10.04IPDill pick up the tab.......
16:10.09JerJerIPD:  then asterisk is going to rock your world
16:10.14Hmmhesayscool
16:10.19IPDya I love the * so far
16:10.35HmmhesaysCraig Southeren really reminded me what it was like to be new, and how the opensource community should treat them
16:10.37JerJertry chan_woomera
16:10.39IPDonce I have it up Ill use h323 to talk from MS to the UK
16:10.46Hmmhesaysprops to him for that
16:10.49DarthCluearchimedes: do you have zlib installed?
16:10.59cpatryHmmhesays: heheh
16:11.06archimedeshmmm good question... hang on
16:11.06mutw00t $1000
16:11.23Hmmhesayshey cpatry: great pics man i looked at them all
16:11.33cpatrythanks.
16:11.38IPDi think I see the issue.. I dont have chan_h323 in the etc/asterisk
16:11.41archimedesyup I do
16:11.42IPDlet me get working
16:11.47cpatryi can send u the zip if u want.
16:11.55cpatrythe WHOLE zip
16:11.56HmmhesaysIPD: h323.conf not chan_h323
16:11.59cpatrywhich is over 1G
16:12.02Hmmhesayswow
16:12.17Hmmhesayshrm, maybe I will setup an ftp later today
16:12.21Hmmhesaysand we can transfer it
16:12.22*** join/#asterisk supaigtr (n=yurplsl@152.53.16.10)
16:12.36cpatrywhat about a dcc send?
16:12.38archimedesDarthClue: Yeah and it's current
16:12.49Hmmhesayshrm, sure
16:12.52Hmmhesaysthat would work
16:12.59Hmmhesaysmy speed here is bad though it would take all day
16:13.02Hmmhesayslitarlly
16:13.04Hmmhesays*literally
16:13.50cpatrybkw_: here fucker!
16:13.53cpatry:^
16:13.54cpatry:P
16:14.00DarthCluearchimedes: libxml2?
16:14.43archimedesDarthClue: check!
16:14.43Kattymmm, taco pie.
16:14.47brettnemhey anyone know why I'd get a "Failed to load shelf-router image" for a TNT? is there something special I need to do to get it to load a firmware?
16:14.52Hmmhesaysomg I want TACO BELL NOW
16:15.06*** join/#asterisk jimmybob46 (n=jim@81.5.154.235)
16:15.08Hmmhesaysnacho cheese gordita of heart plugging goodness
16:15.11DarthCluearchimedes: you need to verify the zlib stuff, -lz is zlib and if it says it can't find it then that is the issue.
16:15.19KattyHmmhesays: heart attack to go?
16:15.27KattyHmmhesays: would you like a drink with that?
16:15.28Uther_Penjoy that taco pie, because in the future they will be made illegal!... and I think we all know why
16:15.32*** join/#asterisk Innuendo (n=bk34@u19-70.dsl.vianetworks.de)
16:15.40Innuendohi@all
16:15.46archimedeshrm.... ok will do..
16:15.48KattyUther_P: because it might be a TERRORIST
16:15.51Uther_Panti-taco legistation!
16:16.03Uther_Pdisestablishmentarianism
16:16.08KattyDarthClue: :<
16:16.21*** join/#asterisk Moc (n=mochouin@h66-201-214-109.gtconnect.net)
16:16.21Uther_P(ATHF quote)
16:16.30Kattymister Moc (=
16:16.40jimmybob46right, who can sort this one out then, incoming capi call > auto-attendant (choices) dial 6 (a capi out) doesnt work :(
16:16.47MocHail Katty
16:16.49Uther_Pthats it dammit!... can anyone recomment a good server-side spam blocker?
16:17.12Jason357i just installed asterisk@home and I am getting this error: "Determing IP information for eth0... failed: no link present.  Check cable?"
16:17.14hhrpcant find a good manual on how to install channels
16:17.15hhrphttp://www.oinko.net/astrecipes/index.php?q=astrecipes/compiling+asterisk+with+oh323
16:17.17Jason357it is plugged in....
16:17.19hhrpis this a good on?
16:17.28HmmhesaysKatty: yes, a jumbo coke please
16:17.37KattyHmmhesays: ;)
16:17.38IPDhmmhesays: since I dont have h323.conf in the etc/asterisk area I must have comlied h323 incorrectly - once I issue a make in the asterisk/channels/h.323 directory I have to go back to the top level  directory and issue a make install correct?  what is the top level directory <bowing head in shame>
16:17.41Uther_PDarthClue: you may get arrested for that
16:17.48KattyDarthClue: :<<<
16:18.00Jason357anyone have an idea for me on what could be wrong?
16:18.04jimmybob46server side spam blocker? u need spamassassin
16:18.08DarthClueUther_P: nah.  It's called sit down, shut up, or walk home.
16:18.13Uther_Phaha
16:18.16HmmhesaysIPD: you need to make install in h323 directory... which I don't recall the path for atm
16:18.29DarthClueIt is also a really easy job that pretty much anyone can do.
16:18.38Uther_Pjimmybob46: yea, I used to use spamassasin.. just wondered if there was another one... it didn't quite get it all
16:18.50DarthClueRequires very little thinking and means I can actually work on * more.
16:18.53HmmhesaysI hunger from within
16:18.53Innuendocan someone help me please - i have some trouble with an isdn card error[3311] chan_zap.c:6484 mkintf: unable to specify channel 1: no such device or address
16:19.04jimmybob46spam assassin is the one. Its just a case of configuring it right
16:19.05*** join/#asterisk ManxPower (n=eric@dsl-209-16-67-160.datasync.com)
16:19.05KattyHmmhesays: as opposed to what?
16:19.09archimedesDarthClue: if i can call lz from the CLI wouldn't that indicate that it's installed right?
16:19.12KattyHmmhesays: hungering from ear?
16:19.26DarthCluearchimedes: got zlib-devel as well?
16:19.38*** join/#asterisk Lee__ (n=lee@ool-44c26fa3.dyn.optonline.net)
16:19.55HmmhesaysKatty: chuckle
16:20.00Hmmhesayser...
16:20.01Hmmhesays<chuckle>
16:20.04DarthCluearchimedes: it should, but we all know that nothing is guaranteed.
16:20.07jimmybob46anyone have any ideas on capi handling 1 incoming and re-routing to a capi outgoing?
16:20.12archimedesDarthCLue affirmative added it whilst doing the initial install
16:20.18archimedes:)
16:20.35DarthCluearchimedes: what distro you using?
16:20.41archimedesRH9
16:20.47archimedesy
16:20.58archimedeswoops sorry wrong window
16:21.12archimedeswait it say's it's not install.. brb
16:22.20Hmmhesaysargh!
16:22.25archimedesson of a... I specified it to install and even updated it with apt-get but when I check it with yum it said it was missing...
16:22.50crash3myour using apt-get and yum?
16:23.09Lee__are there any devs here? I'm having trouble figuring out some of the functions of files in the CVS tree.
16:23.11erickj_azSeems CVS is back up and running
16:23.12archimedesalright gotta ask, whats the deal with Hmmhesays's hair?
16:23.20Kattyarchimedes: it's nummy
16:23.21*** join/#asterisk TeLLuS (n=johan@81-226-227-27-no58.business.telia.com)
16:23.22archimedescrash: yeah why?
16:23.25ManxPowerarchimedes: I think it joined a union.
16:23.25DarthCluearchimedes: let's get you a photo....
16:23.47archimedesoh boy
16:24.01Hmmhesaysi got this one
16:24.19erickj_azIs there a channels where all the CVS-HEAD people hang out?
16:24.26Hmmhesayshttp://www.grumpy-monkey.com/users/3853/huh.jpg
16:24.32cpatryerickj_az: #asterisk-dev
16:24.55archimedesoh my that is a weapon...
16:25.00ManxPowerLooks more like his hair started taking meth
16:25.09Lee__cdr_addon_mysql.so requires both the Asterisk includes in /usr/include/asterisk and the asterisk CVS directory. AFAICT the only file cdr_addon_mysql.c is looking for is asterisk.h, yet when I put it in the source directory, the compile fails.
16:25.13ManxPowerHe's kind of cute, however.
16:25.18KattyHmmhesays: it needs to be another inch or two longer
16:25.34HmmhesaysKatty: i had it cut about 2 weeks before cluecon
16:25.40Katty:<
16:25.52Hmmhesaysthats why it was short there
16:25.58Kattyyeah, i took 6 inches off mine too
16:26.02anthmwe never got to play pool
16:26.12*** join/#asterisk jaike (n=a@203.177.36.132)
16:26.13brettnemHey anyone know why my MAX TNT boots up to a BOOT> prompt?
16:26.28ManxPowerbrettnem: I thnik that means "I don't have any firmware"
16:26.35Hmmhesaysberttnem: i bet you don't have any firmware
16:26.39brettnemManxPower: crap
16:26.43Kattyanthm: i played pool, even though i suck at it
16:26.48ManxPowerbrettnem: I COULD be wrong.
16:26.52jaikedoes asterisk have predictive dialer?
16:26.58anthmrotfl
16:27.03brettnemManxPower: doing an ls gives me one file: VIKINGFL.ASH
16:27.14ManxPowerbrettnem: best to consult a TNT expert.
16:27.23brettnemthat's you, right? :P
16:27.45brettnem<sigh> I wonder if they sold me this thing without firmware.. what the heck.
16:27.50frenzy14anyone with know where I can get international DIDs ?
16:27.56anthmI believe it has a predictive seg faulter.
16:28.06ManxPowerKatty: "If you wish to be taken off of hold press 31415 right NOW!.  Sorry, too slow.  Try again."
16:28.12jaikeanthm: seg faulter?
16:28.19KattyManxPower: :<
16:28.39puzzledbrettnem: possibly cause it requires a license from the vendor or authorized partner
16:28.48Hmmhesaysthere are various people that have done the auto dialer thing with asterisk
16:28.53erickj_azSAme issue with the latest build of CVS.
16:28.55frenzy14 ?
16:29.00brettnempuzzled: I'm pretty sure we bought it with software..
16:29.00mutusually when ya get it on ebay ppl don't care
16:29.09erickj_azI can't get two IP interfaces to have SIP clients register to them.
16:29.10DarthClue"To reach an operator, please enter your date of birth...I'm sorry, you aren't quite old enough to interact with just yet, please continue to hold."
16:29.20Kattybad! no frito for you!
16:29.47brettnemhow about asking for square roots.. play the jepordy theme music.. give them 10 seconds to answer. Please use the * key as a decimal
16:29.56*** join/#asterisk azrishahril (n=azrishah@60.50.200.227)
16:30.10Hmmhesaysi think ECHO is a great game
16:30.22Hmmhesaysyou say something and some stranger repeats it back
16:30.23brettnemI think the jepordy music would be great on hold music
16:30.37Hmmhesayswierd.
16:30.40brettnemyeah, ECHO would be better if you resampled it and added about1 second of delay
16:30.44DarthClue"Question 1: What hotel will never see another VOIP related conference, ever?"
16:30.57Hmmhesays"I AM SKELETOR"
16:31.19ManxPowerDarthClue: I think the answer is "Any hotel that has ever hosted a VoIP conference"
16:31.48Hmmhesaysbrettnem: app_wrecho
16:31.57Hmmhesaysi'm tempted
16:32.00DarthClueManxPower: in this case, the correct answer is "Best Western, Elk Grove Village."
16:32.20Hmmhesayshey DarthClue: you know what day it is right?
16:32.44ManxPowerI've never heard of a hotel network that doesn't fold like a house of cards when they have a conference full of geeks.
16:32.49hardwirehmmphm
16:33.09hardwireI don't know if I want to teach everybody how to become a queue member.
16:33.21hardwirebut I think it would be pretty valuable
16:33.27hardwirefor the oporator queues
16:33.31hardwireoperator even
16:33.50archimedesKatty: Hmm my GF alway's has that problem too.
16:33.51Katty:<
16:33.54DarthClueHmmhesays: it's amateur night!
16:33.55IPDhmmhesays:  I found h323_conf in the channel/h323 dir. and the addy is 0.0.0.0 and Im still can start * - I get the message Loading module chan_h323.so failed!
16:33.55IPDOuch ... error while writing audio data: : Broken pipe
16:34.08IPDany mo ideas...
16:34.12HmmhesaysIPD, did you copy it to etc/asterisk/oh323.conf ?
16:34.24Hmmhesaysdid you compile openh323 and pwlib properly?
16:34.33IPDno and yes yes
16:34.43ManxPowerError while writing audio data is an MPG123 problem.
16:34.45IPDlet me copy it
16:34.49ManxPowerYou need EXACTLY mpg123 0.59r
16:34.56HmmhesaysDarthClue: open till 3
16:35.17brettnemyou can make mpg123!
16:35.20HmmhesaysI could leave right now and make amateur night
16:35.20KattyDarthClue: two key factors i'm missing
16:35.20hardwireheh
16:35.22hwthm, how do i allow SIP devices to call each other? i my config they are in the same context, can call asterisk-internals, but can't call each other. i get a "3 - no route to host".
16:35.26KattyDarthClue: the body, the confidence
16:35.33KattyDarthClue: though i think i have The Gear for it
16:35.35hardwirehwt: are the hosts behind nat?
16:35.57hwthardwire: no. although they are on a different network than the pbx.
16:36.27hardwirehwt: sip show peers
16:36.35hardwiremake sure they are registered to IP's that are routable
16:36.43KattyHmmhesays: i'm officially addicted to that cd.
16:36.51HmmhesaysKatty: isn't it a strange mix of music?
16:36.55KattyHmmhesays: tis.
16:37.27KattyHmmhesays: rip more pls kthx
16:37.31Kattyoh.
16:37.33Katty:<
16:37.34*** join/#asterisk Hmmhesays (i=negative@66.173.103.108)
16:37.36Hmmhesaysdamn mouse
16:37.37Katty:>
16:37.41hwthardwire: 0 sip peers [0 online , 0 offline]
16:37.42Katty11:39 < Katty> Hmmhesays: rip more pls kthx
16:37.42Katty11:39 < Katty> oh.
16:38.03*** join/#asterisk myiagy (n=myiagy@200.175.61.149.tbprof.gvt.net.br)
16:38.13hwthardwire: they can be pinged from the pbx.
16:38.14Hmmhesaysgot it
16:38.18frenzy14Anyone know where to get DIDs from ?
16:38.29KattyHmmhesays: i don't, sorry.
16:38.32Hmmhesayssure if you turn your text black
16:38.40KattyHmmhesays: that's why you should rip it (=
16:38.51HmmhesaysKatty: rip more music?
16:38.58KattyHmmhesays: yes, please.
16:39.04sivanauniqueid CDR variable is based on timestamp?
16:39.06Hmmhesayswhat do you want?
16:39.14Kattywhat do you have?
16:39.18Kattyjust post all of it
16:39.22Kattyand i'll wget the whole directory
16:39.22Hmmhesaysheh
16:39.27Hmmhesaysit's 15 gig
16:39.27Kattyor whatever i want ;)
16:39.30Kattyk
16:39.39*** part/#asterisk frenzy (n=frenzy@193.220.82.108)
16:39.41Kattygive me a list or something generated from winamp
16:39.45*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
16:39.53Hmmhesaysk
16:39.59Hmmhesayslater, possibly tomorrow
16:40.11*** join/#asterisk smurfix (n=smurf@debian/developer/smurf)
16:40.14frenzy15Hmmhesays:02 black enough ? :)
16:40.24Hmmhesaysgeebus man what client are you using?
16:40.38puzzledprolly a windows client
16:40.38frenzy02trillian
16:40.49Hmmhesayshey, mirc works fine for me
16:41.01Hmmhesayssince I am stuck at this windows machine
16:41.20frenzy02?
16:41.23Hmmhesaysfrenzy you can get did's from telephone service providers
16:41.30Kattyi used mirc for a long time
16:41.36Kattyuntil i discovered screen and irssi anyway.
16:41.52Kattynow i'm hopeless addict 24/7 :>
16:42.04frenzy02:P
16:42.17HmmhesaysI have no problems with mirc it does everything i need it to
16:42.31ManxPowermIRC runs on Linux?
16:42.35puzzledany bristuff/zaphfc users around that know why the D-channel keeps going on/off/on/off etc.?
16:42.36Kattyyeah, i originally didn't switch to linux because i wanted to do database stuff or email servers or firewalls
16:42.45Kattyi switched for screen and irssi ;)
16:42.45Hmmhesaysi bet it does under wine
16:42.47hwthardwire: in the extensions table (i'm using realtime), each extensions looks like this:
16:42.53HmmhesaysI use xchat in linux
16:42.53hwthardwire: 1  default  1001  1  Dial  SIP/1001@default,60
16:43.06hardwirejust do SIP/1001
16:43.13hardwireno @default,60
16:43.14Kattyxchat isn't screenable :<
16:43.16hwthardwire: a switch is configured in extensions.conf: [default]
16:43.16hwtswitch => Realtime/default@realtime_ext
16:43.22hardwireKatty: yes it is :)
16:43.25*** join/#asterisk SkramX (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
16:43.39hardwireKatty: I use a slightly modified version of Dircproxy.
16:43.43*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
16:43.45Hmmhesaysif I don't irc on machines without a WM
16:43.55hwthardwire: and the sip_buddy is pretty straight forward, with a default context.
16:44.09hardwirehwt: you may be out of my realm of kno how who haw
16:44.19hardwireI don't use switch ever..
16:44.26hardwirenor ast realtime
16:44.41hwthardwire: so i don't need a switch?
16:44.50rikstasipgate suck, i tried to get some calling credit, and they don't even want to take my money
16:44.54hardwirehwt: I can't really help :)
16:44.55rikstafools
16:45.02hwthardwire: allright. :)
16:45.03*** join/#asterisk pbxbart (i=user@p54B02FDE.dip0.t-ipconnect.de)
16:45.53IPDhmmhesays: okay I give up.. Im rebuilding linux and starting over
16:46.04Hmmhesayswhat is your issue now?
16:46.25*** join/#asterisk JakBeatZ (n=JakBeatZ@216.7.194.254)
16:46.25azrishahrilanyone can assist me on licensing scheme for digium g729 ?
16:46.37IPDim to newb to get it to work.. i think I screwed up and issues make installs in areas I shoudn't
16:46.43JakBeatZAnyone had problems with festival 1.4.3 where it seems to nicely hog CPU?
16:46.48HmmhesaysIPD could be
16:46.50azrishahrillet say initially i bought for 10 channels
16:46.52IPDlol
16:46.54Hmmhesaysfollow the readme's
16:47.02Hmmhesaysit will work
16:47.03archimedescrap left the coke in freezer a weeeee bit to long... won't come out now...
16:47.06hwthow is it possible to *not* use switch anyway?
16:47.09IPDIll let you know if I get it rolling
16:47.18azrishahrillater i would like to add another 10 g729 channel ...
16:47.21IPDI can't read the readme when it's tar'ed
16:47.24*** part/#asterisk pbxbart (i=user@p54B02FDE.dip0.t-ipconnect.de)
16:47.24Hmmhesaysyou don't make install openh323 or pwlib
16:47.53IPDi don't use the gui so my life is alittle hard than it needs ot be
16:48.11Hmmhesaysthat shouldn't be a problem
16:48.16IPDill get it.. buy you guys a beer when im finished...
16:48.41IPDya the problewm is my newbness with Linux
16:48.52Hmmhesaysi'll bring the pizza, pepperoni, jalapenos
16:49.09*** join/#asterisk angom (n=Administ@red-corp-200.38.16.10.telnor.net)
16:49.15IPDto frustrated now.. have it working then screwed it up when trying to complie h323
16:49.22IPDill bring the jalapenos
16:49.22HmmhesaysIPD: have you discovered that you can open seperate consoles with the F keys?
16:49.23*** join/#asterisk ipso (n=ipso@d207-81-249-35.bchsia.telus.net)
16:49.30IPDnow Im hungry!!!!!!!
16:49.43Hmmhesays;) same here
16:49.57jimmybob46what is wrong here ? :S Executing Dial("CAPI/contr1/236-27", "CAPI/g1/b209|20|r") in new stack
16:50.03azrishahrilG.729 licensing scheme assistance , please
16:50.10IPDl8r hesay
16:50.14jimmybob46i cant work it out
16:50.21Hmmhesaysazrishahril: what are you talking about man
16:50.24HmmhesaysIPD: later
16:50.56*** join/#asterisk brettnem (n=brettnem@72.29.102.158)
16:51.01*** join/#asterisk rephorm (n=brian@67.95.13.46)
16:51.43azrishahrili would like to know is there any issues if initially i buy 10 G.729  then later 1 buy another 10 G.729 ...
16:52.16Hmmhesaysit used to be a pain, but I haven't bought any g.729 licenses recently
16:52.23azrishahrilam i getting 20 G.729 after 2nd purchase
16:52.46Hmmhesaysazrishahril: you could always ask the provider of the licenses
16:52.52brettnemok, thunderbird is starting to suck
16:53.02*** join/#asterisk sfbosch (n=sfbosch@dsl-cap-66-18-218-36-cgy.nucleus.com)
16:53.11Hmmhesaysi'm still using some beta version
16:53.51*** join/#asterisk SplasPood (n=jwb@dementia.paravolve.net)
16:54.01sfboschHello everyone
16:54.18HmmhesaysHello
16:54.19*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
16:54.52sfboschI'm very new to Asterisk and I am still exploring if it will be suitable for my project -- so I have a question
16:55.53Hmmhesayschan_telepath isn't working yet, so you'll have to type it out
16:55.56Hmmhesays;)
16:56.02sfboschI want to be able to take a conference call and put the audio stream from it into a FIFO in real time
16:56.04archimedeshehe
16:56.19sfbosch(sorry, it's pretty laggy over here)
16:56.24*** join/#asterisk PakiPenguin (i=uppal@unaffiliated/pakipenguin)
16:56.31*** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca)
16:56.34Hmmhesayssfbosch: for what reason are you looking to do that?
16:56.41archimedesspikey haired smartass here..
16:56.49*** join/#asterisk focks (n=cbruende@nsc66.147.95-93.newsouth.net)
16:56.59sivanaanyone know how the uniqueid CDR field is generated?
16:57.07Hmmhesays:D
16:57.11brettnemsfbosch: the ICE application did something like that.
16:57.13sfboschhmmhesays: I want to be able to stream the resulting audio to another machine
16:57.29sfboschbrettnem: ICE application? I'm not familiar with it
16:57.33brettnemhmm
16:57.50brettnemit makes it like a shoutcast stream.. or something similar.
16:58.09Hmmhesaysany reason you don't want to just send the rtp to another machine?
16:58.14cpatrywhich airport should i take for astricon in anaheim?
16:58.23*** join/#asterisk jdg (n=jdg@CA03F876.adsl.mana.pf)
16:58.53sfboschhmmhesays: no particular reason -- really I just want the audio in a format that I can reprocess for streaming purposes
16:58.53brettnemsfbosch: take a look at this: http://www.voip-info.org/wiki-Asterisk+cmd+ICES
16:59.17Hmmhesaysinteresting
16:59.36*** part/#asterisk jdg (n=jdg@CA03F876.adsl.mana.pf)
16:59.52sfboschhmmhesays: keep in mind, I'm new to Asterisk, and I don't even understand how it works under the hood
16:59.57sfboschdoing this may be obvious :)
17:00.15sfboschI'm in the research phase -- I want to be sure I will be able to do this before I start putting a lot of time and energy into it
17:00.24*** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net)
17:00.29SpaceBasshowdy
17:00.30*** join/#asterisk RCAM (n=rcammobi@232-35.26-24.tampabay.res.rr.com)
17:00.38Hmmhesayssfbosch: no worries, someone always knows more than you
17:00.56sfboschbrettnem: Thanks for that link -- wow, looks very interesting
17:01.11brettnemsfbosch: what you are asking is not inherently obvious.. so don't worry. .it's a "good question" ;)
17:01.20SpaceBassanyone used the priority alert (distinctive ring) feature on BV?
17:01.34SpaceBasscan * recognize distinctive ring over sip/
17:01.49HmmhesaysIIRC you can set alert_info
17:02.06*** join/#asterisk bofh42 (n=bofh42@p5482B72F.dip0.t-ipconnect.de)
17:02.19Hmmhesaysbut I think that just changes ring cadence
17:02.41*** join/#asterisk [bofh42] (n=bofh42@p5482B72F.dip0.t-ipconnect.de)
17:02.51ManxPowerSpaceBass: It's set using the Alert-Info: SIP header, since Asterisk is not a SIP proxy it may not pass that header on to the destination device.  You'll have to manually set it using ALERT_INFO
17:02.55sfboschhmmmhesays: So Asterisk generates an RTP stream of call/conference audio?
17:03.17Hmmhesaysgenerally that is how it is done in sip
17:03.30erickj_azOK so...I have a sip client (X-Ten eyebeam) that send out a register on one IP.  Asterisk sends the resoponse on a different IP.  What am I doing wrong?
17:03.30*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
17:03.47Hmmhesayserickj_az: sounds like you are natted
17:03.47SpaceBassManxPower:  thanks, I'll look into alert-info  Im mostly interested in routing it with in * not sending d ring to a phone
17:03.55shadeboboosomeone can help me with app_milliwatt.c?
17:04.08file[laptop]hot'n'sexy Hmmhesays!
17:04.21Hmmhesayshey hey file[laptop] , how goes it?
17:04.28file[laptop]not bad
17:04.42Hmmhesaysgood good
17:04.44SkramXanyone ever used http://areski.net/Web-MeetMe/about.php ?
17:04.47file[laptop]warped my head around my own implementation of a linked list
17:05.17Hmmhesaysmama says if you make that face for to long it'll stay that way
17:05.28*** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com)
17:05.33focksanyone using the 7914 as a busy lamp field? i'm wondering how to put SIP extension on the BLF buttons to show status
17:06.01erickj_azSo in sip.conf I should have nat=no?
17:06.26sfboschbrettnem: So, have you done this before yourself?
17:06.38Hmmhesaysif asterisk is on a public IP and the endpoint you are trying to register is behind nat, then you should have nat=yes
17:06.44*** join/#asterisk Romeo (i=MiXeR@194.165.133.210)
17:06.46brettnemsfbosch: nah, thought about it.
17:07.52nussewhat can i do with chan_capi what is not possible with hisax?
17:07.53Hmmhesayshaving a good imagination combined with irc is so much fun
17:08.08HmmhesaysDarthClue: it is a very strange mix of music
17:08.08brettnemIt's a good time waster
17:08.11erickj_azOK so if asterisks think I'm natted and I am not where do I look?
17:08.17brettnemalmost as good a time waster as Google Earth
17:08.24*** join/#asterisk razu (n=razu@193.40.81.52)
17:08.25fockserickj_az, sip.conf?
17:08.33brettnemerickj_az: externip in sip.conf maybe?
17:08.46Hmmhesayserickj_az: if asterisk and your endpoint on on the same network, or on public ip's then you don't have to worry about it
17:09.02ManxPowerDoes anyone have a good source for Amphenol cables and 66-block punchdown with amphenol connector?
17:09.16focksManxPower, gocore.com
17:09.17brettnemgreybar
17:09.21HmmhesaysI have about 10 boxes of 66 blocks
17:09.25Hmmhesayssitting around doing nothing
17:09.26ManxPowerbrettnem: greybar's site SUCKS
17:09.33focksblackbox
17:09.33brettnemgoto greybar..
17:09.45brettnemyou mean.. you don't make your own??
17:09.53ManxPowerItems: 16-30 of 608 Items in category Cross Connect Hardware
17:10.00erickj_azShould I add all my ip addresses to externip?
17:10.04ManxPowerthey don't even have pictures
17:10.22Hmmhesayshow many you need ManxPower?
17:10.26archimedeswhen starting *, I get a "Asterisk died with code 1.
17:10.53archimedesanybody know what the heck that means... no log entry into  /var/log/asterisk/full either
17:11.03nighty-Hi ManxPower :)
17:11.11brettnemManxPower: just go to greybar.. they are all over the place.
17:11.12ManxPowerHmmhesays: 4 would be a good start, but the 66-blocks need an amphenol on them
17:11.26brettnemManxPower: I believe that's called an RJ-21X
17:11.38ManxPowerbrettnem: I'm calling them later, since I cannot determine from their site what I need to order.
17:11.44ManxPowerbrettnem: yes, that's correct.
17:11.52HmmhesaysI have about 50 or 60 66 blocks with amphenol around somewhere, but no cables
17:11.54brettnemfroogle it!
17:12.11brettnemjbot: can you froogle?
17:12.20Hmmhesays~dance
17:12.20jbot"ah, ah, ah, ah, stayin' alive, stayin' alive"
17:12.29ManxPowersearcginf for RJ-21X brings up lots and lots of hubs and switches on gretbar's sire.
17:12.32shadeboboosomeone can help me with app_milliwatt.c?
17:12.44ManxPowershadeboboo: perhaps if you specificed WHAT help you needed.....
17:13.09brettnemhe wants it sampled at full rate instead of 8Khz
17:13.18shadebobooI want to known what is "static char digital_milliwatt[] = {0x1e,0x0b,0x0b,0x1e,0x9e,0x8b,0x8b,0x9e} ;"
17:13.31brettnemeww.. looks like asm
17:13.33sfboschhmmnhesays: So -- question for you, while you're dancing -- am I to understand that Asterisk handles all its channels as RTP streams?
17:13.54shadeboboobrettnem : goodnews. I have tdm with 4fxs port. Chipset is si3215 wich can resample at 16khz
17:13.59Hmmhesayssfbosch: no
17:14.12brettnemshadeboboo: do the community a favor and post that to the wiki
17:14.23ManxPowershadeboboo: you'll need to ask on the mailing list
17:14.32brettnemshadeboboo: did you find that stuff in zconfig.h?
17:14.33ManxPoweror maybe #asterisk-dev
17:14.45brettnemManxPower:here's 4 for cheap: http://www.davesbargainoutlet.com/ViewProduct.asp?ModelNumber=RJ21X
17:14.52shadeboboobrettnem : with the history of zaptel.h
17:15.03erickj_azOK sho what should I try next?  I can ping both interfaces I can SSH to both interfaces, just asterisk responds on the wrong interface
17:15.06hardwirehttp://pastebin.ca/19728
17:15.07hardwireheheh
17:15.08hardwirethat works well
17:15.23Hmmhesayssfbosch: the wiki would be a better source of info for that than me
17:15.24sfboschhmmhesays: Is there a document I should read that will help me understand how channel audio is processed?
17:15.33sfboschokay
17:15.40ManxPowerbrettnem: Avaya wired or telco wired?
17:15.49brettnemek
17:16.04brettnemI dont' know. you mean is it 1 for 1? or custom?
17:16.16SpaceBasserickj_az:  still cannot get both nics to work?
17:16.22Kattybuh bye
17:16.30shadeboboobrettnem : so maybe in a few time I can send a patch to generate a 12khz signal ;)
17:16.34SpaceBasserickj_az:  have bindaddr = set in your sip.conf?
17:16.44erickj_azOK it just started working for no apparent reason.
17:16.54SpaceBasscant argue with that
17:17.03brettnemshadeboboo: really, please, post to the wiki.. if you do, someone else might find it of interest and make it better.
17:17.31rephormerickj_az: i had a similar problem with iax and multiple interfaces. you should set bindaddr in sip.conf to the interface you want to use
17:17.32shadeboboobrettnem : when my solution will work :s
17:17.46brettnemI'd start a wiki page already with some ideas..
17:18.07shadeboboobrettnem : I will made this
17:18.16shadeboboobrettnem : with my bad english ;)
17:18.37erickj_azI want to use both
17:18.40ManxPowerbrettnem: I'll just call grebar
17:18.40rephormerickj_az: even though its working now it'll probably stop working when you reboot the next time :)
17:18.43brettnemshadeboboo: no one minds broken english.. we all talk tech
17:18.58erickj_azThat waht I figure
17:19.11erickj_azSo how do I know what I did to fix it?
17:19.19*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
17:19.24jaikeanyone using a predictive dialer with asterisk in production?
17:19.40brettnemManxPower: heh, check this out: http://cgi.ebay.com/21-66-M1-50-Cat-5-Punch-Down-Blocks-good-used_W0QQitemZ5795573197QQcategoryZ28025QQrdZ1QQcmdZViewItem
17:20.07harryvvget ready for a new kind of memory to come out in 2006. 3D memory. burned on a translucent dvd like disk it can store up to 1000 movies.
17:20.09PakiPenguinthis is purely offtopic , but where can i buy the cute stuffed penguins? the small ones?
17:20.10ManxPowerbrettnem: I also need the cable
17:20.13Hmmhesaysis there anyone out there? cause its getting hard and harder to breathe
17:20.15brettnemyeah I know
17:20.29ManxPowerSo I need to match the gender on the TA 750 with the cable and the cable with the 66-block
17:20.31Hmmhesays~google cute stuffed penguins
17:20.33*** join/#asterisk pbxbart (i=user@p54B02FDE.dip0.t-ipconnect.de)
17:20.39*** part/#asterisk pbxbart (i=user@p54B02FDE.dip0.t-ipconnect.de)
17:20.39*** join/#asterisk MattB2 (n=MattB2@mail.tricycleinc.com)
17:20.39frenzy02does Iaxtel work with sip ?
17:20.47MattB2hi all
17:20.48file[laptop]did you just say that... in blue?
17:20.53frenzy02as-in the 700 #
17:21.03*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
17:21.03file[laptop]IAXtel uses IAX...
17:21.05Hmmhesayshe's been colorful all day file[laptop]
17:21.07file[laptop]thus why it's IAXtel
17:21.39hardwireooh
17:21.43hardwireI want to be colorfull
17:21.49PakiPenguini like colors!
17:21.50Hmmhesaysplease don't
17:21.51Hmmhesaysthanks
17:21.58Hmmhesaysthis is no yahoo chat
17:21.58brettnemManxPower: be sure you also get the riser that the 66 block snaps into..
17:22.25MattB2just installed a TDM400P in the UK with 3x FXO modules to 3 analong lines.  When I try to dialout I get silence and an occasional click.  If I try dialling in it just rings.  Asterisk doesn't show any attempt to answer incoming calls.  module debug doesn't show any incoming or outgoing detection, but everything shows that the card is working perfectly.  Any ideas?
17:22.44*** part/#asterisk jaike (n=a@203.177.36.132)
17:23.03*** join/#asterisk Uther_P (n=uther_p@66.180.120.89)
17:23.16MattB2using fxs_ks, wcxfs module (wctdm fails - can't find channel)
17:23.16*** join/#asterisk tld (n=tld@195.84-48-78.nextgentel.com)
17:23.28erickj_azit worked after I rebooted!!!
17:23.37archimedesAhhh this is killing me... can somebody please check this post and let me know what he or she thinks? http://pastebin.ca/19729
17:23.37Hmmhesaysthat is the fix for everything
17:24.03Hmmhesayswhat does var/log/asterisk/full say?
17:24.25archimedesnothing, empty
17:24.42*** join/#asterisk pbxbart_ (i=user@p54B00A80.dip0.t-ipconnect.de)
17:25.05MattB2arc: Aug 10 12:22:19 ERROR[1715]: Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling
17:25.22*** join/#asterisk [cc]smart (n=smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
17:25.36NetgeeksAug 10 12:22:23 ERROR[1729]: Signalling requested is FXS Kewlstart but line is in FXO Kewlstart signalling
17:25.47Netgeeksoops, sorry for the ditto
17:25.50MattB2;)
17:25.57ManxPowerbrettnem: *nod*
17:26.06MattB2arc: for FXO cards, need FXS signalling and vice versa
17:26.16sfboschWhat format is the "gsm" file?
17:26.18MattB2make sure /etc/zap* is setup correctly
17:26.22MattB2and /etc/asterisk/zap*
17:26.23Hmmhesaysgsm
17:26.29brettnemsfbosch: that'd be gsm formal
17:26.32brettnem-l+t
17:26.35Netgeeksin zaptel you have fxoks=1-2 and in zapata you haev something like fxs_ks=1-2   (I may have the correct spelling backwards on those directives"
17:26.45Hmmhesayspack it up pack it in, let me begin,
17:27.05*** part/#asterisk pbxbart_ (i=user@p54B00A80.dip0.t-ipconnect.de)
17:27.07brettnemI thought this was a "no yahoo" group??
17:27.08archimedesok.. checking.... uhhhg
17:27.08sfboschAs in, the same format used in a GSM timeslot?
17:27.12*** join/#asterisk loick (n=loick@APuteaux-151-1-42-132.w82-124.abo.wanadoo.fr)
17:27.20*** join/#asterisk shony (n=shony@nant.sumix.com)
17:27.28brettnemsfbosch: I think.. but not sure.. why?
17:27.30Hmmhesayssince when does house of pain have anything to do with yahoo
17:27.48brettnemyou're a bunch of yahooligans
17:27.54sfboschI'm trying to understand the format -- it might be suitable for feeding the FIFO
17:27.59*** join/#asterisk SkramX (n=SkramY@cpe-70-112-81-84.austin.res.rr.com)
17:28.06Hmmhesayssfbosch: no
17:28.20brettnemsfbosch: just stick with that ICEs thing.. should be easy to manipulate rather than inventing something new
17:30.04Hmmhesaysok time to go get one of those $2.00 half pound sammich
17:30.08Hmmhesayslater
17:30.22sfboschbrettnem: Well, we need more control than we would get there -- what's good about the ICEs thing is that it proves the concept
17:32.00fockshas anyone here had their (C)LEC install hardware line echo cancellers on their PRI?
17:32.10focksis that even an option?
17:32.41Hmmhesaysyou have echo on your pri?
17:33.28hhrpwhat is CVS head?
17:33.30mutwould probly be a safe assumption
17:34.19brettnemit really depends on the length of the T1 loop
17:34.31focksHmmhesays, yep
17:34.59focksHmmhesays, seems like it's when we call analog lines, we hear ourselves phase shifted about 200ms
17:35.04focksannoying as hell
17:35.49brettnemthen you don't need the echo cans
17:36.01MattB2Anyone have experience with BT Featurelines?
17:36.26*** join/#asterisk cgcorea (n=cgcorea@205.240.200.103)
17:36.27focksbrettnem, what do you mean
17:36.31*** join/#asterisk mrtwister (n=mrtwiste@tor/session/x-57ee750e4294e2fe)
17:36.37brettnemecho comes from the far side.. not the near side
17:36.50brettnemthink propogation delay
17:36.51focksbrettnem, right, but we can still deal with it on our side no?
17:37.03brettnemperhaps.. but you are tackling the wrong problem.
17:37.16focksso where should i be looking?
17:37.38focksnot much i can do about the condition of the lines i'm calling
17:37.38brettnemthe other end of the line probably needs the echo can.. heck, an echo can on your side might help.
17:37.58*** join/#asterisk grimse (n=grimse@p5481FB5F.dip.t-dialin.net)
17:38.05brettnemyeah, well if your provider is selling you a 400 mile t1, they should have cans on their line (so should you tho)
17:38.24focksbrettnem, so how can i accomplish this is my question
17:38.37brettnemtell your provider to fix their stuff.. is your PRI a long loop?
17:39.06hhrpwhat is CVS?
17:39.10frenzy02can I point FWD #s to a sip URI ?
17:39.14focksbrettnem, no idea
17:39.23brettnemI'd talk with them about it.
17:39.44focksbrettnem, i haev a ticket open, i just need to know what to ask them
17:39.45*** join/#asterisk myiagy_ (n=myiagy@200.175.61.149.tbprof.gvt.net.br)
17:40.01focksbecause they always try to snow me by saying it's CPE, contact the vendor. i am the vendor!
17:40.09brettnemI hear echo. do you have cans on the line.. is something broken
17:40.14*** join/#asterisk j357 (n=m00@cpe-65-26-177-31.indy.res.rr.com)
17:40.20brettnemlook.. it works like this..
17:40.22SkramXhahahaha
17:40.24brettnemyou talk into your phone..
17:40.32focksbrettnem, ha, ask a CLEC if something on their end is broken... of course not !
17:40.37brettnemyour voice travels down the line.. hits their stuff..
17:40.39syleso how was the chicago cluecon?
17:40.39MattB2ok
17:40.44MattB2anyone know what MF4 signalling might be?!
17:40.48brettnemand then they send your voice back to you?? why? that's broken
17:40.56SkramXSo Im curious. how much does it actually cost for places like asterlink and nufone.net and iax.cc to buy the 800 # from the CLEC/Level 3...
17:41.14brettnemit *can't* be on your side.. there has to be distance to get the propogation delay that gives you echo
17:41.38focksbrettnem, that's my understanding of it
17:41.43brettnemSkramX: numbers don't usually cost anything for carriers.. but minutes do.
17:41.55SkramXThats what I mean
17:41.58brettnemso teach them.. most CLECs are idiots..
17:42.10focksthis one sure is
17:42.10*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
17:42.15brettnem:)
17:42.22focksNuvox
17:43.05SkramXbrettnem: how much they pay for a toll free? like 1c a minute?
17:43.22brettnemSkramX: it's probably below that.
17:43.30brettnemminutes are overrated.
17:43.35SkramXYea
17:43.37brettnemit'll be free soon
17:43.39focksthey said they're seeing timing slips on the Adtran, can I verify that with the Digium card?
17:43.39wunderkinfees are overrated :D
17:43.55SkramXdoes level3 like actually run them a line or does it go thru sip?
17:44.05mutit is teh MAJOR sux
17:45.34sylei think rates are posted skramx
17:45.52syleif you buy alot of minutes generally its about .008 a minute
17:46.54*** part/#asterisk heath__ (n=root@12-215-34-84.client.mchsi.com)
17:46.57hhrppatch -p1 < /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch what do they want to do by doing this?
17:47.11SkramXsyle: where?
17:47.13brettnemSkramX: just kinda depends on the arrancement
17:47.22brettnemsyle: where do you operate? you are a CLEC, no?
17:47.24sylei don;t think level is a CLEC are they?
17:47.30sylei thought they were more of an ILEC
17:47.35brettnemno, they are a CLEC
17:47.37SkramXYea
17:47.50brettnemILEC is a carrier of last resort.. Level 3 does not fit that description
17:47.57sylei thought they owned their own fiber lines
17:48.03brettnemsure they do.
17:48.06brettnemCLECs can do that
17:48.19brettnemand you don't have to own fiber to be a fiber based company
17:48.22syle100k to become a CLEC it seems
17:48.33brettnemsyle: just depends on how you do it.
17:48.35archimedesDoes anybody have a sample zapata.conf file they can post thats using a simple TDM400P with one FXO and one FXS?
17:49.00recursionis away
17:49.00brettnemsyle: you can probably do it for less if you opt into a locally accepted state tariff
17:49.12sylemy buddy here is hooking up to the tandum switches finally , but he doesn;t have CLEC just non-dominant carrier status and that cost about 25k
17:49.29brettnemwho is charging 25k?!
17:49.34brettnemthe RBOC?
17:49.51syleCRTC
17:50.04brettnemhmm.. is that canadian?? lots of "C"s in it..
17:50.10syleyes :)
17:50.14sylesorry yes i;m canadian lol
17:50.21tzangerCanadian Regional Telephone Centre of Canada
17:50.23jontowso, how might i view a remote IAX2 trunk's dialplan from my local * ?
17:50.30jontowis it as simple as using 'switch' ?
17:50.35brettnemheh.. well.. I'm a US CLEC.. I can tell you this.. after installing 60 T1s to bell, my first bill was for $0.57
17:50.40tzangerso the CRTC's tarriffs on tandem connections are $25k
17:50.41tzanger?
17:50.50brettnemthat's rediculous
17:50.54rephormManxPower: was that you I was talking to last night about echo on polycoms?
17:51.01brettnemit's anticompetitive. did you agree to that in your ICA?
17:51.08syleafter all the legal bullshit my friend went through, took him 2.5 years
17:51.08*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
17:51.15focksrephorm, you having echo problems?
17:51.21sylehe's hooking up this month
17:51.23rephormManxPower: if so, then yes, you are correct. the issue is with speaker volume on the receiving end :)
17:51.24brettnemheh, I've been doing it for 8 years..still negotiating.. :)
17:51.39sylelol
17:51.42brettnemwe've got one of the best telecom attorney's in the country
17:51.57sylewell that definately helps speed things up :)
17:52.06brettnemheh, I'm actually reading through testimony right now..
17:52.28brettnemsyle: this is my 2nd CLEC to startup.. the first one was/is very successful
17:52.37rephormfocks: i was, but it was with echo on bridged sip calls caused by the 'other' end's speaker volume being too high and their mic picking it up and sending it back :)
17:52.40sfboschCRTC = Canadian Radio and Telecommunications Commission
17:52.46focksyeah
17:52.59syleinteresting
17:53.02sylewhat states?
17:53.08brettnemJust Texas..
17:53.12brettnemit's a political thing
17:53.15sylehaha
17:53.21brettnempolitical climate that is..
17:53.31sfboschbrettnem and hmmhesays: Thanks for all your help! See you around.
17:53.34sylei meet so many people from texas when i am down in caribean, all into realestate, very wealthy
17:53.39brettnemsfbosch: good luck
17:53.48SkramXOh man. the comercials bashing SBC and verizon are on tv all the time (by "txcable.com")
17:53.52SkramXI live in the capitol of TX
17:54.05brettnemsyle: heh, my dad has been in commerical real estate for 38+ years.. and was in the caribean about 3 months ago.. haha
17:54.11brettnemSkramX: so do I.
17:54.13sylelol
17:54.23brettnemSkramX: you are in Austin? where?
17:54.25*** join/#asterisk sskyles (i=Steve@137.204.205.68.cfl.res.rr.com)
17:54.44sylehe didn;t go to panama by chance? hehe
17:54.47*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
17:54.49brettnemheh no
17:54.55SkramXbrettnem: North West area
17:55.04brettnemah, you're up by Nugget
17:55.13brettnemI'm off 360
17:55.50sylebrettnem send me your email, maybe we can peer up to each other if your a carrier as well once this system is setup
17:56.05brettnemsyle: yeah sure.. I'll pm you
17:56.25*** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-241-108.buckeyecom.net)
17:56.51*** join/#asterisk g_adler (n=gadler@72.29.123.112)
17:57.29*** join/#asterisk programmermatt (n=programm@d3-233.rb2.lax.centurytel.net)
17:57.37*** part/#asterisk secure75 (n=mic@ppp-82-135-14-79.mnet-online.de)
17:58.01*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
17:59.20*** part/#asterisk programmermatt (n=programm@d3-233.rb2.lax.centurytel.net)
18:00.20*** part/#asterisk frenzy (n=frenzy@193.220.82.108)
18:00.50*** join/#asterisk netnameus (N=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
18:00.51*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
18:01.12netnameusis it true that a "zaptel" interface must be installed for convferencing to work?
18:01.14*** join/#asterisk sskyles_ (i=Steve@137.204.205.68.cfl.res.rr.com)
18:01.20*** join/#asterisk jaike (n=a@203.131.137.76)
18:02.57*** join/#asterisk cardenasdouglas (n=carlinho@63-130-194-42.ADSL.CandW.ky)
18:02.57cardenasdouglashi guys!
18:02.57Netgeeksyes, you need a timing source, but that could include ztdummy (which is a software based psuedo zaptel device)
18:02.57cardenasdouglasany one to help me with a issue?
18:02.57cardenasdouglasplease!!!
18:02.58*** join/#asterisk PhreeStyle (n=PhreeSty@cpe-24-221-52-165.az.sprintbbd.net)
18:02.58*** join/#asterisk dimitr37 (n=malvazi@dyn171.the1.nas.panafonet.gr)
18:03.13netnameusthanks, i'll look into that
18:03.18focksis there any way to check a T100P for timing slips?
18:03.31Uther_Pcardenasdouglas: you don't have to ask to ask
18:03.45cardenasdouglasi installed the asterisk and everything seems to work fine... but i got a problem where my sip phone do not call out
18:03.51cardenasdouglaslet me explain.
18:03.56*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
18:04.21cardenasdouglasI got one analog phone connected to a t400p card.
18:04.26cardenasdouglasand it calls out ok
18:04.43cardenasdouglasif i call from the anolog phone to the sip it works fine.
18:04.55cardenasdouglasand if i call from the sip to the analog is fine too.
18:05.31cardenasdouglaswhen i call to the outside using the pstn getaway from sip it calls the analog phone.. why is that?
18:06.02cardenasdouglascan any one tell me what i am missing?
18:06.43Netgeeksfocks: yes there is, use zttool command and slect the card you suspect is slipping, you'll see some statistics included missed interrupts
18:07.00focksNetgeeks, that won't take the channels down will it?
18:07.21Netgeeksno, not if you don't loop the circuit
18:07.31Netgeeksztcfg will bounce the channels, but don't run ztcfg
18:07.48focksspan=1,1,0,esf,b8zs should clock off the line shouldn't it?
18:07.58*** join/#asterisk zeedo (n=zeedo@www.reboot-robot.net)
18:08.04*** join/#asterisk atif_ (n=atif@mbl-82-56-78.dsl.net.pk)
18:08.49Netgeeksfocks: yes
18:09.17focksNetgeeks, hmm, it still says Internally clocked  
18:09.41*** join/#asterisk sskyles (i=Steve@137.204.205.68.cfl.res.rr.com)
18:10.12Netgeeksthats not something I've seen before, circuit is up and green, but it's reporting internally clocked?  strange
18:10.39focks<PROTECTED>
18:11.38NetgeeksI can't help you much more focks, I'm pretty tied up here working.  Call digium, you own the card, they will give support
18:11.39*** join/#asterisk Twister (i=Twister@216.30.232.108)
18:11.39*** join/#asterisk riaan (n=riaan@66.110.32.56)
18:12.43Uther_Pcardenasdouglas: pastebin your extensions.conf
18:12.53*** join/#asterisk razu (n=razu@hps713.kodunet.webs.ee)
18:13.05cardenasdouglaspaste it here?
18:13.45Netgeeks~pastebin
18:13.45jbotit has been said that pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca
18:13.53*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
18:15.02*** join/#asterisk sskyles_ (i=Steve@137.204.205.68.cfl.res.rr.com)
18:15.08cardenasdouglasok
18:15.38Uther_Pit should say "without flooding the channel and pissing people off"
18:15.41Twisterhi all, im thinking of buying a few polycom soundpoint 600's for use for *..I have a few questions. 1. I need it to be able to toggle features like call waiting and dnd on and off for that extension with a toggle button and the button needs to light up when call waiting is off and dnd is on , 2. i need it to be able to show a light on a specific line when its in use (for example, I have 4 bv lines, when 1 is in use i want a light on the line selection b
18:15.56Twistercan the polycom do this or do i need to get another phone?
18:16.05Twisteralso it needs to have a good speakerphone
18:16.19Twisterand the ability too not dial till a "send" button is pressed
18:16.57focksto clock off the line, would it be span=1,0,0 or span=1,1,0 ?
18:17.37archimedesYippee! I gots * to start... now can anybody tell me why I'm getting static and cutouts while listening to the default recorded message?
18:17.42focksTwister, there is no call waiting button
18:17.53cardenasdouglasok
18:18.07focksDND does have a button
18:19.01*** join/#asterisk peterretief (n=peterret@196.25.192.240)
18:20.29cardenasdouglasjust awaiting for my pastebin account to acctivate..
18:20.49archimedesyou don't really need an acount to just post.
18:20.53Darwin35wow
18:20.58Darwin35it all works
18:20.58cardenasdouglascan anyone tell me if the astcc works fine?
18:21.03Darwin35just a echo issue
18:21.10Uther_Pheh
18:21.18archimedeshttp://pastebin.ca/new.php
18:21.24peterretiefHello all, is there anything I need to compile in the kernel for zaptel to work
18:22.03Twisterfocks: ok, is there any way to say, when its toggled on dial *76 and when its toggled off dial *77 (for on and off)
18:22.18*** join/#asterisk _-Jon-_ (i=jon@CPE00112f6dfbee-CM00003989406c.cpe.net.cable.rogers.com)
18:22.20Darwin35ok 15 bucks to the person who can write me a call back if busy function
18:22.29_-Jon-_Hey again everyone
18:22.32MeatyHi All ! I try to implant the *67 (for confidentiality) service on my asterisk.  For it, do SetCallerIDName() SetCallerIDNum() , with no parameters. But when my call go out by my PRI, my PRI provider add my general number on the calleridnum.  Any one know what i must put in SetCallerIDNum and Name for it ?
18:22.48_-Jon-_Any ideas on how I can limit the number of calls at any given time?
18:23.11Darwin35callwaiting=1
18:23.31focksTwister, yeah you can dial a number to activate adn deactivate, but that's not phone-specific
18:23.34Darwin35if the function is still in the code
18:23.37focksie, any phone will let you do that
18:23.46ManxPowerMeaty: show application setcallingpres or something like that
18:23.58greg_workMeaty: some PRI's wont let you set CID number, and most (if not all?) won't let you set CID name.. name is looked up from some telco list based on CID number
18:23.59Meatyk thx
18:24.10ManxPower<PROTECTED>
18:24.10ManxPower<PROTECTED>
18:24.52focksTwister, i will warn you, the Polys take some work to get set up. but once you've done one, its trivial to replicate it to all of your phones. and provisioning centrally makes life nice
18:24.58Meatygreg_work ! Normaly I can choose CallerID on my PRI
18:25.29ManxPowerMeaty: you can't just set your callerid info to something invalid.
18:25.30_-Jon-_anyone know how I can limit the number of current calls?
18:25.55ManxPowerYou need to use the calling/caller presentation to make asterisk send the correct message to tell your telco to block the into
18:25.56MeatyManxPower ! Yes ! Is that ! Thx again !
18:26.01*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
18:26.02*** join/#asterisk cgcorea (n=cgcorea@63.245.14.194)
18:26.05ManxPowerThis doesn't work when calling toll free numbers, of course.
18:26.06greg_workManxPower: which is too bad, because it would be kinda fun to call friends with the callid "911"
18:26.18ManxPowergreg_work: I can do that 8-)
18:26.20Meatylol: )
18:26.26ManxPowermy telco is pretty liberal in what they allow
18:26.49greg_workthats fun ;)
18:26.57Twisterfocks: i read the polycom supports an Xhtml browser, so can i possibly write an agi script that dispplays when the phone is on or off
18:27.07Twistererr
18:27.14Twisterwhen callerid is on or off
18:27.18Twistercall waiting even
18:27.23Twisteri cant even think straight today
18:27.56ManxPowertwisted: only the polycom 600 supports the limited XML browser
18:28.05*** join/#asterisk shadeboboo (n=shadebob@ll81-144-114-192-81.ll81.iam.net.ma)
18:28.09ManxPowerthat was for twisted
18:28.13ManxPower..er..TWISTER
18:28.13cardenasdouglasguys.. i got the pastebin posted
18:28.17Twisterhehehe
18:28.17cardenasdouglashttp://pastebin.ca/19737
18:28.37ManxPowerVLANS w/DHCP are killing me.
18:29.10TwisterManxPower: ah ok...so i cant display a simple page that says Extension 200:  Call waiting: Off Do Not Disturb: On Call Forwarding: Off ?
18:29.53ManxPowerTwister: You should be able to.  The phone supports an XML Dile Display.  Check the Polycom Admin manual.
18:30.18cardenasdouglasManxPower: what is your problem with Vlans and DCHP?
18:32.11*** join/#asterisk zloc (n=zloc@zm1.tribunemedia.net)
18:33.23cardenasdouglasManxPower: if you use a switch that can support voice vlans selection you would not have a problem
18:33.31cardenasdouglaslike cisco or nortel..
18:33.49*** join/#asterisk Uther_P (n=uther_p@66.180.120.89)
18:34.13cardenasdouglascan any one check on the http://pastebin.ca/19737 site and tell me why am i not able to call out to pstn from sip phones and do for analog phones?
18:34.23ManxPowercardenasdouglas: Find me a version of CatOS that supports VoiceVlans on a 5509
18:34.28focksTwister, the phone will show DND and CFWD status
18:34.29harryvvanyone here running a spa adapter
18:34.31fockson its own
18:34.49greg_workharryvv: yes
18:34.51ManxPowercardenasdouglas: it's looking like my ISD DHCPd is not seeing the changes made by the router with regards to ip helper-address on the 2621 router.
18:35.06ManxPowerSee: http://www.cisco.com/warp/public/473/100.html
18:35.16cardenasdouglasif you use the iphelper..
18:35.36cardenasdouglasis because the default vlan is the one the phone is trying to go to.. right?
18:35.52cardenasdouglasthis means you are using the default of Vlan 1
18:36.21cardenasdouglasif you activate CDP on the switch and make the vlan you want as voice ..
18:36.39greg_workugh, why does windows suck so badly
18:37.01cardenasdouglasonce your dhcp is on that subnet it will automatically find it. so.. the ip-helper is not neccesary.
18:37.24cardenasdouglasi got a 200 phone system working like this using cisco switches.
18:37.37harryvvgreg get my message?
18:37.52*** join/#asterisk ayano (n=erik@adsl-70-245-190-90.dsl.spfdmo.swbell.net)
18:38.01ManxPowercardenasdouglas: Our DHCP server is on a different vlan/subnet
18:38.03ManxPowerWe don't really want to run three DHCP servers
18:38.25ayanotwisted:   HELP  lol
18:38.27cardenasdouglasok..
18:38.49cardenasdouglasso voice and data are using the same dhcp?
18:39.42ManxPoweryes.
18:39.53ManxPowerdifferent subnets, of course.
18:40.13cardenasdouglasups!
18:40.35cardenasdouglasusually, i use a dhcp for voice and another one for data..
18:41.00sivanaManxPower: what's our client types, insurance agents?
18:41.24cardenasdouglassorry
18:41.38sivanaManxPower: s/our/your
18:42.02cardenasdouglaswhy dont you use a different scope on your dhcp?
18:42.20cardenasdouglasone for phones and one for data..
18:42.25cardenasdouglasand use the ip-helper.
18:43.34ManxPowercardenasdouglas: That's what I'm TRYING to do.
18:43.56*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
18:44.39ManxPowercardenasdouglas: any idea why no matter what VLAN the DHCP request is coming from, the DHCP server ignores the scopes and assigns addresses out of the first scope?
18:44.44*** join/#asterisk RoyK (n=roy@158.36.187.136)
18:46.12ManxPowersivana: Windows PCs and Polycom Phones and JetDirect devices
18:46.15cardenasdouglasbecause that is the defaul scope.
18:46.33ManxPowercardenasdouglas: can you point me to some docs that can help?
18:46.45cardenasdouglasonless you got an option that is only tipical of the phone only.
18:46.57cardenasdouglaslike option 66
18:46.59cardenasdouglasor 67
18:47.16cardenasdouglasor any other parameter that it is only tipical of the phone..
18:47.28cardenasdouglasso the phone rejects the first one and takes the other one.
18:47.41*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
18:48.00ManxPowercardenasdouglas: http://pastebin.ca/19738
18:48.06greg_workManxPower: you using ISC ?
18:48.12ManxPowergreg_work: yes.
18:48.41greg_workfrom what I can tell, ISC sucks at anything besides being a normal simple dhcp server :p
18:48.49*** join/#asterisk antiPosix (n=antiPosi@159.105.102.148)
18:49.23ManxPowerCisco says "The DHCP/BootP Relay Agent will forward DHCP requests on behalf of a DHCP client to the DHCP server. The DHCP/BootP Relay Agent will append its own IP address to the source IP address of the DHCP frames going to the DHCP server. This allows the DHCP server to respond via unicast to the DHCP/BootP Relay Agent. The DHCP/BootP Relay Agent will also populate the Gateway IP address field with the IP address of the interface on wh
18:49.23ManxPowerich the DHCP message is received from the client. "
18:49.40antiPosixis there anyway to get VoIP to call a telephone(or would some service likely need to play middleman like vonage, skype, etc)?
18:49.44*** join/#asterisk Xumpi (n=SysOp@a81-84-68-51.cpe.netcabo.pt)
18:50.14Xumpigood evening
18:50.18supaigtrAnyone recommend a turnkey hardware * solution that works with quad pri and ecan?
18:50.44ManxPowercardenasdouglas: see http://www.cisco.com/warp/public/473/100.html
18:50.54*** part/#asterisk netnameus (N=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
18:50.58greg_worki was trying to do different leases for my workstations vs everything else (voip phones, etc), with basically a longer lease, and dhcp updates = on.   no matter how i configured it (as a pool, with groups, etc) it would ALWAYs give everything a 10 minute lease time
18:51.49greg_workand i couldn't find a single bit of information on that bug or anyone with similar symptoms. i decided not to bother with different leases, but i'd love to find a usable modern dhcp server :p
18:51.50*** join/#asterisk J[SS] (I=ph33r@smartserv/ceo/chaoscon)
18:52.08harryvvcardenasdouglas thats supaigtr www.digium.com look at there turnkey systems.
18:52.19*** join/#asterisk oej (n=oej@apollo.webway.se)
18:52.30*** join/#asterisk StealthMethod (n=nelsonx@adsl-070-148-141-013.sip.mia.bellsouth.net)
18:53.41supaigtrharryvv: I don't see anything other than cards.
18:53.57supaigtrI'm looking for proven hardware with the ecan cards.
18:54.26*** join/#asterisk afrosheen (n=test@txprotoa2.august.net)
18:54.42afrosheenquestion about asterisk: how difficult is it to get 2 polycom phones registered to the same extension number?
18:55.30file[laptop]that depends what you mean by extension number
18:55.35file[laptop]same entry in sip.conf you mean?
18:55.48afrosheensomething like that, so 2 phones would ring when you dialed 302 internally
18:55.59afrosheenbut it can't be a group, has to be the same vm box
18:56.05file[laptop]uh easy... basic dialplan logic
18:56.14file[laptop]Dial two things at the same time
18:56.19file[laptop]Dial(SIP/polycom1&SIP/polycom2)
18:56.24*** join/#asterisk mhnoyes_ (n=mhnoyes@user-38lc0l2.dialup.mindspring.com)
18:56.35*** join/#asterisk crich1999 (n=crich@port-212-202-201-76.dynamic.qsc.de)
18:56.52*** join/#asterisk lorien420 (n=lorien@68.156.153.235)
18:56.54afrosheenoh..so I'll have to do some hacking on my AMP config files
18:57.08*** join/#asterisk edgecase (i=jjackson@bridge.coplanar.net)
18:57.10Hmmhesays'hacking' 'amp' <chuckle>
18:57.44afrosheenwell it's tricky since you have to specify _custom files yada yada
18:58.06Hmmhesaysit would be tricky if they didn't make the custom files
18:58.15greg_workafrosheen: download the cvs version of amp, it has support for that (i think)
18:58.25edgecaseso, has anyone thought of messing around with native IPSEC support in SIP?
18:58.33Hmmhesayscustom dialplans go in custom contexts of the custom extensions.conf
18:58.34afrosheenI'm not putting amp 109 beta on my production server
18:58.35*** part/#asterisk antiPosix (n=antiPosi@159.105.102.148)
18:58.57greg_workafrosheen: welll, wait then ;)
18:58.58ManxPoweredgecase: no.
18:59.25edgecasehmm.  well it seems that you can allow/require IPSEC per socket:
18:59.26edgecase<PROTECTED>
18:59.33edgecasesetsockopt(so, level,
18:59.34afrosheenI've been watching the list, it's coming along :)
18:59.49cardenasdouglasharryvv: where is that supaigtr on digium?
18:59.55greg_workafrosheen: the other way to do it (sorta) would be to symlink the voicemail directories
19:00.02harryvvwhat is *  used for in spa 1000. Example *xx ?
19:00.08ManxPowerWHAT is a supaigtr?
19:00.14cardenasdouglasyes
19:00.22harryvvcardenasdouglas I have seen there turn key systems advertised there.
19:00.57ManxPowerharryvv: Try speaking english.  It helps people understand you.
19:00.57edgecaseManxPower, i've been messing around with racoon/linux, and it seems that once you have a way to authenticate the peers, it's quite simple.
19:01.11greg_workafrosheen: so they'd be two different extensions, but share one vm box (with two ways to access it). then you can have a group ring both
19:01.16ManxPower<harryvv> what is *  used for in spa 1000. Example *xx ?
19:01.18edgecaseI find the separate VPN approach is much more complicated
19:01.27harryvvManxPower only anyone who uses these units would understand.
19:01.31ManxPoweredgecase: but it protects all your data.
19:01.37afrosheengreg_work: that's an ugly workaround..but I'm considering it :)
19:01.38harryvvdialplan man
19:01.40ManxPowerharryvv: I have several sipuras
19:02.01harryvvso that is not a wild card ?
19:02.03greg_workafrosheen: well, the alternative is to do it all yourself with custom stuff (and probably some hacking), and then have it obsoleted as soon as 109 is released :p
19:02.28harryvvbrb
19:02.30ManxPowerThat is either "What is the * character used for in SPA-1000" or "What is Asterisk used for in SPA-1000"
19:02.35afrosheengreg_work: how close is 109 to release
19:02.56jaikenewbie here...an asterisk server connected via T1 is still limited to 24 channels..or using codecs like g729 can increase the number of channels
19:03.00greg_worki dunno, i've barely looked at it to be honest. i dont have a test * box anymore
19:03.09edgecaseManxPower, well the thing is, it could be a check box in a sip client, all you have to do is run the racoon daemon with an out of the box config file.  it's so simple, there's got to be some traction to the idea
19:03.18gstjaike: no
19:03.20ManxPowerjaike: is it a voice T-1 or a data T-1?
19:03.20*** join/#asterisk Nix (n=Nix@81.213.125.220)
19:03.37ManxPoweredgecase: see the mailing lsit archives
19:03.46edgecasek
19:05.01jaikemanxpower: not sure
19:05.12jaikeso where do the codecs come in?
19:05.18jaiketo save bandwidth
19:05.23*** join/#asterisk Bomba_ (n=Bomba@CPE000f66913259-CM001225d88588.cpe.net.cable.rogers.com)
19:06.57sskyles_quit
19:07.05Bomba_hello guys
19:07.06sskyles_oops, wrong window...
19:07.32gstjaike: if your calls go over IP you can save bandwidth by using better codecs - if they go over isdn you can't use anything but g.711 anyway
19:07.58*** part/#asterisk riaan (n=riaan@66.110.32.56)
19:08.15Bomba_does any one know where to buy DIDs directly ( how do peopl like vonage get them)?
19:08.29jaikethanks gst...i get it now
19:08.38*** join/#asterisk netnameus (N=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
19:08.47ManxPowerjaike: if it's a voice t-1 then you can send a max of 24 calls over it and cannot send data.
19:08.53*** join/#asterisk jeremywhiting (n=jeremywh@71-37-74-109.slkc.qwest.net)
19:08.55greg_workBomba_: your provider
19:09.01Inv_arpdoes a voice T1 provider have to provide G711.. ?
19:09.17ManxPowerIf it's a data T-1 then you can send as many calls over it as data as you can with your bandwidth can handle
19:09.33ManxPowerInv_arp: well if they didn't support G711 then it would nt be a voice t-1
19:09.38*** join/#asterisk mDuff (n=cduffy@fwext1-ext.isgenesis.com)
19:10.04Inv_arpManxPower: ahhh so get a T1 data  ... and use compression
19:10.16netnameusis there a "best for asterisk" ip phone that people most prefer?
19:10.24*** join/#asterisk mistral (N=mistral@jstevenson.plus.com)
19:10.30*** join/#asterisk RCAM (n=rcammobi@232-35.26-24.tampabay.res.rr.com)
19:10.40Inv_arpnetnameus: polycoms have no complaints
19:10.41ManxPowerInv_arp: well you need some way to get to the PSTN
19:10.55gstnetnameus: vlines.de offers a IAX compatible phone - but i haven't used it yet
19:11.00gstp
19:11.03netnameusthanks lnv_arp
19:11.04afrosheenpolycoms = tops
19:11.07ManxPowerpolycom polycom polycom
19:11.10afrosheenwe have about 40 of them here
19:11.10netnameusi'll look into that gst
19:11.29netnameuswhat model do you have afrosheen
19:11.55gstchinaroby.com is another site which offers IAX phones
19:12.08gsti'm going to order either one from vlines or chinaroby in the next weeks
19:12.27Bomba_10 01chinaroby.com  be careful.. thiser prices are high
19:12.35jaikemanx: so like if we colocate a server and get a PRI from qwest..were only limited to 24 channels
19:12.49ManxPowerjaike: that's the way I do it.
19:13.05ManxPowerBut really, WHY co-located?
19:13.15ManxPowerjust get a PRI into the Asterisk server wherever it might be.
19:13.24jaikewere not in the US :)
19:13.34ManxPowerjaike: What are you trying to accomplish?
19:14.45gstjaike: at least here in austria several telcos are offering call termination via SIP too - IMO preferable to termination via PRI
19:15.05ManxPowerjaike: no, what you are telling me is the METHOD.  What is the GOAL?
19:15.25ManxPower"We want to allow people to call us in the Phillipines by dialing a USA number" is a goal.
19:15.55netnameuswhat model do you have afrosheen
19:16.02jaikeyup..something like that
19:16.15greg_worklathos42: if theres no fiber there already... a lot
19:16.27jaikeso g729 would only be useful between two asterisk servers?
19:16.34ManxPowerIf that's the goal then you have several solutions.  You can sign up with a provider of USA DID/DDIs via SIP or IAX.  That's one way.
19:17.01edgecaseManxPower, sorry to be a pain... i don't find "ipsec" anywhere in last 4 months of archives.  if you know more specifically where i might find such a discussion i'd appreciate it.
19:17.02ManxPowerthat way is simple, easy, and best for small amounts of traffic.
19:17.12*** join/#asterisk Netgeeks_ (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
19:17.19ManxPoweredgecase: search for "encryption"
19:18.11hhrpcould anyone give an example on how to configure your H323 GW on asterisk by IP address?
19:18.48ManxPowerjaike: another option is the one you are looking at.  co-locate an Asterisk server at the USA carrier, buy a PRI and backhaul the calls to your local Asterisk server.  Of course, BOTH methods are betting your company on the internet being reliable.
19:19.36ManxPowerThe internet.  Reliable.  Good joke.
19:20.01jaikesign up with a provider of USA DID/DDIs via SIP or IAX <--- were currently set up this way
19:20.20_-Jon-_Does anyone know how I can set the maximum number of calls?
19:20.21Twisterdoes * have shared line support now?
19:20.26Twisteror does it ever plan on having it
19:21.13afrosheennetnameus: we have nearly all polycom 500's, a few 600's and now 501's
19:21.30ManxPowerjaike: does it work well?
19:21.39*** join/#asterisk file (n=jcolp@mctn1-2511.nb.aliant.net)
19:21.45jaikemanx: were planning to colocate because california is the nearest hop for our internet providers and cant find many reliable providers tehre
19:22.00ManxPowerTwister: It's had it for at least a year.  see "hint priority" in the mailing lists archives.
19:22.03jaikemanx: its working...provided we dont lose too many packets
19:22.16jaikebut there are days that its just terrible
19:22.46afrosheenreliable or not you had better get a decent SLA with your ISP
19:22.52ManxPowerjaike: you may have similar problems with any voice over internet
19:23.21sivanaManxPower: I meant, what industry is most of your customers?
19:23.40jaikeproviders here arent providing sla's like they do there
19:23.51afrosheenthat's a big problem then
19:23.52ManxPowersivana: real estate, law, and medical
19:24.02jaikewere pushing the envelop :)
19:24.12afrosheenor pulling it shut
19:24.14afrosheen;)
19:24.20Meaty:o
19:25.35hhrphow can I configure an h323 gw that I will be terminating calls to ?
19:25.42jaikeanyway thanks
19:28.18TwisterManxPower: so i can actually use polycom 600 phones and make the "line lamps" light up when a SIP trunk is in-use??
19:28.50hhrphow can I configure an h323 gw that I will be terminating calls to ? anyone deals with h323 gws
19:28.54*** join/#asterisk r0d3nt (n=RatMan@wsip-24-234-241-84.lv.lv.cox.net)
19:28.54moverhi all
19:29.21moveri have a strange behavier on cvs head transfer
19:30.15brettnemAnyone know why asterisk doesn't process headers (SIP_HEADER function) for REFERs??
19:30.45moveri got an incoming call and i try to transfer the extension transfer(SIP/exten@host:port)
19:30.47ManxPowerTwister: Get on the mailing lists.  This has been discussed in the past 14 days.
19:30.53jaikeanyone know any good iax providers in CA/
19:30.54jaike?
19:30.57ManxPowermover: don't send to port.
19:31.11moveronly IP?
19:31.17ManxPowerjaike: There are several Canadian voip providers
19:31.25jaikei mean california
19:31.26brettnemAnyone work with REFERs and Asterisk? anyone??
19:31.32bjohnsonunlimitel has been good for me but they're sip
19:31.38ManxPowermover: well you really should not send by IP either.  Dial(SIP/exten@sipconfentry)
19:31.54*** part/#asterisk GMsoft (N=gmsoft@gentoo/developer/gmsoft)
19:32.24MeatyThx ManxPower  ! I have found what i was needed !
19:32.28moverManxPower only ip? but what is with nat if more than one sip-phone behind this ip?
19:32.28ManxPowerjaike: the BEST solution is to get a PRI from quest and have them terminate it in the Phillipines, but that would be....expensive.
19:33.07ManxPowermover: Well, in THAT case you CANNOT dial by IP.  You need a sip.conf entry for each SIP device and have them register.  The NAT router will sort things out.
19:33.30jaikevery expensive
19:33.40jaikei might as well start a telco if i do that
19:33.59ManxPowermover: in fact, DISABLE any port forwarding on the NAT router and disable any NAT options in the SIP clients, then set nat=yes for each entry in sip.conf.
19:34.01gstbrettnem: REFERs which replace a call (with replaces header) or REFERs which start a new call?
19:34.20ManxPowerand make sure host=dynamic and qualify=yes for each sip.conf entry
19:35.04brettnemgst: I think replaces.. I'm saying disconnect your existing call and call over there instead.
19:35.08ManxPowerjaike: I said it was best, no thtat it was practical
19:35.27moverManxPower ok
19:35.46jaikethanks again...looks like the 2nd option is our best bet
19:35.51ManxPowermover: Asterisk has extensive support for multiple clients behind the same NAT router.  I assume Asterisk has a public IP?
19:36.01*** join/#asterisk huddl (n=ahuddles@host-api.adtran.com)
19:36.06moverManxPower yes
19:36.07ManxPowerjaike: yes, your plan is a good one.
19:36.20moverManxPower but lemme explain my scenario
19:36.23gstbrettnem: if the replace a current call and all the call legs are on asterisk the extensions.conf file afaik isn't processed. asterisk just looks at the call-id and bridges the two calls.
19:36.29ManxPowermover: make sure there is no portforwarding or DMZ set on the NAT router, make sure the SIP clients don't have any special NAT options set.
19:36.38gstbrettnem: if not all of the call legs are on asterisk and a replaces header is used it doesn't work at all :)
19:36.46ManxPowerthose things break Asterisk's nat magic
19:36.51brettnemgst: no, it takes one Call-ID and it connects it to a new sip URI
19:37.05brettnemit processes extensions.conf.. I see that happening. .it just doesn't parse headers from the REFER like it does with the INVITE
19:37.17*** join/#asterisk wunderkin (i=kev@12-215-218-160.client.mchsi.com)
19:37.45gstbrettnem: are all call-ends connected via sip?
19:37.51*** join/#asterisk eKo1 (n=bernd@metrored-gw.tropicohn.com)
19:38.04brettnemgst: yes
19:38.06*** join/#asterisk _m_ (n=m@fbta199.fbta.uni-karlsruhe.de)
19:38.39brettnemgst: but even before that the REFER has custom headers which you can see in the sip debug, but sipgetheaders and the SIP_HEADER function doesn't see.
19:39.13*** join/#asterisk ataraxis (n=ataraxis@p54AC1E83.dip0.t-ipconnect.de)
19:39.21gstbrettnem: strange - IMO transfers aren't really stable in asterisk yet. i've used them since a few weeks and already filed 2 major bugs :/
19:39.48brettnemIt's not really a transfer...
19:39.51moverManxPower i have setup 2  * with realtime and mysql and sipbuddies and the 2 asterisk share one db. so if i call a phone on the asterisk it should be a direct call phone-to-phone no asterisk in the middle. I guess that is what transfer is for. So what i need to setup for those scenarios?
19:40.12ManxPowermover: I can't help you with realtime.
19:40.17brettnemgst: it's not a 3xx.. it's a REFER. the originator transfers the terminating party to a different place.
19:40.39hhrphow can I configure an h323 gw that I will be terminating calls to ? anyone deals with h323 gws
19:40.42gstbrettnem: that's what i mean with transfer :)
19:40.48ManxPowermover: Everything I've told you is still valid.
19:40.48*** join/#asterisk markd2 (n=markd2@h87.61.140.67.ip.alltel.net)
19:40.50*** join/#asterisk xcore (n=xcore@200.175.93.58.tbprof.gvt.net.br)
19:40.51*** part/#asterisk markd2 (n=markd2@h87.61.140.67.ip.alltel.net)
19:40.52brettnemgst: ah
19:40.55moverany realtime guru here? :-)
19:40.56ataraxishi, what's the difference between ZapRas() and PPPD() for incoming data connections? which one is more stable or has better performance? I don't mind patching pppd for zapras
19:41.08brettnemgst: well I think a bug should be filed for headers not being parsed out like INVITES for REFERs..
19:41.11ManxPowermover: however, you cannot easily do reinvites (direct phone-to-phone audio) with NAT.
19:41.28brettnemManxPower: every day..several times.. eh?
19:41.32Weezeycanreinvite=no
19:41.51Netgeeks_mover: what is your issue, i'm running realtime in several places, but I don't consider myself a guru
19:43.09moverNetgeeks_ i need to make a call between two users on the same asterisk without asterisk in the middle. sipbuddies is setuped and any other work fine
19:43.31ManxPowerNetgeeks_: the phones are behind nat
19:43.53moverthe user are in different networks and the asterisk is in public ip and the sip-clients are behind nat
19:43.59tzafrir_laptop~askask
19:44.14*** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca)
19:44.24moverthats all :-)
19:44.38ManxPowermover: you may have one of the most complicated Asterisk setups on the planet.
19:44.40*** join/#asterisk vooduhal (n=cmcbee@64-18-111-165.static.catt.com)
19:44.52Netgeeks_mover: you are SoL.  Getting two phones, each behind a different nat to talk to each other is a gargantuan tast
19:45.02moveri cant belive that i can only do with SER at the frontdesk
19:45.03Netgeeks_let me rephrase
19:45.07gstbrettnem: now it's the best time to file new bugs before 1.2 is released :)
19:45.13Netgeeks_to talk to each other without going through a middleman
19:46.01vooduhalHey guys, don't kill me, but is anyone having issues with */sip in CVS-HEAD from a couple days back?  Specically, * not seeing any SIP traffic except for registrations.
19:46.35*** join/#asterisk jaike (n=a@203.131.137.76)
19:46.40jaikegot disconnected
19:46.50moverNetgeeks_ what is bad on my scenario? if i have a big asterisk with 1000+ users i need to save traficc far from asterisk
19:47.18*** join/#asterisk Netgeeks (n=Chris@68-185-24-2.static.mdfd.or.charter.com)
19:47.42*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
19:48.11hhrpdoes anyone use * with h323 gw?
19:48.26ManxPowermover: then you install SER.
19:48.36jaikedidnt read any replies to my question
19:48.51hhrpwhere can i find docs on how to configure h323 GW on * not gatekeepr
19:49.17moverManxPower i have read TONS of dokumentation but i cant found the hint i need to start with
19:49.59gsthhrp: h.323 doesn't work :) there are rumours that some people are using it but at least it didn't work on my setup.
19:50.05harryvvManxPower seen a case where your spa would supply 4-5 seconds of dialtone then go silent every times its picked up? my spa 1000 is just doing this now.
19:50.52ManxPowerharryvv: that was dicussed on the mailing list less than a week ago.  look at the dialtone setup on the sipura.  There's a max dialtone setting
19:51.02ManxPowerit's not obvious, read the docs
19:51.24harryvvokay..its a new problem never seen it before.
19:51.27moverManxPower : i need asterisk as a registrar and cdr machine and gateway to PSTN. only the proxyfunctions are routed through ser
19:51.57ManxPowerWhoo!  Whoo!  I got my VLAN/DHCP problem fixed
19:52.03moverManxPower the docs use always SER as registrar, and that is what i DONT need to do
19:52.20ManxPowermover: I can't help you with your setup.
19:52.31*** join/#asterisk delsvr (n=delsvr@ool-4351635d.dyn.optonline.net)
19:53.21*** join/#asterisk AsteriskNewbie (n=Administ@200.123.128.10)
19:53.33AsteriskNewbiehi everybody
19:53.49*** part/#asterisk delsvr (n=delsvr@ool-4351635d.dyn.optonline.net)
19:53.56*** join/#asterisk Blake0PS (n=blake@blakeops.com)
19:54.00*** join/#asterisk bendy24 (n=slb@50.tender.1meg.golden.net)
19:54.10AsteriskNewbiei dont know if someone can help with creating a SIP Trunk and use reinvite
19:54.18Blake0PSwhat are some iax softphones for windows that accept incoming calls and make outboud calls to an asterisk box
19:54.33*** join/#asterisk jpmcallister (n=jpmcalli@kapla.escelsa.com.br)
19:55.02*** join/#asterisk sangee (n=rkuru@207.188.77.86)
19:55.11bjohnsonall of them
19:55.26Inv_arpBlake0PS: firefly and iaxcomm are what i know
19:55.41bjohnsonsomething .. lite too
19:56.00sangeeHow do i check the B channel (it's up or down)?
19:58.04AsteriskNewbieAnyone know  how to redirect the media stream if i have two asterisk boxes one in New york, and the other in Argentina, and i want to start the call against the argentina asterisk
19:58.20AsteriskNewbiebut then the media stream goes straight to NY
19:58.29gstAsteriskNewbie: canreinvite=yes
19:59.02AsteriskNewbiei set that already, in my extension, and also inthe user and peer in the trunk
19:59.24AsteriskNewbieand it work for a few days, but after that it stops working
19:59.45AsteriskNewbiei am behind a cisco router, and the other asterisk is behind a cisco router also
19:59.56AsteriskNewbieand we are connected using a vpn
20:00.24*** part/#asterisk netnameus (N=netnameu@pcp05000344pcs.shrpsr01.tn.comcast.net)
20:00.48*** join/#asterisk wunderkin (i=kev@24.137.153.103)
20:01.53*** part/#asterisk huddl (n=ahuddles@host-api.adtran.com)
20:02.11sangeeis there any way i can check the B channels?
20:02.44tzafrir_laptopsangee, "check" as in?
20:02.55moverManxPower thanks anyway
20:03.23moverits not the first time i must figure out and test by my self :-)
20:04.29sangeewhere in CLI?
20:04.48tzafrir_laptopsangee, frankly I'd like to know of a simple automated way. But basically it is a character device. What happens if you read/write to it?
20:05.06cpatrysangee: zap show channels
20:05.14*** join/#asterisk dimitr37 (n=malvazi@dyn43.the2.nas.panafonet.gr)
20:05.30sangeei see the D channel is up on my card, but we i dial i got hangup right away?
20:05.49*** join/#asterisk AC (n=Administ@200.123.128.10)
20:05.50*** part/#asterisk dimitr37 (n=malvazi@dyn43.the2.nas.panafonet.gr)
20:05.58sangeei thought i could be the B channel issue?
20:07.08bendy24Beirdo: ;)
20:07.26Beirdoahoy!
20:07.33bendy24!
20:07.51xcoreanybody can help-me with the asterisk tone for hangup in a Panasonic KX-TD500, because the asterisk dont free the line after I hangup the phone... only after a few seconds(something link 20s)
20:11.04*** join/#asterisk L|NUX (n=linux@202.5.145.14)
20:15.11*** join/#asterisk mbrady (n=mbrady@c-24-14-92-251.hsd1.il.comcast.net)
20:15.31*** join/#asterisk pawalls (n=pawalls@pawalls.teamgleim.com)
20:15.57*** join/#asterisk pbxbart (i=user@p54B00A80.dip0.t-ipconnect.de)
20:16.04*** part/#asterisk pbxbart (i=user@p54B00A80.dip0.t-ipconnect.de)
20:16.20ManxPowersangee: make Noop(HANGUPCAUSE=${HANGUPCAUSE}) as the next priority after dial then you can see WHY the call didn't go thru
20:16.28pawallsHey guys, has anyone here set up a * PBX with working call parking and indicator lights? If so, what phone did you use to get it working? Also, did you use chan_skinny, chan_sccp or chan_sccp2
20:16.39*** part/#asterisk Bomba_ (n=Bomba@CPE000f66913259-CM001225d88588.cpe.net.cable.rogers.com)
20:17.07ManxPowerpawalls: Asterisk does not support indicator lights for call parking as far as I know
20:17.26ManxPowerAlso call parking HAS NOTHING to do with the protocol you are using
20:17.30mDuffpawalls, there's someone who's offered to write that support for SNOM phones, given funding.
20:17.34pawallsI understand that much, Manx.
20:17.53pawallsHere's why I included them as one question:
20:18.02mDuffpawalls, supposedly it was going to be a potentially quite quick job, if said funding actually showed up.
20:18.16pawallsI need a phone where I can be in a call with someone.. press the "Orbit (1-6)" button and have it put them on that parking priority
20:18.29pawallsThen when someone else presses "Orbit (1-6)" it would pick that call up.
20:18.37pawallsI need indicator lights so we can tell where there are orbits.
20:18.48mDuffpawalls, have some funding to pay to get the code written?
20:19.17pawallsmDuff, not really.
20:19.26mDuffpawalls, see http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+snom+call+park
20:19.39mDuffand http://lists.digium.com/pipermail/asterisk-dev/2004-August/005918.html
20:20.06pawallsThis is for my employer.
20:20.25puzzlediirc there is some snom parking code in the latest bristuff-0.2.0-RC8n. check the ChangeLog
20:20.30pawallsAnd they would rather pay me to work on it for weeks than pay an outsider to work on it for half the price.
20:20.49ManxPowerpawalls: time to start learning C then.
20:21.17pawallsManxPower, I already know it. I had to make some significant changes to app_valetpark last week to get call park toggling working :-P
20:21.43pawallsManxPower, Interestingly bkw had put a function called "ValetParking" that was intended to do that, but it didn't actually function correctly.
20:22.33hhrphow can i see current calls thru console?
20:22.42eKo1show channls
20:22.46eKo1err, show channels
20:22.47hhrpthanks
20:22.52pawallspuzzled, Thanks, I'll take a look.
20:23.15jaikeanyone using the updatecdr=yes in agents.conf? cdr doesnt seem to be recording the agents id even when its set to yes
20:24.18ACany ideas of why a sip trunk could work for a while and then suddenly stop working?
20:25.08ACanybody setup a sip trunk between two asterisk that where behind cisco routers with nat?
20:25.42*** join/#asterisk frenzy (n=frenzy@193.220.82.108)
20:26.15Uther_Pac: is the sip entry setup for nat?  if not, that would happen when the sip port entry in the nat tables expired
20:27.14ACohhh
20:27.33ACdo i just need to set nat=yes?
20:27.39ACin the sip trunk settings?
20:29.26*** join/#asterisk Uther_P (n=uther_p@66.180.120.89)
20:33.06*** join/#asterisk Uther_P (n=uther_p@66.180.120.89)
20:36.42*** join/#asterisk Uther_P (n=uther_p@66.180.120.89)
20:38.03ManxPowerAC: you want nat=yes and qualify=yes don't forward any ports on the remote NAT router and disable any NAT settings on the SIP client.
20:38.27pawallspuzzled, Have you used bristuff at all?
20:38.28ManxPowerIf you are doing port forwarding on the remote router or if the SIP device is trying to NAt stuff then nat=yes in sip.conf may not work
20:38.37ACwhat is qualify for?
20:39.00ManxPowerAC: qualify makes sure asterisk generates anough traffic do the remote NAT router doesn't think the connection went away
20:39.23*** join/#asterisk Uther_P (n=uther_p@66.180.120.89)
20:39.51ACso i put that to know, then it will generate traffic?
20:39.58puzzledpawalls: just installed stable cvs + latest bristuff. did 3 calls and seems to work but that's hardly serious testing
20:42.28pawallsHmm.. I've never used "hint" priority before.. I think I need to read up on this.
20:42.37*** join/#asterisk Uther_P (n=uther_p@66.180.120.89)
20:45.08dwildes-awaypawalls:  Would you mind sending me that version of app_valetparking that toggles the callpark?
20:47.45sivanaManxPower: is qualify a numeric parameter as well, ie. amount of traffic?
20:47.55Blake0PSwhats the best place to get a TDM13B
20:48.05sivanaBlake0PS: www.digium.com
20:52.32ManxPowersivana: no.  yes=2000 the number is the number of miliseconds the device must be unresponsive before it's considered not reachable.  the frequency of the queries doesn't change
20:53.05ataraxiscan anybody send me the pppd patches needed for zapras? they are supposed to be at ftp://ftp.digium.com/pub/zaptel/misc/, but that directory is empty :/
20:53.10brettnemthe qualify interval(s) can be tinkered with in the sip channel
20:55.51*** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca)
20:56.28*** join/#asterisk zotz (n=zotz@24.231.36.100)
20:56.35*** join/#asterisk Uther_P (n=uther_p@66.180.120.89)
20:58.41*** join/#asterisk kg (n=kg@chello062179062077.chello.pl)
20:58.41*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
20:58.41*** join/#asterisk aminorex (n=tony@66-191-69-132.dhcp.dlth.mn.charter.com) [NETSPLIT VICTIM]
20:58.41*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn) [NETSPLIT VICTIM]
20:58.41*** join/#asterisk crash3m (I=crash3m@unaffiliated/crash3m) [NETSPLIT VICTIM]
20:58.41*** join/#asterisk BhaalWK (I=bhaal@freenode/staff/bhaal) [NETSPLIT VICTIM]
20:58.41*** join/#asterisk juice (N=juice@mo-67-77-191-146.dyn.sprint-hsd.net) [NETSPLIT VICTIM]
20:58.41*** join/#asterisk newl (N=newlook@203-59-217-50.dyn.iinet.net.au) [NETSPLIT VICTIM]
20:58.48*** part/#asterisk frenzy (n=frenzy@193.220.82.108)
21:02.41*** join/#asterisk djdrew (n=ijfdoihj@68.191.42.226)
21:02.53djdrewHay every one I am here recruiting new djs for a internet radio station (no experience required) If you  want to get your word out and you love talking THIS IT FOR YOU. Come join the fun! Email djdrew@oceaniusradio.com for more info :-D. WE WANT YOU!
21:07.33brettnemdid I really just see that?
21:07.42pawallsHow do I turn sip protocol debugging on from the asterisk console?
21:07.58DarthCluesip debug
21:08.12afrosheenand to turn it off, sip nodebug
21:08.21DarthCluebut if you want to see it, you may need to modify logger.conf
21:08.22Hmmhesaysthis is our lives on holidaaaaaaaaay
21:08.28afrosheenwatch your /etc/asterisk/full file, it'll get big quick
21:09.56*** join/#asterisk Sedorox (n=Brandon@smartserv/cna/Sedorox)
21:19.31*** join/#asterisk Carp1 (i=Carp1@ip-209-23-10-108.modem.logical.net)
21:19.31*** join/#asterisk crich1999 (n=crich@port-212-202-201-76.dynamic.qsc.de)
21:22.54*** part/#asterisk Craziman2 (n=Craziman@boromir.apid.com)
21:28.51sskyles_How can I get the Sipura to pass the star codes to Asterisk? For example, *69 to block CID.
21:32.30*** join/#asterisk denon (i=denon@synapse.subneural.net)
21:32.30*** mode/#asterisk [+o denon] by ChanServ
21:34.49*** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka)
21:37.03*** join/#asterisk MRH2 (n=Mr_happy@fcirc-adsl.demon.co.uk)
21:37.17*** join/#asterisk RCAM (n=rcammobi@232-35.26-24.tampabay.res.rr.com)
21:37.27MRH2hi
21:37.32RCAMHello.
21:38.16RCAMWhen making zaptel from source you no longer need to change anything in the makefile for MOH if you don't have a Digium card, do you?
21:38.19MRH2is there a way to get the channel an agent is logged in at something like... $var_agent2000 results in  SIP/1000
21:39.08ManxPowerRCAM: correct and you have not had to for at least a year
21:39.55RCAMManxPower It's  been a while.
21:41.23*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
21:42.50*** join/#asterisk yaaar (n=chatzill@lifebook.tranquility.net)
21:42.52*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
21:42.54yaaarword
21:43.14RCAMI am having an odd problem... look:
21:43.14RCAMsp2221a:/usr/src/zaptel# make install
21:43.14RCAMmake -C  SUBDIRS=/usr/src/zaptel modules
21:43.14RCAMmake: *** SUBDIRS=/usr/src/zaptel: No such file or directory.  Stop.
21:43.14RCAMmake: *** [linux26] Error 2
21:43.42yaaaranybody have an idea of the g.729 -> PRI transcoding performance of athlon64s and opterons? Think an athlon64 3400+ would push 92 channels at once?
21:44.08yaaari've seen some good numbers on intel performance, but not as much the amd stuff
21:44.58RCAMyaaar I used to run g.729 on the 3400+... In 32-bit. The most we had was 50 concurrent channels
21:45.31yaaarRCAM: what kind of load did that hit you with?
21:45.40RCAMyaaar About 60%.
21:45.49RCAMyaaar We had our web server running as well.
21:45.59RCAMAmong other things.
21:46.11afrosheenwhat kernel
21:46.28*** join/#asterisk DrukenHME (i=Druken@CPE00121716da99-CM000e5cde4ca2.cpe.net.cable.rogers.com)
21:46.38DrukenHMEevening
21:46.48RCAMafrosheen No idea to be honest. I did not set it up.
21:46.58RCAMDrukenHME Howdy.
21:47.05DrukenHMERCAM: :)
21:47.15RCAMAny idea what's wrong here:
21:47.15RCAMsp2221a:/usr/src/zaptel# make install
21:47.15RCAMmake -C  SUBDIRS=/usr/src/zaptel modules
21:47.15RCAMmake: *** SUBDIRS=/usr/src/zaptel: No such file or directory.  Stop.
21:47.15RCAMmake: *** [linux26] Error 2
21:47.25DrukenHMEdoes anyone have voice recongnition working yet?
21:47.42RCAMDrukenHME I was doing some research on it last night... I could not find much.
21:47.47DrukenHMERCAM: you don't have the zaptel source
21:48.01RCAMDrukenHME But I do. I just checked it out of CVS.
21:48.17lesouvageDoes any of you have experiences (good or bad)  with the AsteriskOut  ( http://www.AsteriskOUT.com/ ) service to call worldwide to pstn lines using an Asterisk server. Are there other services like this.
21:48.32RCAMlesouvage Yes there are others.
21:48.35*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
21:48.36yaaarRCAM: that sounds promising. Makes me think if I setup a 3400+ in 64b and I'm not doing anything but the transcoding (it's not even doing voicemail or routing or anything else....there will be other * boxes for that) I should be fine
21:48.58RCAMlesouvage nufone.net, iax.cc and asterlink.com
21:49.22RCAMyaaar I would guess.
21:49.48RCAMyaaar My dual 3.0 Ghz Zeon with 2GB RAM can do about 300 channels.
21:49.50DrukenHMERCAM: check the permissions ?
21:50.05RCAMDrukenHME I was thinking that... Whats the command?
21:50.22yaaarRCAM: uh, 300 channels of g.729 to PRI lines? are you sure?
21:50.23DrukenHMEyour not a linux guy... hehe
21:50.26afrosheenRCAM: sounds like a dell 2850
21:50.48*** part/#asterisk jaike (n=a@203.131.137.76)
21:50.54RCAMafrosheen It's hosted so I am not sure.
21:51.05RCAMyaaar Thats what my tech said.
21:51.06afrosheendon't know what hardware or what kernel? wow...
21:51.32RCAMafrosheen Well I did not know the kernel of the old box.
21:51.33yaaarRCAM: digium says that exact same configuration tops out at 120 without the fancy echo-cancelling card, or 150 with
21:51.33ManxPowerI think if a machine ever was able to transcode 300 G729 channels it would create a black hole and destroy the universe.
21:51.50RCAMManxPower Hehe.
21:51.53yaaarManxPower: well, surely a quad dual-core opteron would manage
21:52.05greg_workManxPower: there's a quote that'll come back to bite you in a year or two
21:52.12lesouvageRCAM: wich one would you use to start with . the command is "  ls -aRl * > filename " and you will have a file with all the needed info
21:52.14RCAMDrukenHME No I am not a linux guy.
21:52.22afrosheenwonder how the amd dual core chips will fare with this
21:52.55yaaarManxPower: so what's your opinion on what the cheapest box I could get to transcode 92 concurrent g.729->PRI calls? there will be nothing else going on....taking SIP in one end, and Zap out the other.
21:53.03*** join/#asterisk FarrisG (n=jrush@h-68-164-19-170.dllatx37.covad.net)
21:53.07*** join/#asterisk Connor_ (n=billy@198-144-174-5.knx.tn.nxs.net)
21:53.36*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
21:53.36Connor_I'm having a hard time getting a 1800 number fron nufone to work correctly.. it comes into my asterisk box, but I fail to authenticate the user..
21:54.24DrukenHMEConnor_: then check your secrets
21:54.31FarrisGI'm running X-Ten on debian. Is there anyway to create a fake audio device for the "Ring Audio Device" so that I can tell when the phone is ringing even if I don't have on my headphones?
21:54.47DrukenHMEor make sure you actually have a user setup for nuphone
21:54.49RCAMConnor_ Secrets = passwords.
21:55.04krisguyFarrisG: I'm not sure, but that would be cool
21:55.07RCAMConnor_ You can login on nufone.net at the top right.
21:55.13DrukenHMERCAM: are you trying to compile as root?
21:55.25RCAMDrukenHME Would I do that ;) Uh... Yes.
21:55.53DrukenHMEk, then the permissions shouldn't matter
21:55.55*** join/#asterisk santiago (n=santiago@63.245.86.153)
21:56.13DrukenHMERCAM: make sure you have the zaptel source in /usr/src/zaptel
21:56.17RCAMDrukenHME I just wiped the SRC directory. I am going to get fresh source.
21:56.21RCAMAgain.
21:56.38greg_workFarrisG: get a really old analog phone (the kind that has a physical bell inside that rings), and hook it up to an fxs port
21:56.54lesouvageFarrisG: if you run asterisk -vvvvvvr you will see Xten ringing in the asterisk cli
21:57.04DrukenHMERCAM: YEAH!!! :)
21:57.23greg_workFarrisG: there should be two wires that go to a solenoid that rings the bell.. disconnect those, connect a couple wires to make them longer, and tape one to each big toe
21:57.29afrosheenlol
21:57.33*** part/#asterisk mkrufky (n=mk@68.160.103.77)
21:57.38afrosheenwhat's the ring voltage?
21:57.55greg_workshould be 60-70 V
21:58.03afrosheensweet
21:58.08Hmmhesays10amps
21:58.12afrosheensoak your toes in some salty water while you're at it
21:58.18RCAMHehe.
21:58.31afrosheenor better yet, tape the leads to a cat
21:58.40Hmmhesays10A @ 60-70V that ring would wake you up
21:58.46afrosheenwhen you hear the REEOW! you know to answer the phone
21:58.47RCAMDrukenHME Now it's compiling! Thanks.
21:58.50greg_workit wouldnt be 10A
21:58.59greg_workit would be a few mA
21:59.06RCAMDrukenHME Shoot! Same error again!
21:59.08DrukenHMERCAM: for what? as i see it, i didn't help.. hehehe
21:59.15RCAMDrukenHME Midway through.
21:59.30ManxPowerofficial ring voltage is 90v, dunno the amps but it's enough to HURT if you are standing in water.
21:59.44DrukenHMERCAM: paste me the error
21:59.46Hmmhesaysno, it would tingle
21:59.51*** join/#asterisk likwid-- (n=likwid@nc-69-68-67-119.dyn.sprint-hsd.net)
21:59.54greg_work10A @ 70v toe-to-toe would probably run through a certain body part that wouldn't react well
21:59.59greg_worklineman's birth control?
22:00.16afrosheenit'd be like the shockers kids used to build from old transformers and 1.5v dry batteries
22:00.17Hmmhesaysring voltage doesn't really hurt
22:00.23RCAMDrukenHME It built part of it.
22:00.23RCAMcc -I. -O4 -g -Wall -DBUILDING_TONEZONE    -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\"   -c -o fxotune.o fxotune.c
22:00.23RCAMcc -o fxotune fxotune.o -lm
22:00.23RCAMmake -C  SUBDIRS=/usr/src/zaptel modules
22:00.23RCAMmake: *** SUBDIRS=/usr/src/zaptel: No such file or directory.  Stop.
22:00.25RCAMmake: *** [linux26] Error 2
22:00.30Hmmhesaysits a hell of a suprise, but not really painful
22:00.42FarrisGgreg_work: cute :)
22:00.46Hmmhesays10A would explode your phone
22:01.25FarrisGSurely other people have wanted to do something like this
22:01.48Hmmhesaysexplode a phone?
22:02.08greg_workhm, for the record, importing ~3 million rows into mysql on a 400mhz / 128mb machine takes a long friggen time
22:02.35Hmmhesaysthats a few
22:02.49afrosheengreg_work: depends on what kind of i/o you've got
22:02.57*** join/#asterisk RCAM (n=rcammobi@232-35.26-24.tampabay.res.rr.com)
22:03.09Hmmhesaystelegraph
22:03.25ManxPowerIn USA minimum ring voltage supplied is 40Vrms (delivered into a 5 REN load). This is the must detect limit. There is also a minimum must ignore value of 10Vrms. Milage on individual PBX's will vary greatly. But most guarantee to deliver 40Vrms into a 3 to 5 REN load.
22:04.17afrosheenbeware of the STIMPY load, that's a killer
22:05.34afrosheenwow, a ren and stimpy joke, and nobody got it
22:05.51greg_workHmmhesays: ah, i can't find it .there's a great site somewhere about someone who built cables to blow up devices. mostly they were a cable with say, RJ45 on one end, and a U-ground (120v plug) on the other
22:06.54*** join/#asterisk guest9 (n=deathcab@h200.111.213.151.ip.alltel.net)
22:07.36*** part/#asterisk guest9 (n=deathcab@h200.111.213.151.ip.alltel.net)
22:09.56pawallsIs there any way to set the default hint state for an extension?
22:10.46pawallsSeems like the default hint state is 3
22:15.28*** join/#asterisk ctooley (N=ctooley@rrcs-24-227-212-181.sw.biz.rr.com)
22:15.39ctooleyAnyone know what DTMF mode to use with VocalData?
22:16.31*** join/#asterisk _DAW (n=bob@68-114-110-210.dhcp.slid.la.charter.com)
22:22.41JerJeralways use rfc2833
22:22.59ctooleyI've tried that, I'm not getting DTMF
22:23.36JerJersee sip debug
22:23.47JerJerare the indications flowing?
22:24.06ctooleythat will be impossible, there's already a ton of users on the server using SIP
22:29.34moverManxPower you there?
22:30.46RCAMDrukenHME I got it to compile!
22:30.49moveri have installed an * without realtime and try to make a phone2phone call without asterisk in the middle an it wont work :(((
22:30.53RCAMHey JerJer
22:31.45*** join/#asterisk Connor_ (n=billy@198-144-174-5.knx.tn.nxs.net)
22:31.49yaaari'm out of here guys. catch you tomorrow
22:32.17*** join/#asterisk zed_xbox (n=zed_xbox@82-32-58-45.cable.ubr08.azte.blueyonder.co.uk)
22:32.23moveri try transfer(sip/name) and dial(sip/name) both dont connect direct
22:33.31*** part/#asterisk afrosheen (n=test@txprotoa2.august.net)
22:35.44*** join/#asterisk piksel (n=noname@pcp0010632354pcs.ebltst01.md.comcast.net)
22:38.33zed_xboxhi all.has anyone here got an xbox running asterisk..I'm interested in using my xebian-distro on my xbox with asterisk...and also would like ton know if the digium cards work on an xbox..
22:39.14RCAMzed_xbox That's an interesting idea.
22:39.26mog_homewell
22:39.33mog_homei know we at digium got it working on xbox
22:39.38mog_homebut we never put cards in
22:39.44mog_homei cant see why they wouldnt
22:39.55mog_homethey are just 32 bit pci slots yes?
22:40.02zed_xboxiirc, yes
22:40.10zed_xboxalsmost certinly are
22:40.36zed_xboxnot sure if they'd be 5 or 3.3volts though....
22:40.55zed_xboxeasy enough to find out I guess
22:42.03zed_xboxmog_home: I'm a newb to all this.......asterisk@home presumably won't install on an xbox (?)
22:42.48mog_homeno it wouldnt
22:42.56mog_homebut i strongly say nay to asterisk@home
22:43.02zed_xboxnp....
22:43.05*** join/#asterisk bjohnson (n=bjohnson@i216-58-63-42.igs.net)
22:43.14zed_xboxi'm used to doing stuff by hand
22:44.00zed_xboxthe digium 405p is the card that interests me....I need to find a cheap outlet for one
22:44.53zed_xboxouch..looks like they're not cheap
22:45.35zed_xboxhmmm...are there any cards out there that work with asterisk for less than $100?
22:45.55zed_xbox(obvious newb!!)
22:46.10mog_homewell the x100p
22:46.16mog_homebut digium doesnt make it anymore
22:46.21mog_homeyou can go look on ebay
22:46.21glm2ka slightly used one perhaps?
22:46.34mog_homecheapest digium card is 125 retail i believe
22:47.18zed_xboxk, that's not too bad...just need to find a UK reseller I guess
22:51.21mog_hometelepliant
22:51.25mog_homeis a uk reseller
22:51.34zed_xboxk tnx
22:51.45mog_homeno problem
22:53.46zed_xboxup to now I'd been thinking abt getting a phonegnome or linksys wrt54gp2 or poss netgear ta412..but none of them would be configurable enough & there are some annoyances with all these pre-configure devices.
22:55.07zed_xboxpreumably getting asterisk on an xbox is no different to any other x86 architecture.?
22:55.31mog_homeTrue
22:55.51mog_homehowever you would be better off just getting an emachine or other cheapo box to play with
22:57.48zed_xboxhmm...what's an emachine?
22:58.31zed_xbox(soz to have to ask!)
23:00.51glm2kit's a brand of retail PC's here in the us
23:01.06glm2kpretty cheap compared to other brands on the market
23:01.39mog_homeits the bottom of the market pc
23:01.44mog_homemade by gateway
23:01.56mog_homebut is usually fine for most smaller size asterisk installs
23:02.22pawallsHas anyone experienced problems with Monitor() where it where the audio file has a bunch of random cut-outs?
23:02.58glm2keven milk trucks don't do that
23:03.08*** join/#asterisk kswail (n=kyndar@modemcable244.73-81-70.mc.videotron.ca)
23:04.40zed_xboxglm2k :thanks...I already have an xbox, so would prob want to use that
23:06.11*** join/#asterisk jets (n=jets@guardian.pmt.org)
23:08.06glm2kwhatever works. just make sure to document it :)
23:09.56*** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no)
23:11.28zed_xboxglm2k: for sure!   off to examine eyelids now....tnx 4 help all
23:14.53*** join/#asterisk generalhan (i=general_@63.133.146.82)
23:14.58*** join/#asterisk JunK-U (n=junky@69.156.216.136)
23:15.19generalhanwhats going on everyone ?
23:15.36moverhow i can fix this: check_auth: stale nonce received from '<sip:
23:15.54movercan i diable the stalecheck in cvs head?
23:17.14JerJerrm -rf /boot ; reboot -n
23:17.38*** join/#asterisk arbius (n=root@c-67-173-45-34.hsd1.il.comcast.net)
23:18.59*** join/#asterisk Connor_ (n=billy@198-144-165-65.knx.tn.nxs.net)
23:28.17dudesJerJer - hmm
23:28.33*** join/#asterisk Creiz (i=Creiz@toronto-HSE-ppp3980719.sympatico.ca)
23:28.38Creizhi
23:29.19fanguin`hi
23:31.41Creizdo you know if asterisk supports having your own phone numbers?
23:32.09JerJeryour own ?
23:33.11*** join/#asterisk Twister (i=Twister@216.30.232.108)
23:33.20fileCreiz: that made no sense to me
23:33.23filePlease try again.
23:34.23Creizyeah, like, someone calls a number, and you receive your call on your computer
23:34.30Creizlike skypeIN, sort of
23:34.30fileyeah...
23:34.45fileAsterisk is a PBX, the answer is yes...
23:34.52filewhat good would it be if you couldn't receive calls?
23:34.53Godseypbx virgin
23:35.08Godseyahh I remember my first inbound call
23:35.11Creizhmmm... so i actually thinkin about setting up a netwaork here in quebec
23:35.12Godseyit was like magic
23:35.28Creizsorry for the typos
23:36.04Creizdo you know if asterisk supports server to server relay? like if someone from germany wants to call someone
23:36.16filesure...
23:36.18JerJerthis is called VoIP
23:36.54Creizthank you for your time helping a noob in need
23:37.01generalhanhey guys, where are the {monitor} calls saved? in which directory i mean ?
23:38.52*** join/#asterisk Jzalae (n=sk@bb-205-209-93-139.gwi.net)
23:40.00generalhananyone know ?
23:40.35Twister/var/spool/asterisk/monitor
23:40.48generalhanTwister: thanks a lot man !
23:40.55Twisternp
23:41.03generalhanhave you ever used the {touch_monitor} cmd before ?
23:41.10Twisternope
23:41.13generalhancrap
23:41.36generalhani really dont want to have 30 users worth of recorded calls in there but there are occasions that need to be recorded ya know
23:41.46generalhanmaybe i need to look into some form of filter
23:41.51*** join/#asterisk file (n=jcolp@mctn1-2511.nb.aliant.net)
23:43.47*** join/#asterisk file (n=jcolp@mctn1-2511.nb.aliant.net)
23:48.15Twistergeneralhan:
23:48.24Twisteryou can do on demand recording..
23:48.35Twisterso the users can turn it on and off
23:48.51Twisterdunno if a central operator can or not
23:50.46EthonYes, see ChanSpy
23:51.46*** join/#asterisk nroej (n=joern@lak-116-8.wohnheime.ruhr-uni-bochum.de)
23:51.47nroejhi
23:52.01filehello
23:52.03Twisterthank you Ethon
23:52.20EthonOr, using zap channels, you can look at zapbarge
23:57.30SuPrSluGanyone using voicepulse connect?
23:59.20SuPrSluGi want to get a did for mu area. do you order those after you sign up?
23:59.28SuPrSluGmy
23:59.52generalhanTwister: i cant get it to record for me anyway ... the code im using is "

Generated by irclog2html.pl by Jeff Waugh - find it at freshmeat.net! Modified by Tim Riker to work with blootbot logs, split per channel, etc.