irclog2html for #asterisk on 20051007

00:00.08obsidian-studiosjskcr: in ~x86 tree?
00:00.19marc324how to delete full lines in vi?
00:00.26Ariel_dd
00:00.27Mother_AAAARGH
00:00.28IcemaannAriel_: I can use pastebin if you want. Basically, i have extensions 1000, and 2000 in home. they dial VoiceMailMain. The context line for the sip user is home. When he dials 1000 it doesnt work. Ill figure it out. Thanks!
00:01.01Ariel_marc324,  use nano instead
00:01.18hardwiremarc324: you are gonna cry when I say this
00:01.21Ariel_Mother_, are you ok, do you need some water?
00:01.25hardwirebut learn how to run a linux workstation first
00:01.33hardwirejust do it
00:01.38Mother_Ariel_: thanks, I just fell off the chair, I'm OK now...
00:02.06*** join/#asterisk joat (n=joat@laketaylor.org)
00:02.18SkramXlocalhost / #
00:02.27Mother_wrong window
00:02.33SkramXwooops sorry.
00:03.35jskcr<obsidian-studios>  yes the beta in the x86 tree
00:04.03obsidian-studiosjskcr: I sync on Sundays and did not see it?
00:04.22jskcrthe beta-1.2.0 is in the cvs tree
00:04.30jskcrerr not cvs in the portage
00:05.07harryvvrain rain rain
00:05.14harryvvwe are in the rainy season
00:05.18Ariel_yes we are
00:05.28harryvvyour in florida
00:05.29harryvv:)
00:05.29marc324vi crashed. swap file created. where is .swp file
00:05.30Ariel_looks like it's going to rain till the morning.
00:05.35harryvvliquid sunshine there
00:05.36Ariel_harryvv, yes so
00:05.36harryvv:)
00:05.51Ariel_well at least it cools it down to 80 degrees
00:05.56obsidian-studiosharryvv: opposite of normal weather, normally more sun than rain, this year?
00:06.08harryvvmy backyard is looking like a mud pit ;) yanked a large tree stump out of the ground now filling in the hole.
00:06.15Ariel_Our sunny weather comes after this month
00:06.17SarahEmmmarc324: this really isn't the place for unix help.
00:06.26SarahEmmmarc324: if you need * help, okay, but... unix help this isn't the place.
00:06.36harryvvobsidian-studios most days of the week are rain in the winter for washington and bc canada.
00:06.42IcemaannAriel_: thanks for your help. i was doing it correctly, just had a typo.
00:06.45Ariel_marc324, hahaha look for .name.swp
00:06.50harryvvonce you head into the interior of bc is mostly snow.
00:06.52*** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
00:07.00sleepy_onemsg marc324 ls -al |more
00:07.06sleepy_onewhoops
00:07.06obsidian-studiosharryvv: yes, why I left Northern CA, I could not take 3-4 months of overcast rainy weather, much more north, no thank you
00:07.09sleepy_onels -alt |more
00:07.12Mother_locate .filename.swp
00:07.26sleepy_onefind . -iname "*.swp"
00:07.38Mother_so many ways to do the same thing :)
00:07.45*** join/#asterisk Softpac2000 (i=user@rrcs-24-227-32-203.se.biz.rr.com)
00:08.10harryvvWwe pacific northwesterners used to say we have webbed feet :)
00:08.18sleepy_oneahhhh yes indeed, Mother is absolutely correct :-)
00:08.21justinudry as a bone here
00:08.46obsidian-studiosharryvv: FL is known for rain but we have been getting allot this year, more than normal
00:08.54Ariel_I used to live in the northwest. It's nice. But wet
00:08.59harryvvAriel did you read the nifty news on cnn.com technoligy page how a wisp owner headed to new orleans and hooked up a voip asterisk service?
00:09.14harryvvsummers here are great
00:09.16Ariel_obsidian-studios, it's ok it's going to be sunny soon
00:09.37harryvvits the soggy wet days of winter where it can rain almost consistantly for days.
00:09.42*** part/#asterisk Softpac2000 (i=user@rrcs-24-227-32-203.se.biz.rr.com)
00:09.44Ariel_harryvv, yes saw a something on it.
00:09.56obsidian-studiosAriel_: I am holding out hope, running out of beach time though
00:10.11Ariel_well it's suppose to be sunny this weekend
00:10.24Ariel_too bad water is too cold in winter for me.
00:10.26*** join/#asterisk wundaboy (n=asdf@67.189.30.47)
00:10.35harryvvI rmemeber the winter of 98, pinapple express one after the next slamming our state for 3 weeks non stop. Put a world seasonal record of 100 feet of snow on mount baker.
00:10.59obsidian-studiosAriel_: nothing compare to Pacific temp though ;)
00:11.09fiferharryw: where exactly are you?
00:11.13harryvvThat was the time when I worked in the cozy warm buildings of microsoft.
00:11.25harryvvfifer, near Vancouver BC
00:11.26obsidian-studiosAriel_: what part? Im in Jax
00:11.33Ariel_Vancover is nice. But I like Victoria Island better.
00:11.35fiferAh
00:11.37Juggieharryvv, how about 1meter of snow in 2 days
00:11.41Ariel_obsidian-studios, Miami
00:11.51harryvvohh yea there are much dryer parts of vancouver island then here.
00:12.07*** part/#asterisk blkbearnh (i=turner@c-24-147-155-3.hsd1.nh.comcast.net)
00:12.07obsidian-studiosAriel_: almost tropical there, I would be heading to Bahamas every chance I got ;)
00:12.16fiferVictoria is my favorite city on the continent
00:12.17Ariel_no need
00:12.21Ariel_we have the keys
00:12.46harryvva tanker fire on the bridge to the keys the other day.
00:12.48obsidian-studiosAriel_: it's ok, but not the same mentality, or kettle drum music
00:12.51Ariel_victoria has great gardens
00:13.10Ariel_obsidian-studios, you have not been to the tiki hut then.
00:13.20Juggieharryvv, http://www.stuffintheair.com/images/Snowhouses.jpg
00:13.24pc2how bad is iax/g711u over 65 ms?
00:13.25Ariel_harryvv, yes it happens every other week
00:13.29obsidian-studiosAriel_: love to fish in the keys though, but hardly any beaches when I was there long ago hotels made their own beaches ?
00:13.44harryvvbtw, want to goto a soaker city and live? goto Forks washington. Thay get 80 inches of rain a year. On the oposite side of the olympics resides Squim Washington with 18 inches of rain a year.
00:13.46harryvv:)
00:13.49Ariel_beaches you need to head south to the dry tortugus
00:14.29harryvvjuggie, did u take that pic?
00:14.39Ariel_wow 80 inch. This month along we are going to get 17 inches
00:14.42obsidian-studiosAriel_: tortugus?
00:14.49Juggieharryvv, no.
00:14.55justinui think we got 18 inches of rain this year... it was a record
00:15.22Ariel_18 inch is nothing
00:15.22obsidian-studiosjustinu: end of June we got like 10 in less than 5 days in Jax
00:15.30justinuyeah, this is a desert
00:15.49harryvvhttp://classic.mountainzone.com/news/99/bakerrecord.html
00:15.57*** join/#asterisk huslage (n=huslage@c-67-169-200-122.hsd1.or.comcast.net)
00:15.59harryvva offical record by noaa
00:16.00harryvv:)
00:16.26Ariel_last month we had a low for Sept only 16 inches of rain.
00:16.32harryvv<PROTECTED>
00:16.32*** join/#asterisk opus__ (n=opus@dahphish.org)
00:16.35justinui like it dry
00:16.43justinui can keep the top down year round on my s2000
00:16.45opus__is there some kind of dynamic universal way to detect the external ipaddress?
00:16.53justinuSTUN works ok
00:17.09harryvvThe snow on mount baker was so deep thay had to dig out the chair lifts and dig a ravine for them to go though.
00:17.12opus__are there public STUN servers that do this? is there any example script/code that can pull this from STUN?
00:17.16pc2justinu - rain isn't such a problem here.  But it's damn cold.
00:17.19justinuyeah
00:17.22obsidian-studiosit was nice in CA to wash your car 1 or twice a month in the summer :(
00:17.24Ariel_opus__, fwd has one
00:17.26justinuthere's some stun implementations
00:17.29justinui found one in java
00:17.36justinuand stun.xten.net works
00:17.41opus__ok
00:17.45justinuthere's an RFC for it too
00:19.58harryvvThats funny, mount baker closed for the season because of to much snow
00:19.59harryvv:)
00:20.09pc2lol
00:20.12harryvvwhy need a stun server?
00:20.16pc2Not many skiers think of having that problem.
00:20.16pc2hehe.
00:20.23lanceyanyone here familiar with chan_sip?
00:20.35justinuthe code?
00:20.39harryvvpc2, read that link
00:20.40harryvv?
00:20.47pc2harryvv - I did.
00:20.50harryvvyea
00:20.57harryvvthat was a sucky 3 weeks.
00:20.59pc2harryvv - pic says it all
00:21.00pc2hehe
00:21.09pc2take snowmobiles around town =).
00:21.13harryvvyea
00:21.16harryvvno kidding.
00:21.53harryvv:)
00:21.56Mother_grrr I can't seem to be able to read a variable I pass to an AGI script
00:22.27opus__I think my WRT54g is stripping my ToS bit
00:22.28*** join/#asterisk huslage_ (n=huslage@c-67-169-200-122.hsd1.or.comcast.net)
00:22.36opus__is there a way I can turn this on with IPTABLES?
00:22.38Ariel_use NoOp(${NAMEOFVARIABLE})
00:22.55*** join/#asterisk Pete_Largo (n=PeteLarg@225-196.35-65.tampabay.res.rr.com)
00:23.00Ariel_stripping. no don't say it's so. Get Freeman on it
00:23.00Pete_Largohello all
00:23.09Mother_Ariel_: thanks, what will that do? (out of curiousity)
00:23.28harryvvopus, that is a router with asterisk on it?
00:23.28lanceyopus__ : have a look at openwrt
00:23.32Ariel_NoOP will display via the extension on the CLI what the variable has
00:23.33pc2opus__ - Is it possible to use iptables for good packet shaping?
00:23.40opus__harryvv , no
00:23.46Ariel_Freeman/freeman
00:23.46opus__pc2 I think so
00:23.53opus__http://edseek.com/~jasonb/articles/traffic_shaping/scenarios.html
00:23.58opus__i need to run sveasoft
00:24.02opus__it has a lower TCO
00:24.10Ariel_opus__, use freeman
00:24.15pc2freeman?
00:24.15opus__i can't afford to waste an entire day on it
00:24.19Ariel_it's the gpl of sveasoft
00:24.25Mother_ah OK - what I'm doing is a DBGet(HD_ID=whatever), then the next line is AGI(script.agi,${HD_ID})
00:24.26opus__freewan is kind of buggy
00:24.32jskcrpc2 yes
00:24.52Mother_the variable is correctly set to the DB contents, I can see that in the CLI
00:25.03jskcror you can set tos with iptables
00:25.21pc2jskcr - tos = pita
00:25.26Mother_but no matter how I try to access it from the agi script it's always empty
00:25.32pc2tc isn't standard linux though :P
00:25.34jskcrpita?
00:25.36Juggiemother, what are you programming agi in?
00:25.37pc2pain in the ass
00:25.42opus__'tc' is on the sveasoft firmware, hmm.
00:25.45jskcrnot really
00:25.55opus__jskcr can you explain how i can get it working?
00:26.08pc2Someone needs to make a web front end to tc :)
00:26.16jskcriptables -A OUTPUT -t mangle -p udp -m udp --dport 5060 -j DSCP --set-dscp 0x28
00:26.16jskcriptables -A OUTPUT -t mangle -p udp -m udp --sport 10000:20000 -j DSCP --set-dscp 0x28
00:26.20Mother_Juggie: bash
00:26.42jskcrhttp://www.netfilter.org/
00:26.53Juggiemother, it should be like arg1,arg2, etc
00:27.06pc2jskcr - Your upstream router has to acknolowege tos and act accordingly.  I don't know any that do.
00:27.16opus__pc2 - most do
00:27.19Juggiehowever bash receives parameters from the command line
00:27.19Mother_yep, I can read fine arguments like agi_callerid etc. but this one is a nono
00:27.20lancey[03:22] <opus__> i need to run sveasoft
00:27.27lanceyhave a look at openwrt!
00:27.32lanceyit's worth
00:27.43opus__lancey Linux is only free if your time is worthless
00:27.44Mother_Juggie: you mean the variable's name will also be ARGx?
00:27.50pc2opus__ - Most upstream that I know of will just ignore your tos bit.
00:27.58jskcrget a cheap dlink and throw openwrt on it and it will work fine with tos
00:27.59pc2opus__ - Otherwise, I'd set all my traffic to high priority.
00:28.01opus__pc2 - comcast doesn't
00:28.12*** join/#asterisk tehdely (n=delysiid@home.teambarry.org)
00:28.16tehdelyboy howdy
00:28.23lanceyopus__ but it works...
00:28.23opus__pc2 - i hope you don't run p2p software :)
00:28.36Juggiemother, well, how does a bash script receive command line parameters
00:28.36*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
00:28.37pc2opus__ - why?
00:28.40Juggiei dont know how it does it
00:28.44Ariel_can the openwrt use shorewall?
00:28.46Juggieit should be in there.
00:28.49jskcrpc2:  check out http://peertech.org/IpQosTricks?pollresults%5B108%5D=1 too
00:28.51Juggiephp receives arg1 arg2 and so on
00:29.04Mother_OK thanks...I'll investigate that way
00:29.08tehdelyquick dialplan question... let's say i have a specific context which maps to one trunk, and i want all numbers dialed in that context to dial to the equivalent number on the iax peer
00:29.11jskcrpc2:  use google cache for that one
00:29.14tehdelyhow do i describe the extensions
00:29.17tehdely_ ?
00:29.23*** join/#asterisk rigid (n=The@port-212-202-73-207.dynamic.qsc.de)
00:29.24tehdelyi don't want to use a prefix
00:29.25rigidre
00:29.38Ariel_you don't have to use prefixes
00:29.47tehdelywonderful.  what is the syntax if i don't?
00:30.17opus__hmmm
00:30.30Ariel_1NXXNXXXXXX,1,Dial(IAX2/Blah@ipaddress/${EXTEN})
00:30.33rigidwhat do i have to take into special account when setting up asterisk for use with a hardware sip-router? i read http://gentoo-wiki.com/HOWTO_Asterisk and wanna use asterisk to disallow certain prefixes...
00:30.44tehdelyisn't the '1' a prefix?
00:30.54tehdelyalso what if i want the number to be of arbitrary length
00:30.55Ariel_~docs
00:30.56jbotsomebody said docs was 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
00:31.06tehdelythe peer in question is FWD
00:31.07Ariel_X.
00:31.17rigidit all sounds straightforward, but i didn't find a doc concerning proxying
00:31.58Drukenok... does anyone know how the god damn auth work for sip ?
00:32.25pc2jskcr - I still don't think any upstream providers would honor the qos bit.
00:32.56Ariel_Druken, what do you mean?
00:33.24Ariel_rigid, sip proxy, stun servers? SER what are you trying to do?
00:33.30jskcrpc2 it does not matter
00:33.45jskcras long as the router thats connected to the upstream does
00:34.31opus__god damn it.
00:34.40rigidAriel_, now my sip-router logs into my providers sip.server.com ... i want to login at my server which then forwards everything to my provider. i want to have control over the numbers dialed (forbidding expensive dial-prefixes)
00:34.42Ariel_tos bits are ok for your own network but once out in on the internet it's a waste of time.
00:34.43opus__If I install openwrt, i need to setup dhcp tftp
00:35.01rigidAriel_, +it
00:35.13rigidAriel_, i want it to login
00:35.21opus__Ariel, then what would work?
00:35.21*** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net)
00:35.31jdv79can i gen a call from the cmdline?
00:35.34jdv79just for kicks:)
00:35.46jdv79the console is what i meant
00:35.47DrukenAriel_: i mean i have a sip provider, and they are telling me my shit's broken... (personally i think they are wrong) but anyways... they say incoming calls require authentication
00:36.02tzangerdid that blkbearnh guy get his stuff figured out?
00:36.21lanceyanyone any ideas how chan_sip works out the "fromuser" when none specified in sip.conf ?
00:36.22Ariel_Druken, what is the error your getting
00:36.27lanceyit should use the callerid, right?
00:36.28Ariel_use sip debug
00:36.35filelancey: correct
00:36.44lanceyfile: well it doesn't all the time
00:36.49Drukeni did use sip debug... he's telling me that my server has to auth to his, to receive calls...
00:36.51*** join/#asterisk bjohnson (n=bjohnson@i216-58-58-237.cybersurf.com)
00:36.56Drukendon't exactly make sence in my mind...
00:36.56Ariel_fromuser fromdomain is used more for broadworks setups
00:37.00sleepy_oneDruken, sip debug are Ariel said and are you registering with them ? for your incoming
00:37.01lanceyany hints?
00:37.07fileDruken: not usually...
00:37.07rigidAriel_, is asterisk not the right choice for this?
00:37.11lanceyor i should file a bug for this?
00:37.18tzangerI love the pig named bacon hahaha
00:37.23Ariel_rigid, I kinda don't know whatyou want to do?
00:37.29tzanger<-- wathcing varsity blues
00:37.34lanceywhen receiving call from another * box, it gets set to "Unknown"
00:37.41Ariel_rigid, but asterisk can setup accounts and do routes for dialing
00:37.47Drukensleepy_one: don't get anything on sip debug when i call the number, and yes, i'm registered with them
00:38.00sleepy_oneDruken, have you tried ethereal and filtering for your provider's IP ?
00:38.21sleepy_oneto see what packets are getting exchanged
00:38.24Drukenno... i can dial out them, i just don't get the calls
00:38.38Pete_Largoyou mean incoming calls Drunken?
00:38.42Ariel_is the router port forwarding port 5060 to your box Druken
00:38.42Drukenyeah
00:38.58Pete_Largoin sip.conf do you have them set up as peer/user/friend?
00:39.00rigidAriel_, using asterisk without a phone card... a hardware-sip router (with phone connected) should be proxied... so that illegal prefixes won't be dialed
00:39.01DrukenAriel_: you think i'd run a server behind a router?
00:39.12Ariel_Druken, I do
00:39.20sleepy_onetried sip show registry? iax2 show registry?
00:39.22lanceyfile:     -- Executing Set("IAX2/csvarna-3", "CALLERID(number)=086510510") in new stack
00:39.25Mother_I think it's definitely not passing the variable to the bash script
00:39.32Drukensip show registry shows it as registered
00:39.37lanceyand then : From: "Unknown" <sip:Unknown@x.x.x.x>;tag=as098e3719
00:39.50FuriousGeorgeif i have exten=> s,1,answer(2) in my inbound calls context for zap, what do i gotta do to make it check for fax?  just have a 2nd priority context for the fax?
00:39.57Pete_LargoDrunken, what is the 'type' in sip.conf for that provider?
00:40.05Ariel_wait(3)
00:40.17FuriousGeorgeexten=> fax,2,blah?
00:40.17DrukenAriel_: no... not behind a common firewall/router... 5060 is wide open
00:40.34Ariel_Druken, what service
00:40.42Ariel_FuriousGeorge, no
00:40.43DrukenPete_Largo: friend, and i have autocreatepeer=yes
00:40.54Ariel_all fax extension must start on 1
00:41.03DrukenAriel_: define service
00:41.04filelancey: it should set the user part of the SIP URI to that number, funky
00:41.16lanceyfile: funky, indeed
00:41.25lanceyand when using IAX2 phone to originate the call, it does
00:41.33Ariel_Druken, who is sending you the call?
00:41.35lanceywhen originating the call with another * box, it doesn't
00:41.36FuriousGeorgeAriel_: so it will automagically check for fax after an answer even if i dont tell it to beyond putting fax in the dialplan?
00:41.46lanceythat's why i assume some sort of bug
00:41.50DrukenAriel_: netfone.ca
00:41.50SarahEmmgaah
00:41.54SarahEmmanyone using polycoms with HEAD?
00:42.00sleepy_oneDruken: do you have the username and pass defined in sip.conf? or their public key if they use keys?
00:42.04Ariel_FuriousGeorge, fax is detected via zap and it goes the fax,1 extension
00:42.23Drukensleepy_one: yes of course
00:42.37sleepy_oneI know voicepulse and others use their public key for auth
00:42.40rigidAriel_, i have a hardware sip-router that registers with a sip.server.com ...
00:42.40rigidAriel_, the hardware router has not the option to control prefixes... it allows every number and dials it.
00:42.43rigidAriel_, maybe i can use asterisk to "control" prefixes and send some "cmd failed" to the hardware-sip router so that it gives a busy-tone...
00:42.48rigidunderstand?
00:42.50rigidjust simple
00:43.19Ariel_rigid, if the router does it ahead of your asterisk box and you have not control over it your stuck.
00:43.35Ariel_Remove vonage router exchange for real one
00:43.40lancey:)
00:44.05Ariel_not/no
00:44.07rigidAriel_, before my box? i don't have asterisk, yet...
00:44.08lanceyanyone knows something about this SIP URI weirdness, or i'm posting a bug?
00:44.21rigidAriel_, what do you suggest?
00:44.40Ariel_rigid, ebay
00:44.58rigidAriel_, so i can't control prefixes using linux?
00:45.03rigiddoubt that...
00:45.13Ariel_linux has nothing to do with a dial plan
00:45.47rigidAriel_, my provider seems to use sipX ... i could set it up (in theory) to fake the server, and forward legal numbers to the real sipX server...
00:45.57rigidi thought there was an easier solution
00:46.15rigidi also came along "partysip" but rarely documented
00:46.23Ariel_rigid, your trying to change your callerID
00:46.56rigidAriel_, no... i'm trying to prevent everyone from dialing expensive prefixes... 0900-xxx
00:47.01rigidAriel_, for example
00:47.28Ariel_rigid, your unit does not allow for a dial plan get rid of it get an asterisk box instead.
00:48.11rigidAriel_, so asterisk can not forward? i don't wanna buy isdn equipment... i wanna use 2 analog normal type of phones
00:48.22pc2I have 7 spa-2000 vonage boxes here that I got free after mail in rebate at staples :)
00:48.28justinunice
00:48.31justinucan you unlock them?
00:48.38Ariel_rigid, yes but it we are not on the same page
00:48.40pc2justinu - yp :)
00:48.43pc2justinu - yup :)
00:48.44rigidrigid, i wouldn't be here if i could simply change the equipment
00:48.44justinunice
00:48.45pc2justinu - did, hehe.
00:48.54pc2Don't know what I'm going to do with them (ebaY?)
00:48.56pc2I only need a few.
00:48.59pc2hook up the family I guess.
00:49.04justinuis there anyway to unlock my rtp300?
00:49.22pc2google :)
00:49.23pc2I don't know.
00:49.27pc2probably not
00:49.28pc2it's rare
00:49.32Ariel_rigid, I don't actually understand what your asking.
00:49.47Mother_well...good night
00:49.48rigidAriel_, you're fooling me...
00:50.10rigidwhat else can i tell?
00:50.15justinuoff for the night
00:50.15*** part/#asterisk Mother_ (n=Mother@93.Red-80-32-127.staticIP.rima-tde.net)
00:50.16justinulater
00:50.21*** join/#asterisk robertwaters (i=H2O@ppp-69-223-114-159.dsl.kntpin.ameritech.net)
00:50.28Ariel_rigid, you have a box that sends calls via the internet but you want to have your people not dial numbers then put the asterisk box between them
00:51.01rigidAriel_, i want asterisk to forward/manipulate/generate sip-packets... manipulate packets coming from a sip-router
00:51.07*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
00:51.10Ariel_rigid, your phones (Sip) ----- Asterisk ----- Your funnyrouter box ---- internet
00:51.25*** join/#asterisk wundaboy (n=asdf@67.189.30.47)
00:51.40rigidAriel_, hmm... can asterisk handle analog equipment and generate a dialtone and stuff?
00:51.46Ariel_yes
00:51.54Ariel_tdm400p
00:51.59Ariel_t1/pri cards
00:52.07rigidAriel_, hmm... how much are they?
00:52.19Ariel_that depends
00:52.28Ariel_how many ports
00:52.33rigidAriel_, at least 2
00:52.58rigidAriel_, can you tell me some brands or what is needed to distinguish them from cards that don't work?
00:53.11jskcrIf im using a 64 bit intel proccessor should I use k8
00:53.16Ariel_rigid, 2 analog ports get a sipura 2002 it's cheaper connect it to the asterisk box then an account form the asterisk to the box.
00:53.18jskcrin the makefile
00:53.49rigidAriel_, i have to buy new equipment... is it available? how much round about?
00:53.56rigidAriel_, the router is 80 euro
00:54.05rigidAriel_, more or less?
00:54.23Ariel_rigid, I am in the states but a sipura 2002 is about 70 dollars
00:54.42*** join/#asterisk robertwaters (i=H2O@ppp-66-72-73-200.dsl.kntpin.ameritech.net)
00:56.04*** part/#asterisk tehdely (n=delysiid@home.teambarry.org)
00:56.27robertwaterscan anyone help me find information about routing calls by incomeing trunk?
00:57.10Ariel_robertwaters, what type of trunk
00:57.20rigidAriel_, that's worth a try...
00:57.56lanceyok, here it is : http://bugs.digium.com/view.php?id=5405
00:58.04lanceyif someone has any ideas...
00:58.27robertwatersI am useing 2 x100p modems
00:59.19robertwaterswhat I am trying to do is have a diffrent menu based on what line they call
01:01.28jdv79how do you pickup a line, play some file in the background, and handle DTMF at the same time?
01:01.50Drukento make a call i need a user or peer entry ?
01:02.07jdv79http://sial.org/pbot/13502
01:02.11jdv79that should work, right?
01:02.40Pete_LargoDrunken, user
01:02.42Pete_Largono
01:02.43Pete_Largopeer
01:02.44Pete_Largosorry
01:02.48Pete_Largouser is to receive call
01:02.56Drukenhmm...
01:03.17Drukenseems when i put in the peer entry block, i loose my incoming calls
01:03.26Drukenwhen i remove it... i get my incoming calls
01:03.34Pete_Largorobertwaters, you can send the call to a different context in zaptel.conf
01:03.53Pete_LargoDrunken, you can have two entries for one provider, make on a peer and one a user
01:03.54jdv79it answers and plys the file fine but it doesn't catch my DTMF and switch
01:04.50lanceyisn't the "friend" like a peer + user, or i'm wrong?
01:04.53DrukenPete_Largo: i realize this... your missing what i'm saying... if i don't have a block for it... my incoming works...
01:04.54Pete_Largoor is it zapata.conf
01:05.05lanceyDruken: try type=friend
01:05.26*** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net)
01:05.56jdv79Set(TIMEOUT(response)=seconds)     ?
01:06.04jdv79what would a real world version of that look like?
01:06.26jdv79SetTimeout(5)   ?
01:06.28Pete_Largorobertwaters, you can send the call to a different context in zapata.conf - sorry for the mixup
01:06.53*** join/#asterisk robertwaters (i=H2O@ppp-66-72-73-200.dsl.kntpin.ameritech.net)
01:07.10Pete_Largorobertwaters, you can send the call to a different context in zapata.conf - sorry for the mixup
01:07.55Pete_Largolancey, you are right
01:08.02*** part/#asterisk lters (n=lters@mrtcdsl-034.mis.net)
01:08.09robertwatersthat is what I thought you could do but I must not be reading my zapata.conf right becouse I thought it only shows the one trunk
01:08.20*** join/#asterisk glm2k (n=GLM@rrcs-24-199-11-41.west.biz.rr.com)
01:08.28Ariel_robertwaters, just asign a different context to each cards setup
01:08.31Pete_Largoyou have 2 cards?
01:08.35robertwatersyeah
01:08.39Pete_Largoand they both work?
01:08.41robertwatersboth are answering the phone
01:08.48Ariel_card one channel=1 context pstn1
01:08.57Ariel_channel=2 context=pstn2
01:09.04robertwatersok
01:09.21DrukenPete_Largo: is that amazing or something? 2 cards?
01:09.24Ariel_Then put your rules to do what you want in the extensions.conf [pstn1] blah
01:09.29Ariel_[pstn2] blah
01:09.30Pete_Largono, just asking the question
01:09.32Pete_Largoass
01:10.06Ariel_Pete_Largo, it's hard to get to x100p to work but it's possible
01:10.19Ariel_to/two
01:10.23robertwatersI did not have a problem getting them to answer
01:10.25Pete_LargoIt was hard when I got my single one working
01:10.42robertwatersbut I think my config is not right as the way I look at it they show up as one card
01:10.51Ariel_robertwaters, yes but next comes echo problems and other things
01:10.53robertwatersbut then again I am an amagure
01:11.06robertwaterssorry amature
01:11.28robertwatersreally?  I had great quality out of them today
01:11.56Ariel_robertwaters, nice
01:12.56robertwatersI keep hearing they are bad but I am just playing for a house system
01:13.42Pete_Largorobertwaters, I'm not 100% sure, because I haven't done it, but I think if you just put something in zapata.conf like: channel => 1 \ context = first-line \ channel => 2 \ context=second-line that should do the trick...  I think
01:13.50jskcrx100p make sure your use aggressive supression
01:14.02Pete_Largowhere \ means a new line
01:14.44Ariel_robertwaters, pastebin.ca your zapata.conf and let us look at it. Then we can give you better pointers.
01:15.07Ariel_context goes above the channel=1
01:15.15Pete_Largogood thinking Ariel_
01:15.39Pete_Largowhoops, I new it was something like that though
01:16.04Ariel_Pete_Largo, is it raining on your side of the state
01:16.22Pete_Largonot that I'm aware of
01:16.44Pete_Largobut then again, I haven't moved from my perch for a few hours
01:16.47jdv79what is autofall through?
01:17.01jdv79i thought the t extension would get hit first
01:17.09*** join/#asterisk alrs (n=lars@dsl092-033-090.lax1.dsl.speakeasy.net)
01:17.13Ariel_t is for timeout
01:17.31jdv79wouldn't that get hit before a "auto fallthrough"?
01:17.41jdv79if set timeout response was set
01:17.46*** part/#asterisk joat (n=joat@laketaylor.org)
01:17.51Ariel_yes if it's in the context
01:18.10jdv79i see it execute the set timeout but not the t
01:18.14jdv79how do i debug that?
01:18.33Ariel_what does the cli say
01:18.36robertwatershey I have a question for you ariel
01:19.07Ariel_ok
01:19.09robertwaterswould the info your looking for in the zapata.conf be in the zapta-auto.conf
01:19.25Ariel_robertwaters, are you using amp
01:19.31robertwatersyes
01:19.45Ariel_ahh that is a different thing
01:19.59robertwatersis it possible to do what I am trying to do without killing amp?
01:20.40Ariel_yes but your going to have to do some editing in your extensions_custom.conf
01:21.00jdv79Ariel_, http://sial.org/pbot/13503
01:21.03Ariel_and also in the zapata-auto.conf
01:21.53robertwatersok any ideas of what to search for to find the information as to what I need to change?
01:22.08jdv79and the context is: http://sial.org/pbot/13504
01:22.14Ariel_jdv79, it's not finding the exten => t
01:22.36Ariel_robertwaters, #amportal
01:22.37robertwatersI dont mind looking it up myself and learning just been looking for a week and can not figure out what I am looking for
01:22.38jdv79it looks like its there?
01:23.19robertwatersok thank you very much for the information
01:23.28Ariel_exten => t, 1, Goto(vm-end)
01:23.33Ariel_is your problem
01:23.36jdv79i was just wondering about that:)
01:23.38jdv79thanks
01:23.42Ariel_Goto(vm-end,s,1)
01:24.53lanceybye guys
01:24.57jdv79i think i meant playback
01:25.03jdv79i dont knpw
01:30.43*** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net)
01:31.27groogshey is the person that runs asteriskgui.com (flobi?) here? I'm working on a very similar problem to that dialplan inserter
01:31.33Ariel_jdv79, your goto are not right
01:31.48jdv79why not?
01:31.58Ariel_your going to have to do more reading look at the one I put up for you goto(contex,s,1)
01:32.00jdv79the docs on voip-info seems to say they are
01:32.20jdv79aren't context and extension optional?
01:32.22FuriousGeorgeAriel_: i asked this question ans missed if you answered or not.   do u recall when we were talking about fax detection?  were you saying all i have to do is have the "fax" priority, and * will check calls to see if its a fax?
01:32.41jdv79http://voip-info.org/wiki/index.php?page=Asterisk+cmd+Goto
01:33.05FuriousGeorgeits just that obviously i have to answer it fo it to listen to the tone
01:33.20Ariel_FuriousGeorge, if your system detects a fax in the default context for that zap channel it needs to have an exten => fax,1,BLAH
01:33.21FuriousGeorgeso i figured i'd have to start with an exten=> s,1,answer
01:33.49Qwellhow good is the zap fax detection?
01:33.53Ariel_FuriousGeorge, yes you need to answer it but then 2,wait(3) on 2nd line
01:34.05Ariel_Qwell, seems to work fine for me.
01:34.16FuriousGeorgeso what do i start with?  exten=> fax,1,answer
01:34.30FuriousGeorgeexten => fax,2,wait(2)
01:34.32Ariel_FuriousGeorge, no fax exten is a fall through
01:35.26FuriousGeorgeso i answer on s, wait 2, and if it detects a fax it goes to exten=> fax,1
01:35.34groogsdial() has a failure goto priority (whatever its called) n+101... are there any other applications that do that?
01:35.36Ariel_FuriousGeorge, your channel context is it just for a fax or does it do other things
01:35.46FuriousGeorgeno its for incoming zap calls
01:36.04FuriousGeorgerings extensions then goes to vm, the ususal
01:37.02Ariel_FuriousGeorge, if you have fax detection set to incoming then you just need to have in your context a line for the exten => fax,1,Blah  but also ahead of it exten => s,1,answer exten => s,2,wait(3) exten=> s,3,Goto(what you want)
01:37.51FuriousGeorgeok i get it
01:38.01FuriousGeorgei dont necesarily need a goto with s,3 i assume
01:38.20Ariel_FuriousGeorge, well you need what you want it to do next
01:38.26jskcrwith a xeon should I still use PROC=k8
01:38.38FuriousGeorgeand answer(2) makes it wait two seconds to answer, not answer and wait right?
01:38.53Ariel_I have not used answer(2)
01:39.07Ariel_I always use the next line with wait(3)
01:39.57FuriousGeorgek
01:40.00FuriousGeorgethanks
01:40.20Ariel_FuriousGeorge, np
01:40.38jdv79is it possible that 2way audio could be off?
01:42.14jdv79wow, on the CLI it sets the timeout and kills the call in about the same second
01:42.19jdv79does that sound right?
01:42.38*** join/#asterisk xlogik (n=xlogik@pool-141-154-127-152.bos.east.verizon.net)
01:42.45hardwiremutilator: not to sound odd.. but can I go bowling too?
01:43.05jdv79maybe i shouldn't be using HEAD:)
01:48.15*** join/#asterisk vtsherwood (n=vtsherwo@cpe-24-210-51-17.columbus.res.rr.com)
01:48.30jdv79Ariel_, i added the context like you said and still same thing
01:48.41jskcrchan_zap.so: undefined symbol: ast_pickup_call  anyone know why im getting that with cvs head
01:49.03vtsherwoodhello all
01:52.11*** part/#asterisk robertwaters (i=H2O@ppp-66-72-73-200.dsl.kntpin.ameritech.net)
01:52.38JerJerjskcr:  have you been noload'ing modules in modules.conf ?
01:52.52*** join/#asterisk littleball (n=littleba@bb219-75-114-103.singnet.com.sg)
01:53.20jskcrJerJer this is on a xeon
01:53.30jskcrIt seems to be using the k8 proc
01:53.34littleballhi, who is using TE411P card?
01:53.34JerJerres_features.so is required  :(
01:54.00jskcrWARNING[13979]: loader.c:314 __load_resource: /usr/lib/asterisk/modules/res_features.so: undefined symbol: adsi_available
01:54.05jskcrthis is 1.2.0 beta
01:54.11vtsherwoodI was wondering if anyone might have information on a rather weird problem our *boxes are having. It's rather long so I won't write the whole thing to the group, but anyone who's interested in trying to help, I'll send you the story.
01:54.12JerJerjskcr:  because res_features.so is not loading
01:54.16littleballi got the following error during run ztcfg command :ZT_SPANCONFIG failed on span 1: No such device or address (6)
01:54.28jskcrJerJer I know that :) I dont know why :(
01:54.39JerJerhave you mucked with modules.conf ?
01:54.52jskcrits a sample install
01:55.05jskcrso not at all
01:56.10jskcrits a freshly emerged gentoo system
01:56.28Qwell866-378-1477 - anybody recognize this number?
01:56.35jskcrwith a clean compile of 1.2.0 beta
01:56.59jskcrHmm ill try killing the -m64 and proc=k8
01:57.12jskcrsince its not recognizing the xeon for being a intel
01:57.48jskcrbut it is 64 bit because of the em64t  processor
01:57.50*** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net)
01:58.52jdv79JerJer, any idea? - http://sial.org/pbot/13507
01:59.52vtsherwoodin case you want an advance idea of the problem, it involves incoming/outgoing sip channel limits, 480 "Temporarily Unavailable (Call Limit)" and being unable to make or receive calls for a long time even after the channels in use have returned to an acceptable number
02:01.34marc324unrelated--where is swap directory in linux
02:01.49*** join/#asterisk r0d3nt (i=r0d3nt@tinfoilhat.net)
02:02.54*** join/#asterisk protien (i=jjmtrev@203-173-26-187.dyn.iinet.net.au)
02:03.06protienis skype supported under asterisk the same way fwd is?
02:04.29denonno, skype is a hack
02:04.37denon</unofficial_response>
02:04.39ManxPowerprotien, If Sykpe used the same protocol as FWD (SIP) then it would be
02:04.55protienwhat protocol does skype use
02:04.59denonskype
02:05.02protienoh
02:05.10ManxPowerprotien, Their own protocol that they don't document.
02:05.15protienare there any other free sip providers other than fwd?
02:05.18marc324can someone tell where the .swp are located?
02:05.23*** join/#asterisk ronaldl79 (n=chatzill@c-24-8-54-203.hsd1.co.comcast.net)
02:05.23mmlj4hey ManxPower
02:05.25ManxPower(Hmm...sounds like IAX, but at least there's souce for IAX stacks)
02:05.29ronaldl79Hey folks
02:05.39ronaldl79Just found out about OpenPBX.org. WTF?
02:05.40ManxPowerHello mmlj4
02:06.14ronaldl79Is anyone following this OpenPBX development?
02:06.19mmlj4what's wrong with sticking a wildcard in your box and plugging that into your skype ATA?
02:06.36ronaldl79It's a branch of Asterisk -- but is this really necessary?
02:06.44mmlj4ManxPower: using wireless?
02:07.09ManxPowerronaldl79, Remember by GMOME got started?  They were unhappy with the QT licensing?
02:07.31Lathos42Evening everyone
02:07.46ManxPowerronaldl79, same thing here.  OpenPBX.org is forking Asterisk because they don't like the idea of having to license to Digium any code they want put in the main Asterisk source tree
02:07.48ronaldl79Hmmm, I actually don't recall, Manx. I didn't take Linux seriously until about two years ago.
02:08.28ManxPowerThey seem to want to have Digium include the code in the main Asterisk source tree, but not license it to Digium.
02:08.50ManxPowermmlj4, If I didn't hate wireless before, I do now.
02:08.58mmlj4hee
02:08.59*** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net)
02:09.10AlexCTIHi..
02:09.20ManxPowerronaldl79, You would have to ask one of the OpenPBX people to get the "official story"
02:09.54ManxPowermmlj4, I had to replace all three of my 2.4Ghz cordless phones too
02:10.01mmlj4heh, nice
02:10.03AlexCTIAnyone knows how can I know if my Asterisk box is using "Silence Suppresion"
02:10.07mmlj4switching channels didn't help?
02:10.21ManxPowerAlexCTI, Asterisk NEVER EVER uses silence supression
02:10.21QwellAlexCTI: * doesn't do VAD
02:10.31ManxPowermmlj4, I have the higher end DSS phones.
02:10.39mmlj4channels 1, 6 and 11 seem to be immune to cordless interference
02:10.47mmlj4oh, hmm...
02:10.57ManxPowerI'm on channel 1 for my uplink and 6 for my local wirelss
02:11.13ManxPowerI got the 5.6Ghz phones
02:11.16vtsherwoodcan anyone help with a sip response code 480?
02:11.42mmlj4vtsherwood: i can tell you it's a server error
02:12.00vtsherwoodI understand, I know what it is, I have a rather complicated issue with it
02:12.01ManxPowerThe the guy that runs the "wireless ISP" is a radio guy with a mission to beat the telcos.  As far as I can tell, he can't route himself out of a paper bag
02:12.02AlexCTIManxpower, but the RTP Silence Suppression is not used on asterisk?
02:12.08vtsherwoodI have two servers running *
02:12.15mmlj4heh
02:12.18vtsherwoodand one gets calls from the other
02:12.26ManxPowerAlexCTI, RTP silence supression is not supported with Asterisk
02:12.45AlexCTIok, thanks.
02:12.51vtsherwoodbut when it's limit is reached on the SIP channels my main server sends a 480 back (temporaril unavailable)
02:12.54ManxPowerMy wireless link has been up and down more times today then Linta Lovelance
02:13.07ManxPowerLinda, that is
02:13.10Qwellmy god, worst IVR ever
02:13.13vtsherwoodand then the slaved server cannot make calls even after it's channel usage goes under the limit amount
02:13.18vtsherwoodany ideas?
02:13.43Qwell"Your call will be placed in queue for the next availablone moe reprmentesentativent please.
02:13.48groogsok with this "n priority" stuff, do you do eg:   exten => 1,MainDial+101,...   exten => 1,n,....     ?  or Maildial+102 ?
02:14.45vtsherwoodI'd give more info, but I don't think the _whole_ room wants to hear about it
02:14.50Ariel_groogs, in head you can just do n for next then n for the next statement
02:14.58ManxPowervtsherwood, what version of Asterisk?
02:15.05wunderkingroogs: you don't see the examples in extensions.conf?
02:15.16vtsherwoodCVS-HEAD from about two weeks ago
02:15.28wunderkinif you are running off an old one, check /usr/src/asterisk/configs/extensions.conf.sample
02:15.29groogswunderkin: hm guess i could look at that :p
02:15.34vtsherwoodI don't know  about the slave server, it's a customers
02:15.36ManxPowervtsherwood, update, if the problem persists, report it on bugs.digium.com
02:15.40jskcrhya Qwll
02:15.47vtsherwoodhrm...
02:15.50vtsherwoodok, I'll do that
02:15.58*** part/#asterisk vtsherwood (n=vtsherwo@cpe-24-210-51-17.columbus.res.rr.com)
02:16.00ManxPowervtsherwood, you ARE on the asterisk-cvs mailing list, right?
02:16.04Qwellanybody happen to have an escallation number at sprint (pcs if possible)?
02:16.20Qwell-l
02:16.23protienis there any other totally free sip services like fwd?
02:16.47ManxPowerprotien, If it's free, it's not worth much
02:17.01protieni know manx, but im playing learning asterisk
02:17.05protienim just trying to play arounds
02:18.11Ariel_Proteque, sipphone, stanaphone are sip service out there
02:18.23Qwelloh shit, $68.88 additional usage on my sprint account...thats awesome
02:18.28Ariel_protien, not Proteque
02:18.47protienthanks ariel ill check them out
02:18.54QwellAnytime Minutes  - $0.50/min43.0 $21.50
02:19.28*** join/#asterisk AJ-Mpls (i=DJAJay@63.231.252.9)
02:19.56AJ-Mplswhat config file hold the PORT info that asterisk listens to?
02:20.09*** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net)
02:22.12jdv79is head not working for anyone?
02:22.20*** part/#asterisk AJ-Mpls (i=DJAJay@63.231.252.9)
02:22.33Ariel_jdv79, I don't use head I use stable
02:23.11mmlj4what could cause voicemailmain to not see keypresses?       -- No username but # key pressed. Using CID '2076' /    -- Playing 'vm-password' (language 'en') /    -- Incorrect password '' for user '2076' (context = <any>)
02:23.32jdv79ok
02:23.39Daminmmlj4: Because you have the wrong DTMF mode setup for your device?
02:23.45jdv791.0.9?
02:23.49mmlj4hmm...
02:23.59mmlj4but the thing dials out
02:24.28ManxPowermmlj4, SIP devices collect the digits, then send them all at once in a SIP message.
02:24.32mmlj4actually, no device is able to send DTMF to voicemail anymore, dunno what I broke
02:24.47ManxPowermmlj4, dtmfmode=rfc2833
02:24.53mmlj4ManxPower: so how to actually sent them?
02:24.59mmlj4ah, well, lemme try that
02:25.04ManxPowermmlj4, it's sent as a call setup message.
02:25.04*** join/#asterisk mcf3782 (n=mcf3782@adsl-19-83-37.asm.bellsouth.net)
02:26.30Ariel_jdv79, yes I only use stable for my production systems.
02:28.13jdv79what's the rev for that?  i'm only familiar with svn:(
02:29.08jdv79cvs update -r v1-0-9zaptel libpri asterisk   ?
02:29.35ManxPowerjdv79, cvs update -r v1-0 blah blah blah
02:29.44jdv79thanks
02:32.51mmlj4ok, that fixed it, thanks
02:33.57Qwellanybody wanna do me a huge favor?
02:34.07QwellI need like 5-6 8xx numbers called, and tell me what they are...
02:36.10denonhmm .. I would, except I know when people want that, there's a reason they dont want to do it themselves :)
02:38.14*** part/#asterisk obsidian-studios (n=obsidian@c-66-177-188-197.hsd1.fl.comcast.net)
02:42.19Qwellnevermind
02:45.02*** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
02:45.46DaminSo..
02:45.52DaminWho is going to Astricon?
02:47.04protienhm
02:47.18protienim trying to connect to this stanaphone service, i can register okay
02:47.21protienbut when i try to make a call
02:47.22protienOct  7 12:02:14 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"pro" <sip:1001@203.173.26.187>;tag=as03301ca3'
02:48.21*** part/#asterisk msw (n=msw@rdu-nat.rpath.com)
02:48.29*** join/#asterisk msw (n=msw@rdu-nat.rpath.com)
02:56.03jdv79how does one record a session?
03:08.19*** join/#asterisk n0where (n=kc@d254204.ppp.asahi-net.or.jp)
03:09.57*** join/#asterisk n3u7 (n=neutrin0@69.197.165.118)
03:10.55jskcryay I found my gentoo asterisk problem :)
03:11.22ronaldl79Has anyone tried GNU's PBX? (the name slips my memory currently)
03:12.32hypa7iaronaldl79: yate?
03:13.04ronaldl79Is that it? I'm not sure, I thought it was something else.
03:14.40*** join/#asterisk AlexCTI (n=alex@weston-69.65.86.197.myacc.net)
03:16.08marc324<PROTECTED>
03:16.08marc324<PROTECTED>
03:16.08marc324<PROTECTED>
03:19.36*** join/#asterisk Corydon76-home (i=mauve@pdpc/supporter/sustaining/Corydon76-home)
03:20.52*** join/#asterisk luke-jr_ (n=luke-jr@CPE-65-26-133-171.kc.res.rr.com)
03:22.57hypa7iaronaldl79: there are a couple
03:23.32ronaldl79Which ones? I think there's one which starts with a B, but damn, the name again escapes me.
03:24.02ronaldl79And, hypa, have you tried them? If so, how do they compare to *? I've considered testing other open source PBXs, but why bother? * works just fine, and it's extremely powerful.
03:25.38hypa7iaronaldl79: SER is used a lot and is powerful in some ways * is not, from what i hear.  but i've only used asterisk
03:26.32JamesDotComser is not a pbx
03:26.42ronaldl79SER is definitely something I want to checkout for SIP messaging
03:26.54JamesDotComser rules for a sip proxy
03:27.16file[laptop]yes, yes it does
03:27.27jskcrser 250,000+ on a dual xeon
03:27.31jskcrthat rules
03:27.39hypa7iawhoa :-)
03:28.20jskcrIve personally benchmarked 60,000 on a ser system with only a 20% load
03:28.26jskcrthat was a p4
03:28.29ronaldl79Cool
03:28.35file[laptop]straight proxying is fine
03:28.38ronaldl79I need to learn more about SIP proxies.
03:28.39file[laptop]as it's not that CPU intensive
03:28.55*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
03:29.19hypa7iaronaldl79: http://www.voip-info.org/wiki-Open+Source+VOIP+Software
03:29.48hypa7iado any of you know about the borderware sipproxy / "firewall" thing?
03:29.51marc324<jskcr> that was a p4
03:29.51marc324<ronaldl79> Cool
03:29.51marc324<file[laptop]> straight proxying is fine
03:29.51marc324<ronaldl79> I need to learn more about SIP proxies.
03:29.51marc324<file[laptop]> as it's not that CPU intensive
03:29.52marc324* Ariel_ has quit IRC (Read error: 104 (Connection reset by peer))
03:29.54marc324* Ariel_ has joined #asterisk
03:30.19file[laptop]...yeah
03:30.26*** join/#asterisk jero (n=sflphone@savoirfairelinux.net)
03:31.19jdv79anyone know if legacy carriers block DTMF?
03:31.26jdv79i'm getting some weird stuff here
03:31.32jskcrnoooo
03:31.36jskcrnever
03:31.41jdv79huh
03:32.05jskcrinband signaling, is never blocked
03:32.24jdv79tht's what i though but my buddy is trying to tell me different
03:32.36jskcrwell hes wrong
03:33.09jskcrlegacy systems inheritly encode everything at twice the bandwidth of human speech around 3500hz
03:33.22jskcrdtmf is within that range
03:34.55jskcr697hz,1209hz for 1
03:35.32jskcrdtmf is in the middle land of human speech
03:36.35jdv79i think his CLASS4 swtich is being a little more intelligent than it ought to be:)
03:38.02jskcrin that case the  dtmf-mute-encoder may be active
03:38.21jskcrits a picnic problem in that case
03:38.28jdv79i'm getting really short blips and inconsistent timings when i monitor
03:38.43jdv79what is the time it takes for asterisk to recognize a tone?
03:39.03jdv791/4 second?
03:39.04*** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net)
03:39.07jskcrat least 40ms
03:39.21jdv79i think that much is getting through...
03:39.22jdv79weird
03:41.05marc324why do I get this: Starting Zap/1-1 at test,s,1 failed so falling back to exten 's'
03:41.44jdv79i think its getting through but its definitely being mucked with
03:44.47*** join/#asterisk santiago (n=santiago@208.195.215.231)
03:48.38Sedoroxanyone read/try the howto on unlocking the Linksys PAP2?
03:49.41jdv79asterisk do h323?
03:49.51jdv79cause that CLASS4 swtich is just gwing for me to SIP...
03:51.56*** join/#asterisk bmg505 (n=leon@rndf-146-32-157.telkomadsl.co.za)
03:55.05protienexten => 2002,1,SetCallerId("" <3477274087@sip.stanaphone.com>|a)
03:55.10protienis this syntax incorrect
03:55.19file[laptop]uh no
03:55.41protienhm
03:55.41protienOct  7 13:21:53 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@203.173.26.187>;tag=as55e58ffe'
03:55.48protienstill showing as asterisk@
03:55.55file[laptop]you create an entry in sip.conf, a peer entry
03:55.58file[laptop]with username and fromuser set
03:56.17protiendone that
03:56.38file[laptop]and you're using that to dial out?
03:56.42protienyeah
03:56.46*** part/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net)
03:56.50file[laptop]funky
03:57.30file[laptop]I'd look into it further except I'm literally in bed typing sideways
03:57.36file[laptop]because that's how big a geek I am
03:58.10marc324i try to use asterisk to answer a call using x100p.  i get this:   Starting Zap/1-1 at test,s,1 failed so falling back to exten 's'
03:58.31protienhm
03:58.40protienam i setting the caller id right
03:58.49protienbecause when i dont have the @sip.stanaphone.com
03:58.58file[laptop]you shouldn't have the @sip.stanaphone.com part...
03:59.01protienit identifies me as 3477274087@ip
03:59.06protienwell how can i set
03:59.09protiensip.stanaphone.com
03:59.12file[laptop]you can't set callerid if you're using fromuser
03:59.13jskcrSedorox:  you can unlock th pap2's
03:59.15protienbecause if i dont have that
03:59.27protieni get errors
04:00.05SedoroxI'm reading the article on telephreak.com (I think)
04:00.05file[laptop]protien: have you tried searching for existing working configs?
04:00.06Sedoroxabout how to do
04:00.07Sedoroxit
04:00.13protienyeah
04:00.25protienim using one thats ment o be working
04:00.30protienthis seems to be a regular problem
04:00.44protienOct  7 13:27:41 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"3477274087" <sip:3477274087@203.173.26.187>;tag=as159a8c3c'
04:01.19file[laptop]marc324: it's trying to start in the context test, on extension s, first priority
04:06.13digimeanyone know how to set the ToS bit in Windows for a softphone?
04:06.47marc324the x100p is a fxsks module?
04:07.05marc324should i use modprobe fxsks or modprobe fxoks?
04:07.35*** join/#asterisk mog_home (n=mogorman@user-24-236-84-48.knology.net)
04:07.39*** join/#asterisk jeanmi__ (n=besnard@front.tekkno.net)
04:08.17jeanmi__I would like to test a SIP phone (soft). Is there any echo server I could use for that purpose ?
04:08.31Sedoroxlink2voip doesn't do tollfree anymore huh?
04:09.43jeanmi__and also, could someone recommend me SIP provider with a kind of "pay as you go" offer ?
04:10.19Sedoroxjeanmi__, there is a ton.... check out voip-info.org... they have good info on providers
04:10.51*** join/#asterisk krisguy (n=krisguy@h216-170-039-057.adsl.navix.net)
04:17.32protienOct  7 13:40:09 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"3477274087" <sip:3477274087@203.173.26.187>;tag=as24ac16ba'
04:17.42protienhow can i cahnge that to 3477274087@sip.stanafone.com
04:17.53protieninstead of sip@ip
04:23.24jlewisanyone here going to astricon/been to that area before?
04:23.36Qwelljlewis: I live there
04:23.47Qwellwell...practically
04:24.02Qwellon Harbor?  probably a bit
04:24.37jlewiswhats weather out there been like?
04:24.50Qwellhot
04:24.58Qwellit should cool down by then though
04:26.06jlewistrying to figure out if I should rent a car, or just take the airport shuttle from/to the airport and be stuck at the hotel for the week
04:28.39jlewisis it safe to assume there'll be 802.11b internet at Astricon?
04:29.14mog_homewell
04:29.16mog_homelast year
04:29.20mog_homethere was no bandwith
04:29.26mog_homeas we only had a t1 for everyone
04:29.32mog_homethis year i imagine it will be better
04:29.52twistedno bandwidth was an understatement
04:30.20mog_homehey i was able to co asterisk 1.0 from hotel
04:30.25mog_homeso there was bandwith
04:30.51jlewisIIRC, the hotel itself provides internet to guests
04:30.52twistedyou said there wasn't, ninny
04:31.24mog_homei mean there was bandwith
04:31.27mog_homebut the latency
04:31.30mog_homeoh the latency
04:31.38twistednah
04:31.39twistedlatency was fine
04:31.44twistedit was congested
04:32.07jlewisyeah...they claim highspeed wired and wireless...is this the same hotel as last year?
04:32.18twistedjlewis, no, wrong side of the US
04:32.22jlewisah
04:32.33jlewisso last year it was on my side?
04:32.37mog_homeyeah
04:32.40mog_homeatlanta
04:32.50twisteddriving distance for me :)
04:32.53*** join/#asterisk KaBewM (n=kabewm@66-215-7-106.dhcp.psdn.ca.charter.com)
04:32.55jlewisgeez...that would have been so much more convenient
04:33.08twistedthis year it's a 5 your plane ride
04:33.11twisteds/your/hour
04:33.15jlewisa few hours drive rather than a day of flying
04:33.21mog_homeas all digium could come
04:33.53twistedi'm glad it is where it is
04:34.02twistedanother city i haven't visited yet that i've wanted to
04:34.25twistedall the flights?
04:34.30jlewisfrom FL to CA
04:34.49twistedah, which end of the phallus are you on?
04:35.02jlewisthe layovers, the missed flights, the getting stuck places
04:35.14marc324I have a x100p... do i use modprobe wcfxo or modprobe wcfxs?
04:35.18jlewisneither...central FL
04:35.23mog_homewcfxo
04:35.28twistedah
04:35.51twistedjlews == jerry lewis?
04:35.55twistedas in jerry lewis kids?
04:35.58jlewisand then run fxsks signaling
04:36.02jlewistwisted: no
04:36.05twisted:P
04:36.09twistedsorry, couldn't resist
04:36.43twistedso lessee
04:36.56twistedtomorrow i work, get home, relax.
04:37.10twistedsaturday - clean apt, do laundry, start adding new tracks to ipod
04:37.28twistedsunday - finish cleaning - pack up for trip, sleep lots.
04:37.38marc324I get this with x100p      Channel 'Zap/1-1' sent into invalid extension 's' in context 'default', but no invalid handler
04:37.51twistedmonday - work from 9 till 2, go to airport, leave for LA
04:37.53twistedyay
04:38.35marc324I have this in extensions.conf --------[incoming]
04:38.35marc324exten => s,1,Answer()
04:38.35marc324exten => s,2,Playback(hello-world)
04:38.35marc324exten => s,3,Hangup()
04:38.49twistedmarc324, your problem is quite obvious
04:39.11twistedi imagine if you look a little closer you'll see it
04:39.29mog_homeent into invalid extension 's' in context 'DEFAULT',
04:39.35mog_homethere is a hint ^_^
04:40.17twisted<PROTECTED>
04:40.39twistedirc needs a control character for italics
04:41.37twistedwe have bold, inverse, mIRC colors (*choke*gag*), and actions
04:41.52twistedirc is quite civilized.
04:42.00mog_homeits stone age
04:42.02mog_homeheh
04:42.05marc324alright. that worked. thx
04:42.14mog_homei would like jabber or something but people are here
04:42.18mog_homeand thus im here
04:42.44twistedmog_home, irc is the grandfather of protocols like jabber, aim, and the like.
04:42.52mog_homeyeah
04:43.02mog_homebut i dont use minix
04:43.03twistedshow respect to your elders.
04:43.04mog_homei use linux
04:43.23mog_homei dont use a mac classic i use my mini
04:43.24mog_homeetc
04:43.25mog_homeetc
04:43.27mog_homethe old
04:43.29mog_homethey die
04:43.32mog_homeit happens
04:43.36mog_homeexcept irc
04:43.44twistedirc is in a nursing home on life support
04:44.29mog_homeirc is opensource life support
04:44.33twistedand will probably remain that way until the powers that be decide to let the plug be pulled
04:44.34mog_homeor the freenode
04:44.48twistedoh wait, that was schaivo..
04:44.54mog_homeheh
04:48.15marc324are there additional gsm files that can be downloaded?
04:48.48*** join/#asterisk mut (i=WebChat@i.think.napoleon.dynamiteblows.com)
04:48.52muti need to buy a laptop right now.. where to do.. needs to be financed
04:50.19mog_homeasterisk-sounds
04:50.24mog_homehas more
04:53.16*** join/#asterisk viLeR (i=1000@66.128.47.232)
04:56.40*** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net)
05:01.49*** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-32-45.neo.res.rr.com)
05:02.09sleepy_onehello everyone, what's up?
05:02.34sleepy_onebeta1 is out already????
05:07.09wunderkinalready?
05:07.30wunderkinwhere have you been sleepyhead
05:08.21sleepy_onebeen sleeping
05:08.38wunderkinah haha didnt even look at your nick
05:08.51wunderkinwow i wonder where you got your nick from
05:09.15sleepy_onelol
05:09.18marc324is it possible to listen to the list of availablbe gsm over the net?
05:09.29sleepy_onewhat do u mean?
05:09.34mog_homejust co it marc
05:09.37mog_homeits not that big
05:10.00sleepy_oneyou can Playback any gsm
05:10.14sleepy_onewith Playback(filename)
05:10.54sleepy_oneor cp them to the vm directory, pretend they are voicemails and use the voicemail system to navigate
05:11.01marc324i need: "enter extension" sound file
05:11.09sleepy_oneohhhhh
05:11.23sleepy_onecvs co asterisk-sounds # I believe
05:11.49sleepy_onewget http://ftp.digium.com/pub/asterisk/asterisk-sounds-1.0.9.tar.gz # here's the direct link
05:12.59sleepy_onethe default sounds reside in /var/lib/asterisk/sounds IIRC
05:13.06mog_homethere are more sounds in cvs
05:13.12mog_homethan in web
05:13.13*** join/#asterisk tartar (n=tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
05:18.00marc324how do youplay gsm file in xp?
05:18.26sleepy_onewinamp ? broken player
05:18.33sleepy_oneerrrr I mean media player
05:18.37denonmarc324: download the j2 viewer or WavePad
05:18.43*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:20.01sleepy_onewinamp and or broken player should play them if you have the right libraries / codecs / etc
05:21.35Dr_Rayif you search google for winamp gsm, you'll find a solution
05:23.58marc324how can you playback digits, like 905
05:26.17sleepy_onePlayback(nine) Playback(zero) Playback(five)
05:26.36sleepy_oneIIRC they are in the digits subdirectory
05:27.53*** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal)
05:29.22sleepy_onemy bad Playback(9) Playback(0) Playback(5)
05:29.37sleepy_onels -al /var/lib/asterisk/sounds/digits
05:31.58sleepy_oneexten => 666,1,Playback(digits/9)
05:31.59sleepy_oneexten => 666,2,Playback(digits/0)
05:32.01sleepy_oneexten => 666,3,Playback(digits/5)
05:32.05sleepy_onehere's a real working example
05:32.16sleepy_onevery basic but it works
05:34.35sleepy_onefor your convenience here's the relevant wiki page http://www.voip-info.org/wiki-Asterisk+cmd+Playback
05:35.48sleepy_oneI would also urge you to read up on Background() and ControlPlayback()
05:36.36sleepy_oneBackground will allow you to play a sound file and still be able to process DTMF etc
05:38.21*** join/#asterisk tsp (n=tyler@S01060013102699ac.vc.shawcable.net)
05:38.30tsphow can I quickly and easilly set asterisk up as a sip server so I can use it to talk to my friends?
05:38.50shimitsp: about 5 minutes if you use Asterisk@Home
05:39.25*** join/#asterisk shido6 (n=curtis@d221-68-210.commercial.cgocable.net)
05:39.33shimitsp: note, however, that the installation CD formats whatever you have on the computer you run it on ;)
05:39.43tspouch
05:39.52marc324i found the function SayDigits()
05:39.57tspI just need a chat client to talk to my firends but i'm stuck with console only
05:40.07marc324it works.
05:40.17shimiI think there's an installer there for existing living machines, however I did not try it. The full OS (running CentOS3), works great.
05:40.36sleepy_onemarc324, yes that too :-)
05:40.38tspwhat's a good console only chat client for voip?
05:41.10shimiI'm afraid i am not sure ehow is asterisk related to your client being console or not? :)
05:41.40sleepy_onetsp, are you looking for a text messanging program over SIP????
05:41.58tspnope, just voice
05:42.03tspI"m blind so use console with speakup
05:42.15sleepy_oneyou mean a command line SIP softphone?
05:43.35sleepy_oneit sounds like you may be looking for a commandline capable SIP / VoIP softphone program
05:43.39marc324what directory would be appropriate for the downloaded gsm files?
05:43.50marc324where should it go?
05:43.59sleepy_one<PROTECTED>
05:44.11tspyeah
05:44.15tspminisip won't work
05:44.21tsponly one i've found
05:44.35tspI can use speexenc | ssh speexdec but that lags like hell
05:44.36*** join/#asterisk implicit (n=implicit@ip70-181-114-97.oc.oc.cox.net)
05:46.03sleepy_onetsp, hmmmm well gnophone linphone and kphone are open source - I don' sure if they have a console mode
05:46.03marc324there is a 1.0.9 sound file and a 1.2.0 -- what is the difference?
05:46.15sleepy_onemarc324, not sure
05:47.00sleepy_onetsp, hmmmm well gnophone linphone and kphone are open source - I'm not sure if they have a console mode but I'm sure one can be added or maybe run them in debug mode
05:47.53sleepy_onemarc324, like mog_home said there are more sounds in CVS
05:48.06marc324cvs?
05:48.28marc324how can i download that to xp?
05:48.35sleepy_one<PROTECTED>
05:48.37sleepy_one# export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot
05:48.38sleepy_one# cvs login - the password is anoncvs.
05:48.40sleepy_one# cvs checkout
05:48.43sleepy_onecvs checkout asterisk-sounds
05:48.55shimiI think "Cornfed" supports CLI mode
05:49.23sleepy_oneyou can use CVS for win32 or native CVS under cygwin running under XP
05:49.43sleepy_onehttp://cygwin.com/setup.exe
05:51.10sleepy_onemarc324, if you install cygwin under XP you will be able to use CVS, make, gcc/g++, etc etc
05:51.27sleepy_oneand many unix/ GNU/Linux *BSD apps
05:53.05sleepy_onetsp, http://freshmeat.net/projects/cornfedsipua/          http://freshmeat.net/redir/cornfedsipua/48563/url_rpm/cornfedsipua-0.9.5-1.i386.rpm
05:55.40sleepy_oneasterisk itself allows making phone calls from the asterisk console using your sound card but it can be tricky to setup sometimes
06:04.01*** join/#asterisk _Thor (i=Christia@user-vc8fl7l.biz.mindspring.com)
06:05.45*** join/#asterisk Tili (i=Tili@202-133-67-57-dialup.sat.net.pk)
06:06.25*** join/#asterisk viLeR (i=1000@66.128.47.232)
06:09.01*** join/#asterisk djin_ib (n=djin_ib@gridfox.xs4all.nl)
06:09.18tspgoodnight
06:09.20tspthanks all
06:09.42marc324how do you disable the overwrite prompt when using cp
06:10.02djin_ibdoing a simple 'make install' on asterisk sound v1.2b1:
06:10.05djin_ibNo description for sounds/access-code.gsm
06:10.10djin_ibWhat's wrong ?!?
06:11.02_Thorhi everybody... Anyone using the PrepaidCall application?
06:11.17sleepy_onethere shouldn't be anything wrong, I think that might be a warning not a fatal error
06:12.16djin_ibsleepy_one, I think it's a fatal. The /var/lib/asterisk/sounds is quite empty
06:12.53djin_ibor can I just copy the sounds/ to /var/lib/asterisk ?
06:13.01sleepy_onesure that would work
06:13.13sleepy_onethere's isn't anything that needs installed really IIRC
06:13.46djin_ibThat's what I thought. I looks just like a copy process
06:13.52_Thor<PROTECTED>
06:14.07sleepy_onebrb
06:14.46*** join/#asterisk MrB0B0 (n=bobo@203.94.141.98)
06:14.59Dr_Rayn3u7 - it's a good book
06:15.18MrB0B0anyone feel like lending a hand re: dundi
06:15.43n3u7Dr_Ray:ya I'm pretty excited about this
06:16.08Dr_RayI read it cover to cover, the day I got it
06:16.17Dr_Rayexcept for the isp stuff.. zzz. ")
06:16.21Dr_Rayer, sip
06:16.49n3u7well I just poured myself another coffee and I'm planning on putting a pretty big dent in it tonight
06:17.03n3u7i havean x100p card so far
06:17.05djin_ibDR-Ray, n3u7, what book is that?
06:17.10Dr_Ray~docs
06:17.12jbotdocs is, like, 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
06:17.22Dr_Raythe last link
06:17.22*** join/#asterisk websae (i=websae@207-118-134-96.dyn.centurytel.net)
06:17.26n3u7djin_ib:its the Oreilly Asterisk book
06:17.46djin_ibOk, I've seen that one.
06:17.57djin_ibDoes it cover stuff supported in 1.2?
06:18.02Dr_Raysome
06:18.08websaei am wondering...does anyone here know of company that builds voip phones that if you're a VoIP company, you can provide your users with these phones and update them remotely from a server?
06:18.25Dr_Rayit makes mention of add n, instead of 1,2,3,4 in dialplans
06:18.30djin_ibwebsae, there're many
06:18.34denonwebsae: that'd be .. any voip phone, almost ..
06:18.43denonwebsae: they almost all support tftp provisioning
06:19.01websaeim talking about software package that helps deploy those settings
06:19.04djin_ibwebsae, most support tftp, but SNOM has a http porvisioning as well.
06:19.14websaeSNOM?
06:19.22djin_ibyes.
06:19.34websaedjin_ib: what is that?
06:19.37Dr_RayI bet a cisco provisioning tool would not be that hard to write
06:19.43djin_ibhttp://www.snom.com
06:19.50djin_ibit's a brand
06:20.32websaeahh okay
06:20.53djin_ibDr_Ray, isn't that a bit limited (add n). v1.2 seems to introduce a lot more.
06:21.34MrB0B0has anyone got the dundi enterprise example from voip-info.org working?  I have two test boxes running and peering okay but all lookups return empty :$
06:21.52Dr_Raydjin_ib - perhaps, I'm a bit behind on asterisk.. I mainly use it for zaptel.. so I've not kept up.. for example I did not know that you could spy sip/iax channels now
06:22.01*** join/#asterisk Rowter (n=SilverDr@201.135.26.195)
06:22.11RowterTE205P has E1/T1 jumpers?
06:22.30Rowterasterisk is detecting TE205P as T1, and I need it as E1!
06:22.47djin_ibDr-Ray, it's never wrong to learn the basics (again). I'll order the book as well.
06:23.38Dr_RayThat's what it was for me, a review of the current state of it, Like I had never heard of call files, or the Directory() app.. I bought two copies, one for me to highlight.. and one for me to lend out.
06:24.05djin_ibThe TE210P supports E1, T1, and J1 environments and is selectable on a per-card or per-port basis.
06:24.15djin_ibIt doens't mention jumpers :)
06:24.17Dr_RayWhen I started with asterisk a year or so ago, I learned exactly enough to replace our mitel sx50
06:24.29marc324how do you create a delay between playbacks?
06:24.41djin_ibuuuu, Wait(1)
06:25.07Dr_Raythe person who made call files gets a gold star in my book
06:25.41djin_ibDr-Ray, same problem here. Learning by experience tends to limit your knowledge :)
06:26.05Rowterdjin_ib, yeah but its getting T1, cause Channel 24 is reserved for D-channel when I start *
06:26.51*** join/#asterisk argos73 (n=mike@65-85-207-101.client.dsl.net)
06:26.57djin_ibRowter, I have no TE2XX experiece, aren't there any jumpers on the board?
06:27.24Rowterdjin_ib, I didn't saw any, but I think I'll get it out again >)
06:27.47*** join/#asterisk sleepy_one (n=chatzill@cpe-24-166-32-45.neo.res.rr.com)
06:28.21argos73thought there was a t1/e1 jumper on it
06:28.27argos73(err, two of them)
06:28.38djin_ibRowster, there are jumpers on the TE100P, TE110P, TE410P, TE405P, so my guess TE2xxP has them as well ;)
06:29.58Rowterdjin_ib, ok let me see..
06:30.22websaewhat's a good billing software for asterisk
06:30.29websaeanyone have any ideas or experience?
06:30.56argos73dammit...  gonna need to upgrade our merlin legend to make it do what I want with asterisk...
06:33.07sleepy_onewhy not replace it with * instead ?
06:33.52sleepy_onewebsae, not sure but since * keeps great CDR records it's a piece of cake to sum up usage and invoice clients
06:34.38argos73would like to, but the cost of new phones (150+ of them) is too high right now...
06:35.19argos73easier to integrate the two, and switch out phones a few at a time
06:35.42sleepy_oneyes, I see
06:36.19sleepy_onewill you need a T1 card for your merlin ?
06:36.30argos73heh - just received it today
06:36.54sleepy_onemust have cost you a small fortune
06:37.05argos73nah - $675 or so
06:37.10argos73not too bad
06:37.17sleepy_onestill a good chunk of change
06:37.22sleepy_onenew or ebay?
06:37.30Rowterdjin_ib, yep it had a jumper >)
06:37.35Rowterhehe
06:37.36argos73refurb from our local dealer
06:37.59argos73might go ebay to get one for my legend system at home...
06:37.59sleepy_oneahh I see
06:38.05djin_ibRowter, ran into the same reminder with my TE410P yesterday ;)
06:38.32*** join/#asterisk ozant (n=ozan@85.96.199.40)
06:39.11sleepy_oneI'm sure a lot of vendors would love to offer you a discount on 150+ phones but I can understand that's going to be a huge expenditure :-(
06:40.02argos73to not lose any functionality, would prob go with cisco phones...  around $50K for all of them
06:40.27sleepy_oneplus the hidden costs of adding enough ethernet ports and POE unless you already have enough
06:41.05sleepy_onemost IP phones have a built-in switch but I wouldn't recommend it
06:41.14Rowterdjin_ib, got a diferent jumper for each.. excelent.. lets test now hehe..
06:41.34argos73plenty of ethernet ports (i went "wiring crazy" last summer), but poe isn't there yet
06:42.05sleepy_onehttp://www.voipsupply.com/product_info.php?products_id=505&ref=froogle
06:42.19argos73have six 24-port HP switches just waiting for traffic.. :)
06:42.25sleepy_onethey have 7960 refurbs for 294 each
06:42.47sleepy_oneI wouldn't recommend refurbs but I would recommend the 7960 it's a decent phone
06:42.47argos73hmm - not bad
06:42.59argos73have a 7960 here at home - love it
06:43.05Dr_RayI love my 7960
06:43.10Dr_Raythe smartnet contract blew
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06:43.35sleepy_oneI wish I had one too, I have an Iaxy with an analog phone instead
06:43.53sleepy_onenot bad for 100 bux
06:44.28argos73now the big question is "do I go ebay for the legend upgrade ($100, assuming it works) and risk alienating our hardware support contract, or do I get the upgrade from them for $1750?"
06:44.32Dr_RayI could probably get by with a 7940
06:44.34argos73decisions, decisions....
06:44.52sleepy_one1750???? geeeeez
06:45.10sleepy_one1750 = 10 snom phones IIRC
06:45.25argos73it's the old "cheap bastard vs responsible corporate guy" battle
06:45.41Dr_Raywell, if it's works money, I'd say spend it
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06:46.21sleepy_onethe snom 190 is NOT a cisco but it's a very good entry-level phone for the $$$ (about 180 each)
06:46.28argos73i'll probably blow the $100 myself for my legend system - if it works, i'll have work spend the major bucks for theirs
06:47.07websaeim curious...if someone has vonage...how can they make it so they can use all their existing phones in their house with the ATA adapter that comes with vonage service?
06:47.12argos73still trying to get my wife to appreciate the beauty of a 7960 in the living room.. :)
06:48.03argos73if you don't exceed the REN max of the ATA, you should be able to just split the line
06:48.31sleepy_oneya just plug your analog line into it and disconect it at the telco box
06:48.52sleepy_oneall the phones should work that way
06:48.58websaehmm
06:49.10websaeso at the telco box...
06:49.15websaei would just wire that in?
06:49.19websaethe ata adapter
06:49.25argos73yup
06:49.27sleepy_oneyup
06:49.32websaethat should work out alright?
06:49.35argos73just make sure the telco line is disconnected first...
06:49.43sleepy_onejust disconnect from the telco so you don't fry anything
06:49.50argos73didn't do that with my tdm400 fxs card - POOF!
06:49.52websaeis there a site with a tutorial on this?
06:49.54dersteerI use the old telco lines to charge batteries
06:49.55dersteer:)
06:50.19*** join/#asterisk Bonzai070 (n=pirch@wbs-146-129-162.telkomadsl.co.za)
06:50.29MrB0B0hi guys, I hate to be a pain in the *ss but... theres always a but isn't there... I'm so close to getting enterprise dundi going its driving me crazy
06:50.53MrB0B0any general pointers on failed lookups between otherwise happy peers?
06:51.07argos73websae - any site that describes how house phone wiring should be done will apply...  just replace "line coming from telco" with "line coming from ata"
06:51.08sleepy_onewebsae, label the wires in your telco terminal box and then disconnect them from the telco terminals
06:51.19*** join/#asterisk nesys (n=nesys@2001:1418:1a6:0:20d:93ff:fe28:3ef8)
06:51.31sleepy_onemake sure there's no power on the line
06:51.51sleepy_onethen put an RJ11 on the wires and plug that into your ATA from vonage
06:52.17argos73easiest way I can think of - disconnect telco line, then plug ata into a normal phone jack somewhere in your house
06:52.35dersteerthats how I did it
06:52.36sleepy_oneaye that would work too :-)
06:53.08argos73i firmly believe in the "stupid simple" way of doing things.  :)
06:53.08sleepy_onehowever some phones may prevent other phones from getting caller ID info and such
06:54.14argos73if your telco demarc is one of those big gray boxes they attach to the back of your house, just open the cover and unplug the rj11 inside it
06:54.15sleepy_oneI have tons of  tools and crimpers so I can put an RJ11, RJ45, RJ48 or RG6 connectors on just about anything ;-)
06:54.50argos73yes, but do you have a crimper for DB9/25 pins?  :)
06:54.56*** join/#asterisk jdiskywlkr (n=kvirc@ip68-0-83-251.tu.ok.cox.net)
06:55.06sleepy_oneyes actually
06:55.17argos73heh - one of my prized posessions
06:55.19sleepy_onehave an IDC crimper too
06:55.29sleepy_onecan crimp BNC too
06:55.40sleepy_onealthough I haven't done that in 10 years
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06:55.53argos73pull mine out occasionally for my radio work
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06:56.58*** part/#asterisk MuppetMaster (n=MuppetMa@62.57.152.193)
06:59.19sleepy_oneI misplaced the remote terminator for one of my LAN testers once so I made one with some CAT5e a breadboard and 8 LEDs
07:00.17argos73heh - cool
07:00.45denonsure, but will it tell you how far down the wire your break is?
07:00.57sleepy_onelol
07:01.04sleepy_oneno, of course not
07:01.19denonsounds to me like you wasted that breadboard ..
07:01.29sleepy_oneit wasn't permanent
07:01.29denona couple ICs and a power source, and you could have somethin
07:01.36argos73sure it will...  you just need to be really good at measuring the time between when you flip the switch and when the light comes on
07:01.36sleepy_oneaye
07:01.45sleepy_onelol
07:01.50denonargos73: if there's a break, the light wont light :)
07:02.24denonnow, you could slowly increase the voltage, until there's enough for it to arc over the gap .. and that might tell you something ;)
07:02.28sleepy_onethat's what my inductive amp and probe is for
07:02.32argos73hence the reason people have built "power off" lights into stuff.  :)
07:02.37denoncourse 480vac may not traverse cat5 too well
07:03.26argos73cya
07:03.31sleepy_oneIIRC CAT5 will only survive to about 120VDC MAX before it blows
07:04.05sleepy_oneat low amps
07:04.22sleepy_onenite denon
07:05.06sleepy_oneIIRC T1s use 100VDC and I run that over CAT5 all the time
07:05.09argos73sigh... one more day, and I'm outta that place for two weeks....
07:05.31sleepy_onevacation?
07:05.36argos73yup
07:05.50sleepy_oneoh good :-)
07:06.13argos73i'm sure they'll be calling me every two hours, though...  they always do.
07:06.26sleepy_onethe question is how will they survive without you?
07:06.35argos73usually, not well.
07:07.00sleepy_onebe happy 2 cover 4 u :-)
07:07.11*** join/#asterisk samyl (n=samyl@194.167.18.244)
07:07.51argos73absolutely LOVED it when they called my cell phone when I was in Florida about 2 years ago... "Help!  We're too dumb to follow your simple instructions!"  "Well, I'm on a boat in the middle of the Gulf of Mexico, and can't do a damn thing about it right now."
07:08.23samylHi
07:09.00sleepy_onethat's why I always have a laptop and one or more internet-capable phones with me when I leave the batcave (at least when I was able to afford the cell phones)
07:09.02*** join/#asterisk Uberbot (n=Uberbot@69.252.219.76)
07:09.37sleepy_onenaturally reception wouldn't be too good in the gulf
07:09.58argos73funny thing about it was that my laptop was back at the place I was staying - condo owned by my boss
07:10.07*** part/#asterisk Uberbot (n=Uberbot@69.252.219.76)
07:10.20argos73surprisingly, we were about 6 miles out, and the reception was still pretty good
07:10.35*** join/#asterisk Mw3 (i=mw3@national.t-error.hu)
07:10.39samylI've a problem with meetme
07:10.47samylwhen i try to use it
07:10.56samylmy asterisk server tell me
07:11.09samyl<PROTECTED>
07:11.14timecopheh
07:11.21samylUnable to open pseudo channel - trying device
07:11.24timecopbecause meetme needs a zap channel.
07:11.43sleepy_onemy old cell phone service sucks for voice but data coverage is pretty good coast to coast worked cross-country
07:11.48samylI know but i've zaptel module loaded
07:11.52samylwith ztdummy
07:12.03samylbut i don't have a digium card
07:12.18samyli use ztdummy as timer for meetme
07:12.35sleepy_onetry running ztcfg -vvvvvvvvvvvvvvvvvv
07:12.46sleepy_onemodprobe zaptel
07:12.52sleepy_onemodprobe ztdummy
07:12.57sleepy_oneztcfg -vvvvvvvvvvvvvv
07:13.00sleepy_onesafe_asterisk
07:13.15samylZaptel Configuration
07:13.15samyl======================
07:13.15samylChannel map:
07:13.15samylChannel 01: FXS Kewlstart (Default) (Slaves: 01)
07:13.15samyl1 channels configured.
07:13.16samylZT_CHANCONFIG failed on channel 1: No such device or address (6)
07:13.32sleepy_oneyou might have to run modprove ztdummy before zaptel I forget
07:13.55sleepy_oneis zaptel there when you lsmod ?
07:13.59samylyes
07:14.04sleepy_onehmmm
07:14.15sleepy_onedid you configure udev for zap?
07:14.16samylwcfxo                  12192  0
07:14.17samylztdummy                 3620  0
07:14.17samylzaptel                186756  2 wcfxo,ztdummy
07:14.22samylyes
07:14.30sleepy_onels -al /dev/zap
07:14.49samylcrw-rw----  1 root asterisk 196, 254 2005-10-06 12:59 channel
07:14.49samylcrw-rw----  1 root asterisk 196,   0 2005-10-06 12:59 ctl
07:14.49samylcrw-rw----  1 root asterisk 196, 255 2005-10-06 12:59 pseudo
07:14.49samylcrw-rw----  1 root asterisk 196, 253 2005-10-06 12:59 timer
07:15.28samyli must loaded zaptel before ztdummy?
07:15.40sleepy_onemaybe
07:15.45samyli can try
07:15.46sleepy_oneI don't remember exactly
07:15.51sleepy_onermmod and try
07:15.57sleepy_oneshouldn't hurt anything
07:16.02sleepy_onetry it both ways
07:16.31sleepy_oneoh wait a min
07:16.53sleepy_onechannel 01 doesn't exist because you don't have an FXS card
07:16.57sleepy_oneor FXO car
07:16.59sleepy_oned
07:17.00sleepy_oneright?
07:17.39samylyes, i don't have a digium card
07:17.51sleepy_onecomment that out in zaptel.conf and zapata.conf
07:18.00samyli use the zaptel driver for ztdummy
07:18.05samylfor meetme
07:18.09sleepy_onethen ztcfg -vvvvvvvvvvvv
07:18.12sleepy_onesafe_asterisk
07:18.19djin_ibbut i don't have a digium card <-> wcfxo  ??
07:18.45samylI've tried all
07:18.47sleepy_onejust comment out the channels in zaptel.conf and zapata.conf
07:18.50samylok
07:18.55sleepy_onermmod
07:19.05sleepy_onemodprobe zaptel and ztdummy again
07:19.08sleepy_oneztcfg -vvvvvvvvvvvvvvvv
07:19.11sleepy_onesafe_asterisk
07:19.11*** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no)
07:19.15sleepy_onesee what it does
07:20.10samylI'll try
07:20.15samylthank you
07:20.19sleepy_oneyw
07:20.23argos73hmmm..  this zyxel wifi sip phone isn't too bad, but the battery life sucks...
07:20.54argos73leave it turned on (idle) for 24 hours, and it's dead.
07:21.03sleepy_onedang :-(
07:22.01*** join/#asterisk gvag11 (n=g@84.254.12.236)
07:22.04sleepy_oneI guess you need to carry more than 1 battery or a charger with you
07:22.12sleepy_oneor both
07:22.45sleepy_onehave you seen those solar chargers?
07:23.08sleepy_oneor those hand-crank emergency radios that can charge cell phones ipods, etc ?
07:23.15argos73unfortunately, the designers didn't seem to think of that possibility..  the battery is a real pain to take out - battery cover is difficult to remove, and getting the battery out requires a backhoe.
07:23.42*** join/#asterisk Cresl1n (n=matt@user-24-236-124-147.knology.net)
07:23.52sleepy_onedoes it have charging contacts underneath ?
07:23.53argos73but it looks like the charger jack is one of those mini-usb connectors
07:24.14argos735vdc charger
07:24.34argos73maybe..  there's two rubber plugs on the bottom
07:24.38sleepy_oneI saw a hand-crank emergency radio / cell phone / battery / ipod charger / usb power thingy
07:24.53argos73heh - emergency ipod charger.  that's funny
07:26.10gvag11hi, does anybody have used Asterisk + Spandsp + TE110P (or any E1 board) for faxing? It works fine?
07:26.28Cresl1ngvag11: what does it do when you try it?
07:27.43gvag11Cresl1n : I am thinking to buy the equipment to implement such a solution and i wanted to know if it works and if yes how much reliable it is.
07:28.03Cresl1njust make sure you don't get your timing mixed up :-0
07:28.17argos73gvag11: ask me in a week, and i'll let you know how it went
07:28.56sleepy_onehttp://www.lillianvernon.com/catalog/product_display.jsp?pdId=6063
07:29.08sleepy_onesomething like this ^
07:29.09gvag11Cresl1n: when you say timing you mean to sychronise the E1 board correct?
07:29.28Cresl1ngvag11: yeppers
07:29.39*** join/#asterisk struct2 (n=struct@81-17-62-133.dsl.uwadslprovider.nl)
07:29.45gvag11argos73: so you are about to implement such a solution with E1/T1?
07:30.18struct2I have chicage Micro 330 ISDN Dect phones with a Chicaco Vox 390 Access point, i connected those dects with my octoBri cards
07:30.18gvag11cresl1n: so there are no slip frames, right?
07:30.26struct2i want to use 3 way call transfers
07:30.51argos73gvag11: going to try...  want to use PRI with DID to do per-department fax receiving
07:31.02struct2when i let the PHONE handle the 3 way call transfer, then it works nicely , i can recieve the call, connect to the 3th party, take the call back or hangup to transfer
07:31.16Cresl1ngvag11: exactly
07:31.36struct2Only thing when the PHONE handles the 3 wall call transfer it uses 2 channels, while these 2 channels are shared with 2 Phones, so one is out of service during transfer
07:32.11struct2When i let Asterisk handle the 3 way transfer, then asterisk places the call nicely on hold , and i can call the 3th party
07:32.16struct2only i cannot return the call
07:32.28argos73speaking of such, I know that fax over voip is flaky.. how about PRI -> Asterisk -> Channel bank -> Fax machine?
07:32.30gvag11cresl1n: i was thinking to have two PC with one TE110P each one connected back-to-beck (cpe-network config), do you think this is gone be a problem with sync?
07:32.33struct2i alway transfer the call regardless if i try to take it back or hang it up
07:32.48struct2Anyone an idea?
07:33.01Cresl1ngvag11: no, that should be ok
07:33.18gvag11cresl1n: Cool, thanks
07:33.30Cresl1nnp :-D
07:34.30gvag11cresl1n: Any idea by the way about the traffic load that TE110P can handle? Can i have 30 concurent channels in use?
07:34.51sleepy_oneyes if your CPU can handle the load
07:34.56argos73gvag11: if the host machine is up to it, shouldn't be a problem
07:35.48Cresl1ngvag11: sure, that's usually not too bad
07:35.55gvag11argos73: in other words if the CPU has power enough to handle these calls so there is no problem, right?
07:35.57sleepy_onea really nice server should be able to handle 4 Quad Span T1/E1 cards
07:36.03argos73gvag11: yup
07:36.08sleepy_oneyes correct
07:36.33sleepy_oneif you do not have to do serious transcoding you can support hundreds of channels on one server
07:36.51argos73gvag11: i have a Pentium 4 with two T1 channels - haven't seen any problems yet
07:37.52gvag11what about having more than one wildcard boards (lets say 2 TE110P) in one server, is this a good practice? Is it better to stick with one board in each server?
07:38.03sleepy_onefor example if you do ULAW across the board you should be able to handle a large number of channels, however if you have to go from ULAW or ALAW on the PRI to GSM or something that will be a lot more CPU intensive
07:39.01gvag11sleepy_one: Ok i got it...
07:40.02gvag11argos73: what about the traffic load, i mean you could have two T1 channels but there might be low traffic load. What happens when both of the T1 are full, did you ever test it?
07:40.03sleepy_oneas long as you don't run out of interrupts and such you should be fine I've been told you can put up to 4 quad cards in a compatible box
07:40.35sleepy_onehowever some motherboards cannot support that many cards
07:41.10argos73gvag11: number of cards usually doesn't matter as long as you're careful
07:41.14gvag11and for me to take care of interrupts i should take care the of the CPU and the motherboard, right?
07:41.21sleepy_oneand you have to make sure you buy the right version of the TExxx cards there's 5v and 3.3v versions
07:41.39sleepy_oneIIRC
07:41.39argos73gvag11: tested it at 75% busy without any complaints
07:41.41*** join/#asterisk \PsyKo\ (n=xxxxxxxx@golia.caltanet.it)
07:42.16sleepy_oneyes that's right
07:43.03sleepy_onefound a few boards that will not work with the T100p single span T1/E1 card if it is in the wrong PCI slot
07:43.09shimiTE110P supports both 3v and 5v
07:43.14argos73from my experience, intel motherboards (even the cheap ones) are pretty well-behaved...  some other brands can cause weird problems
07:43.34*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
07:44.19argos73(plus, intel boards usually have more PCI slots than most of the other brands)
07:44.20gvag11what about asus motherboards? Did someone knows any problem that can be caused by PCI express (since most of the motherboards comes with it now)
07:44.34sleepy_oneeven on asus boards which are very well respected I had PCI slots that didn't like the digium cards so the card hard to be moved
07:44.58sleepy_onehard = had
07:45.11sleepy_oneand the PCI slot couldn't be used
07:45.36gvag11sleeply_one: while on the Intel motherboard everything was fine?
07:45.47sleepy_onehaven't tried intel
07:46.16sleepy_oneI used several AMD boards some didn't have that problem
07:46.30argos73gvag11: have an intel board with two x100p cards, one tdm400 card, and one te100p card - no problems
07:46.36sleepy_onethe Via K8T800 based board had a problem
07:46.43sleepy_onethe nForce didn't IIRC
07:47.18*** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62)
07:47.30sleepy_oneit sometimes varies from board to board even if the chipset is the same
07:47.30nfi|ermeshi all
07:47.49argos73did have to disable a couple things build onto the board in the bios to free up some interrupts, but who cares about the dumb built-in sound card on an ast server...  :)
07:47.50sleepy_onesometimes the BIOS makes a huge difference
07:48.03gvag11i see... So most of the time you have to gamble and pray that it will work, right?
07:48.07sleepy_oneaye I did too
07:48.21sleepy_onemotherboards come with too much crap nowadays
07:48.55sleepy_onethere is no guarantee until you try it for yourself
07:49.32sleepy_oneunless you buy a system guaranteed to work by the vendor but that's expensive
07:49.35argos73that's another reason I like intel boards - tons of different versions to select from...  even a couple models with no extra crap built in.   (geez, I'm starting to sound like an Intel sales rep!!)
07:50.07shimithe only real problem with intel boards is that they suck in performance. :)
07:50.35sleepy_oneIntel is usually pretty well behaved and pretty stable but slower as shimi said
07:50.37argos73some of them do...
07:50.43argos73not all
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07:50.59sleepy_onebut now Intel is out of the motherboard biz from what I heard
07:51.08sleepy_oneI personally prefer AMD
07:51.36shimiamd doesn't make motherboards, methinks
07:51.38sleepy_oneAMD64 K8 family preferably with nForce chipsets
07:51.57shimii got an amd64 with nf4, and I am very happy. amazing machine
07:52.19sleepy_oneshimi, yes indeed I meant AMD64 K8 compatible motherboard
07:52.44gvag11there is should be a forum or something that user can report the hardware compatibility issues since the compatibility notes from Digium is really nothing....
07:52.48sleepy_onethe nF4 is a great chipset
07:52.58sleepy_oneI take that back!
07:53.08sleepy_oneit's an AWESOME chipset!!!! :-D
07:53.25shimigigabyte recently made a motherboard that supports SLI, and placing the nf4 chipset in it too, resulting in the possibility to put 4 (!!!) graphic cards in one motherboar
07:53.27shimid
07:53.47sleepy_oneonly thing is the new nF4 and K8T890 chipsets have PCI express so they reduce the number of normal PCI slots
07:54.00shimihere's a picture of the beast: http://www.tomshardware.com/motherboard/20051004/images/platform-intro.jpg
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07:54.25argos73off to snooze... one more day of work....  later
07:54.29sleepy_onenice :-)
07:54.32sleepy_onegnite argos
07:55.04gvag11bye bye guys... Thanks for the usefull things...
07:55.09*** part/#asterisk gvag11 (n=g@84.254.12.236)
07:55.18sleepy_oneyw
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07:56.29zoahey ho
07:56.35sleepy_onegreetings
08:01.27*** join/#asterisk montag___ (n=montag@195.223.103.50)
08:02.59montag___hi, my asterisk have a strange beahviour, when i try to connect with a remote natted sip client i receive the register request (i can see it via tcpdump), but asterisk, with sip debug on, show nothing.....my asterisk box have a public ip, i've set bindaddr=0.0.0.0 in my sip.conf, what's the problem ???
08:06.53kaldemarset bindaddr=yourpublicip
08:07.37*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
08:07.56montag___but if i want to bing my public ip and my ethernet ip too ?
08:08.03sleepy_onegnite all
08:08.50*** join/#asterisk flenders (n=fserto@61.8.29.101)
08:09.58flendershey, does anyone know a VoIP provider in the US that doesn't have a monthly fee?
08:10.36flendersa sort of prepaid account
08:15.14flendersMGSsancho, thanks mate
08:15.21MGSsanchonp
08:16.49montag___my netstat show a thousands of Recv packets in queue for port 5060, seems that asterisk don't read the buffer.....
08:18.18*** join/#asterisk T0aD (n=toad@epsylon.org)
08:18.23T0aDasterisk sucks
08:18.31T0aDim sorry but sip auth is a PAIN IN THE ASS
08:18.38JamesDotComoh
08:18.41JamesDotComso you're retarded
08:18.58T0aDdo you know what im talking about ?
08:19.07shimiT0aD, so buy a better solution at $2000 :)
08:19.19T0aDshimi, they received the patch to fix that
08:19.20shimiif you can find one, that is.
08:19.35T0aDbut apparently they prefer to make some conf
08:20.07T0aDcya, dumb protectors of nothing
08:20.11*** part/#asterisk T0aD (n=toad@epsylon.org)
08:20.15MGSsanchowhat?
08:20.18shimiheh
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08:20.25MGSsanchohahahah
08:20.32JamesDotComhahaha
08:20.39JamesDotComwhat a weird unit
08:20.47*** join/#asterisk lehel (n=asd@82.79.20.17)
08:20.52MGSsanchohes just really pissed of
08:20.59MGSsanchodidnt een ask for help
08:21.09lehelhello
08:21.32shimimaybe he wants a better price... after all, * is very costy. ;)
08:26.34iDunnodamned expensive it is ;)
08:27.37shimiThe CD I burned the software on costed about 50 cents... I think I'm going to bankrupt because of *
08:32.36protiencan anyone help me to troubleshoot this error
08:32.37protienOct  7 17:59:28 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"3477274087" <sip:3477274087@203.173.26.187>;tag=as752065d4'
08:32.43protienit happens when i make an outgoing call
08:33.46Delvarwell it look sliek it Failed to authenticate on INVITE
08:33.58Delvartry using the right username/fromuser/password
08:34.13protienim using the right username/fromuser/password
08:34.16protienbecause its registering juse
08:34.28Delvarcheck the from user is corect
08:34.34protienits correct
08:34.37Delvarhmmm
08:34.49Delvarget siptrace and pastbin it
08:34.50Delvar~pb
08:34.53jbotextra, extra, read all about it, pastebin is a place to paste your stuff without flooding the channel - try http://pastebin.ca/
08:34.58protiensiptrace?
08:35.04Delvaro_0
08:35.40Delvarat asterisk cli `sip debug`
08:35.42Delvarthen make teh call
08:35.48Delvarcopy the results to pb
08:36.55fishboy1669morning
08:38.02protienokay delvar
08:38.16protienhttp://pastebin.com/385981
08:40.57Delvarhmm interesting
08:41.05Delvarwhats your sip.conf look like?
08:41.43protienhttp://pastebin.com/385983
08:41.50protienthats my [stanaphone] entry
08:41.56lanceyhi guys
08:42.42montag___there's some problem with asterisk and kernel 2.6 network ? My receive queue for port 5060 it's full, but asterisk (with sip debug on) shows nothing....
08:43.28Delvarprotien: type=friend so you can recive incoming calls.. but that besides the point
08:43.39protienyeah
08:43.44Delvarprotien: config looks ok...
08:43.57protienthats what i cant work out delvar
08:44.00protienthe only thing i can think of is
08:44.03protienOct  7 17:59:28 NOTICE[20804]: chan_sip.c:9385 handle_response_invite: Failed to authenticate on INVITE to '"3477274087" <sip:3477274087@203.173.26.187>;tag=as752065d4'
08:44.09protieninstead of sending 3477274087@203.173.26.187
08:44.13Delvarprotien: is that number in your dialplan on providers side?
08:44.23protieni should be sending 3477274087@sip.stanaphone.com
08:44.29Delvarfromdomain
08:44.59protienoh there is a fromdomain feature, for the sip.conf?
08:45.08Delvaryeha
08:45.12protienlemme try that
08:45.43shimianyone here has a GXP-2000 phone ?
08:45.50Delvarme!
08:45.56shimihow is it?
08:46.01Delvarugly
08:46.12Delvarworks ok though
08:46.13shimifunctionality wise :)
08:46.31Delvarnot realy used it much, im using a snom mainly
08:46.53Delvardoes prety much everything you need form a phone
08:47.02shimidid you encounter a problem where you hear nothing coming from asterisk ?
08:47.06Delvaryou can like make calls.. recive calls..
08:47.24Delvaryeha but that was a hardware fault and got replacement
08:47.40shimionly from asterisk, I mean. other sounds sound well
08:47.42protienwell that didnt work delvar
08:48.00Delvarwell id say go to there support
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09:03.09szerhi
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09:09.10jonathhmorning!
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09:11.04lanceymorning, jonathh
09:15.17jonathhi like today.
09:15.19jonathhit is Friday
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09:16.02lancey:)
09:16.16lanceyi like tomorrow more :)
09:16.23lanceyAND the day after tomorrow
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09:23.41Delvari lik eeating
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09:24.15e3ghi
09:24.41ascois there some one that will help me (complete newbie) with some questions i have got regarding forwarding a group to a queue, and logon/logoff status indication of agents on cisco 79xx phones
09:25.12FABRIZIOxxxhello all .. how do i transmit silence when using pcma codec?? i'm having problems with users that are complaining that the phiones do not work but they actually are .. this because on boith sides they hear silence until one of them says something ..
09:25.24e3gI want to allow _44.,    in my extensions and want to hangup() at _447.,    but everynumber goes to _44 even it is 4477 ..
09:25.46Delvar<FABRIZIOxxx>: disable silance supprestion on the phones
09:26.00lancey*silence suppresion
09:26.06lanceyor, actually, VAD
09:26.06Delvarthat too
09:26.09lanceyon most phones
09:26.19lanceyDelvar it's the same, i just corrected you
09:26.25Delvari know :)
09:26.25lanceyVAD stands for Voice Activity Detection
09:26.49Delvari realy should learn to type
09:26.55lancey:)
09:26.59FABRIZIOxxxi tried .. but on GXP2000 it seems to work only for g729 protocol
09:27.00lancey*really :)
09:27.03Delvaryou think after 15 years id be good at it
09:27.11Delvar:P
09:27.14*** join/#asterisk oej (n=Olle@83.210.106.9)
09:27.50e3gany help ???
09:28.15e3gI want to allow _44.,    in my extensions and want to hangup() at _447.,    but everynumber goes to _44 even if number is 4477123345 ..
09:28.20Delvare3g:
09:28.44*** join/#asterisk Guus_ (n=g@139.63.241.162)
09:28.55Delvare3g: its a problem with asterisk dial plan, it matches on the _44 before it matches teh _447.. you need to add teh _44 to a context and include it in the main contact below the _447
09:29.29lanceyhmz
09:29.39lanceyi've had similar setups with latest CVS
09:29.47e3gok....
09:29.49lanceyoverriding just one of the priorities
09:29.53lanceyand it works like a charm
09:29.54e3gI check it out
09:29.54e3gthanks
09:30.30Guus_Hiya. We're using an asterisk with zaptel setup. Does anyone has an idea why the time between phone-pickups changes everytime someone calls in? (sometimes, asterisk kicks in after one ring, sometimes it takes forever before the line gets picked up)
09:31.16oejUp in the air above Norway on my way to Astricon. On the plane!
09:32.37lancey:)
09:32.52iDunnoe3g: erm, is the . representing a bunch of numbers, or just 1 in your case? because you could use _44X and _447X
09:33.09DelvariDunno: he left already
09:33.14iDunnooh, must remember to read the rest of the sentence too.
09:33.19iDunnoDelvar: oh yes.
09:33.39iDunnoI'm hoping that the afternoon is better ;)
09:33.45Delvar:)
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09:39.59samylhi
09:40.16samyli've a little question about the timer for meetme
09:40.54samyli'd like to know if it is possible to use a card FXS Digium for the timer
09:41.14samylor if i must use FXO card
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09:42.42Szolkehi
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09:52.14Delvarsamyl: just a card on its own should do it
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09:58.02samylDelvar : so, I can use FXS for the timer?
09:58.32Delvarsamyl: i think so... as i understand it it takes timing off teh card not eh module
09:58.45Delvarbut i oculd be wrong :)
09:58.54Delvaryou could always use ztdummy :)
10:07.16szerhi everyone
10:08.48szerI've got some problem with chanspy. When we try to listen a conversation between A and B we can hear only inaudible noises at the end connected to the chan-spy. This is only a problem if either A or B, or the spy is softphone (IAX2 based). If I switch on the jitterbuffer it's solve the problem, but the spying party got a 500ms delay
10:19.51*** join/#asterisk asco (n=just_me@193.173.119.247)
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10:21.29montag___my asterisk have bindaddr=0.0.0.0 in sip.conf, the system receive upd sip packets (checked via tcpdump..) but asterisk wint sip debug enable show no output....any tips ?
10:21.58fishboy1669hi guys, anyone know if there is a command in extentions.conf to detect if a phone has a call on it.
10:22.25lancey`awayfishboy1669: ChanAvail
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10:24.09ascocan someone tell me why i can't use queue's is a ring group? i am new to asterisk and thi is one of the functions it has got to have
10:25.07SzolkeHi all, I would like to ask the following questions of Aastra 9133i phones. We would like to use these phone under Asterisk... is it possible when there is incoming call to a group and it is not answered, then a led can be blinking showing that there is an incoming call in that group. So anyone could answer that call even from a different group.
10:25.17lanceyasco: what are you trying to do?
10:25.21lanceyring several lines at once?
10:25.30kaldemarmontag___: what does 'netstat -nl' say? are your interfaces listening to port 5060?
10:26.12montag___udp    39292      0 0.0.0.0:5060            0.0.0.0:*
10:26.25montag___why this value in Recv queue ?
10:27.50fishboy1669lancey i have looked at ChanAvail it seems to tell me there is a phone plugged in but not wethere there is a call in process to it
10:28.57lanceyIf the option 's' is specified (state), will consider channel unavailable
10:28.57lanceywhen the channel is in use at all, even if it can take another call.
10:29.10ascosorry lancey i was held up biy someone
10:29.31lanceyfishboy1669, actually, it is ChanIsAvail, my mistake
10:29.43lanceywhen used with the "s" option, it does exactly what you need
10:29.55lanceythough i have not tested or used that... but it should work :)
10:30.23ascoi am trying to call a queue via en ring group so that i can add an caller id to identify the alled number
10:30.48*** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca)
10:31.01queuetueHi.  I am trying to create a digital receptionist with a stock AAH 1.5 ...  I Click the menu entry, it asks for my extension, I give it, and then it tells me to hit *77 to record.  I do, and as soon as I hit the second 7, I get back to dialtone.  Nothing shows in the asterisk console .. is my sipura swallowing the *77?
10:31.07lanceyi hardly understood anything, sorry, asco
10:31.44ascothe idea is: caller -> group-> queue
10:32.43ascoi want to use groups to channel in to one queue
10:33.57Delvarqueuetue: check the dialplan on teh sipura default has a *XX, you should remove it or change it to *XXX
10:34.20Delvari think its an X anyway... dont take my word for it :)
10:34.28fishboy1669cheers lancey i will have a play and let u know how i get on
10:34.46lanceydon't beat me if it doesn't work :")
10:36.06queuetueDelvar: grep does not find anything in /etc/asterisk that has *X in it ...
10:36.27Delvarqueuetue: not in asterisk in the sipura
10:36.45queuetueDelvar: Ahhh.
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10:39.10nfi|ermesanyone knows a good printable ansd updated documentation for asterisk ?
10:39.44queuetueDelvar: I changed *XX to *XXX and now, I get a 2-second pause ... and a return to dialtone.  What should the dialplan setting be?
10:40.10Delvarin that case there is a problem with a@h
10:40.28Delvariv not used it so i cant help :(
10:40.33ascolancey: i hope i can make istclear by showing the route a call has to take.
10:40.43queuetueDelvar: I'm not positive. since asterisk still does not put any SI debug information out in the console...
10:40.56queuetues/SI/SIP/
10:40.58lanceyasco, maybe, i'm also not very familiar with queues, this could be the problem too
10:41.01lancey:)
10:41.15Delvarqueuetue: can you dial other numbers?
10:41.21ascook sorry
10:41.29queuetueDelvar: Yes.
10:41.40ascodo you know how a caller is is passed trough
10:41.41Delvarcan you dial other *XX numbers?
10:41.47lanceyasco: you read about queues at voip-info.org ?
10:41.49ascocaller ID i mean
10:42.03queuetueDelvar: *98 wrks for vm, I can dial other extensions and outgoing numbers.
10:42.43CleanerXanyone capable of calling enum numbers?
10:42.54ascoi'llread it rightaway
10:42.56DelvarHmmhesays: look in your sipura config for a *77 code.. i cant remember one but there might be, you can do it quicly by 'view source' on teh page and doing a find
10:43.01CleanerXwould like to test if everything works as expected
10:43.03Delvareh!
10:43.06Delvarsorry
10:43.25Delvari meant Hmm queuetue:
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10:43.35lanceyasco, each channel has its CallerID attached to it
10:43.38lanceyso it's there
10:43.44lanceyrouted wherever you route the call
10:43.53lanceyif the equipment is capable, of course
10:44.15Delvarexept when using SIP and fromuser.. as it overwrites callerid number with your sip username :)
10:45.07ascoi know but a group can adb some text/number to the caller ID but strips is as soon as it leaves the group an goes to the no answer destination
10:45.15*** join/#asterisk _omer (i=p@203.215.180.250)
10:45.18_omerhi
10:45.27queuetueDelvar: Yes, "Block ANC code" is set to *77...  Gah. :)
10:45.28lanceyDelvar, sometimes SIP gets fromuser to show "unknown"
10:45.41lanceyi'm now playing with that
10:45.53lanceyhttp://bugs.digium.com/view.php?id=0005405
10:45.58Delvarqueuetue: whos your daddy :)
10:46.09queuetueDelvar's my daddy.
10:46.14Delvarheheh
10:46.25_omerHow to do Proxy authentication in asterisk ????
10:46.33_omerSIP Proxy
10:47.12SzolkeI would like to ask the following questions of Aastra 9133i phones. We would like to use these phone under Asterisk... is it possible when there is incoming call to a group and it is not answered, then a led can be blinking showing that there is an incoming call in that group. So anyone could answer that call even from a different group.
10:47.18queuetueDelvar: Can I clear these all out?
10:47.34Delvarqueuetue: yeah unless you need them :)
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10:55.12fishboy1669yay lancey u are a genius
10:55.53lanceyno, i'm not
10:56.00lanceyshow application is my friend :)
10:56.03fishboy1669well u solved my problem
10:56.15queuetueOk, I can now see the *77 call coming into the console, but the call is terminating with a "403 Forbidden" error...
10:56.23fishboy1669was battling with that all yest
10:56.46fishboy1669had done some school boy errors though which meant that it wouldnt work no matter what
10:56.48fishboy1669he he
10:56.56fishboy1669but not its fine
10:57.30fishboy1669sweeeet
10:57.35fishboy1669so chuffed here
10:57.39fishboy1669big jump forward
10:58.01lancey:)
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11:05.15CleanerXanyone capable of calling enum numbers?
11:05.41*** part/#asterisk Proteque (n=gjorans@213.184.199.245)
11:08.30ascolancey: thank you for your help,  ill try to find a work around.
11:08.53*** part/#asterisk asco (n=just_me@193.173.119.247)
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11:32.33Zeeek~seen kram
11:32.36jbotkram <n=mark@pdpc/sponsor/digium/kram> was last seen on IRC in channel #asterisk, 4d 18h 36m 46s ago, saying: 'cool'.
11:33.06christomark spencer was out getting sloshed with my buddies last night :)
11:33.21christotoo bad I missed the sesh
11:33.54christoanyway, I'm starving... biab
11:34.40Zeeekwhere?
11:35.10*** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm)
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11:37.55johnmchristo: heh, ditto funilly enough.
11:38.14johnmchristo: I was gonna go to the LWE, but never got the chance in the end.
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11:43.45_omerHi,,,,,is there any Asterisk Based VoIP Service Provider??  Pc 2 Phone, Phone 2 Phone etc?????
11:48.19*** join/#asterisk Szolke (n=Szolke@217.116.36.22)
11:48.45Zeeekthis was London last night?
11:49.01Zeeek_omer there are several in France
11:50.07_omerany url ?
11:51.23lancey`awaybyez all
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11:54.44johnmZeeek: yeah
11:54.51queuetueWhen I dial 411 or #, I get a fast busy - does that indicate the AGI directory script has failed?  How would I debug this?
11:56.14ZeeekTonight it's Paris
11:58.04queuetueGah, spirura was eating it again.
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11:58.55kahuna_Hi. I want to increase the time between digits that a user dials. I have this: exten => _9.,1,DigitTimeout(10) exten => _9.,2,Dial(Zap/g1/${EXTEN:1}) exten => _9.,3,Congestion
11:59.16kahuna_but the inter digit timeout is not 10 like I would like it to be.
12:00.25*** join/#asterisk JunK-Y (n=junky@Toronto-HSE-ppp3781375.sympatico.ca)
12:04.45*** part/#asterisk _omer (i=p@203.215.180.250)
12:11.31Tilikahuna_: use responsetimeout
12:12.48Tilibut digit timeout should work
12:14.08*** join/#asterisk RoyK (n=roy@80.239.107.70)
12:17.25*** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com)
12:23.49kahuna_The only thing I can think of is that * thinks it has a complete number.
12:25.05kahuna_Does responsetimeout and digittimeout work for outbound when a user picks up the hook to dial out? All of the examples I've seen are for inbound contexts after Answer() or something similar
12:34.37*** join/#asterisk apardo (n=apardo@200.Red-83-50-234.dynamicIP.rima-tde.net)
12:37.07christoZeek - yeah in London
12:37.30Zeeekhaving dinner with him tonight in Paris
12:37.45Zeeekso he must be on his way
12:37.49Zeeekhangover time
12:37.52christoyes he is :)
12:39.14christoZeeek - ask Mark if he'd like a double Jameson Whiskey and if he likes we can sponge one up off the pavement in High Street Kensington (where he was drinking last night)
12:39.32christoand where he manage to vomit it back up :) rofl
12:40.18christothat's if he managed to get the train..
12:41.43Zeeekhahaha
12:41.54*** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
12:41.56Zeeekamateau drinkers
12:42.13christoaye
12:42.14Ariel_Morning everyone
12:42.26christomorning Ariel_
12:42.30ZeeekI can't get the BT102 to hear me calling!
12:42.31Zeeek<PROTECTED>
12:42.31ZeeekOct  7 14:41:26 NOTICE[21579]: chan_sip.c:8117 sip_poke_noanswer: Peer '2000' is now UNREACHABLE!
12:42.51Zeeekdrops out right away
12:42.52Ariel_Zeeek, network issue
12:43.02Zeeekboth on same side of NAT roter
12:43.07christowell it's registering at least
12:43.20Zeeekalthough I moved it to an outside network and it still didn't work
12:43.40Ariel_Zeeek, have you upgraded the firmware of the BT102?
12:43.52Zeeekactually I tried without registering and a fixed ip but it never beocmes reachable. It worked for months before
12:43.59ZeeekYes and I suspect the upgrade!
12:44.12ZeeekI'm trying to revert but I don'thave the older files here
12:44.39christoGranstreams are poop anyway afaics
12:44.53christoaltho mine are about 18 mths old now..
12:45.08Zeeekmine have all worked fine for many months
12:45.26Ariel_I only have 1.0.6.2
12:45.35*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
12:45.37ZeeekI even kinda like them, although my newer polycom takes things to a higher level
12:45.48ZeeekI was on 1.0.5.12! that was working
12:46.03Zeeekyeah they're great
12:46.14*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk)
12:47.00ZeeekI do that with the IAXy
12:47.06christoaye
12:47.11Zeeekbut the cordless are old and don't ring, no CID
12:48.15Ariel_I have a Sipura 2002 with a panasonic 2.4gh 2 line phone in my office. Other offices here use the Polycoms and some sipura 841's.
12:49.14Ariel_I even have a head set for the Panasonic.....(love to walk around the office and still get my calls).
12:52.36*** join/#asterisk |cleric| (n=dacleric@p548293C8.dip0.t-ipconnect.de)
12:57.07Kattymew.
12:57.23iDunnomew2!
12:57.25christoggrrrr
12:57.50*** join/#asterisk popvoxdave (i=user@dave2.toad.net)
12:58.01Kattyso, does anyone besides me know what today is?
12:58.06christoI keep getting 'no channel type registered for IAX, but I have a good IAX.conf..
12:58.10christofriday
12:58.12christopoets day
12:58.20Kattybecause it's a very important day in katland.
12:58.22Kattytwisted[asteria]: i know you know.
12:58.31Lathos42Hmm.. Talk like a Pirate day was September 19th, so that can't be it
12:58.40tzangerKatty: friday?
12:58.42KattyLathos42: no, it's October 7th.
12:58.45iDunnoit's the 7th Oct...
12:58.46christoFire christo from his job day - according to my boss
12:58.53Lathos42Katty: The day you first installed *?
12:58.54Kattytzanger: yes it's a friday, but it's much more important than just friday.
12:58.59iDunnois it Katty's birthday? aniversary? first install of *?
12:59.10tzangerKatty: it's Friday the 7th?
12:59.11KattyI'm all grown up today.
12:59.18tzangerit's your bday?
12:59.19iDunnoissa birthday @:)
12:59.20Kattyyes.
12:59.23KattyIt's my 21st.
12:59.23tzangerhow old now?
12:59.28tzangernice
12:59.32Lathos42Well Happy Birthday to you
12:59.32tzangerhappy birthday, katy
12:59.33iDunnoahh - youngster :)
12:59.34tzangerer katty
12:59.36christohappy birthday Katty
12:59.40Kattythanks :>
12:59.42Ariel_happy Birthday to Katty
12:59.45christo21??
12:59.55Zeeekhappy happy kappy
12:59.59protienwow happy bday katty
12:59.59iDunno# happy birthday to you, happy birthday to you, happy birthday deaaaaaaar Katty, happy birthday to you.
13:00.01protienit was mine 2 days ago
13:00.04protienit must be 21st month
13:00.09*** join/#asterisk SpaceBass (n=sp@c-24-125-184-203.hsd1.va.comcast.net)
13:00.10SpaceBasshey folks
13:00.12tzangerheh
13:00.13Zeeeksip show peers
13:00.16tzangerI turn 30 next may :-)
13:00.20Zeeeksip show beers
13:00.23protienthe big 3-0
13:00.23mmlj4dip show peers
13:00.24tzangerZeeek: hahaha
13:00.45KattyNormally work orders a cake for people's birthdays. Since I don't eat cake (cause of milk, eggs, etc) I thought they'd have something else.  ..but they didn't. And no one has said a word.......i'm a little sad :<
13:00.48protienhah if its your 30th maybe it should be sip slow beers
13:01.14iDunnoKatty: go stab them all with sharpened pencils!
13:01.24KattyiDunno: shan't
13:01.25protienim recieving a call but when it comes in im getting
13:01.26protienOct  7 22:25:51 NOTICE[24120]: chan_iax2.c:6690 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@'
13:01.33protienhow can i recieve that call?
13:01.44tzangerprotien: :-)  nah I don't drink much beer.  rye and coke, gin and ginger ale...  oh yeah
13:01.55Zeeekno rum?
13:01.57protienyeah rye and coke ive got one in my hand right now ;)
13:02.02tzangerZeeek: no I'm not a fan of rum at all
13:02.04*** part/#asterisk popvoxdave (i=user@dave2.toad.net)
13:02.10SpaceBasssomeone got a second to look at a config for me... I'm racking my brain here
13:02.11SpaceBasshttp://pastebin.ca/24843
13:02.15Zeeekrum+pineapple mmmmmmm
13:02.16iDunnohmm. dark rum)++
13:02.21protienrum and pineapple is nice?
13:02.26Zeeekgreat
13:02.29SpaceBasshttp://pastebin.ca/24844
13:02.31tzangergin, rye, tequila, vodka... no problem
13:02.38tzangerbeer is alright
13:02.39Zeeekespecially rum, sparkling pineapple juice
13:02.45tzangerI prefer the ales though
13:02.52SpaceBassbasically, randomlly, calls on zap/1 are getting into the zap/2 context
13:02.59ZeeekI'll have what everyone else is having
13:03.04iDunnogood real ales are the way forwards :)
13:03.06tzangerSpaceBass: wow that I find VERY hard to believe
13:03.46tzangerSpaceBass: I've been using * for close to 2.5 years now and it's never "jumped context" on me
13:03.56SpaceBasstzanger i'm sure its something in my configs
13:04.02cpatrysup?
13:04.05SpaceBasstzafrir doubt its a bug or anything...
13:04.05Kattymy age!
13:04.13tzangerKatty: that sucks ass.  we have to bring donuts in on our birthday
13:04.13cpatryyou're 21 !!!!!!!
13:04.15cpatry:)
13:04.18Kattycpatry: yes :>
13:04.18SpaceBasstzafrir but I've been looking at these for weeks and cannot find it
13:04.25cpatryso now we wont have to drink dasani :)
13:04.35Kattycpatry: but i like dasani
13:04.38Kattycpatry: you silly rabbit.
13:04.38tzangerSpaceBass: :-)  That is good for the soul now and again
13:04.48Kattycpatry: but not you can bring your alochol in ;)
13:04.54iDunnohmmm. rabbit in red wine sauce.
13:05.01Kattycpatry: but not drink it...cause you might be a french canadian terrorist.
13:05.02tzangernice
13:05.02SpaceBassarrruuugggggggg I wish zapata.conf was a person so I could hit it in the face
13:05.16cpatryKatty: they sucks!
13:05.38Kattycpatry: indeed. you poor thing.
13:05.48Kattycpatry: should have given it to matt ;)
13:05.53tzangerwow there's 625k files on my hard drive and it's still going
13:06.02Ariel_SpaceBass, could you might have rollover on those lines from the phone co?
13:06.23tzanger700k
13:06.25cpatryi already paid him vodka, thats enuf :)
13:06.30SpaceBassAriel_ no chance...
13:06.34Kattycpatry: k (=
13:06.35tzangermind you I have a few development environments on here so that makes some sense
13:06.53Ariel_SpaceBass, IRQ sharing?
13:07.02*** join/#asterisk IPmonger (n=ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
13:07.32Kattycpatry: what do /normal/ people do on their birthdays?
13:07.34SpaceBassAriel_ maybe, but don't think so... cli looks right, zap/2 picks up and throws the call into the zap/2 context
13:07.37Hmmhesaysoh the roosters rocked last night, what a grand time
13:07.38*** join/#asterisk firestorm-voip (n=firestor@62-181-86-226.skbbip.com)
13:07.38tzanger800k
13:07.43KattyHmmhesays: come visit for my birthday.
13:07.47SpaceBassAriel_ and it worked fine until I upgraded to aah 2.0b or what ever
13:08.01Ariel_ahh
13:08.03tzangerKatty: where do you live again?
13:08.12Kattytzanger: moo ssouri
13:08.26HmmhesaysKatty: probably not going to happen, but if you give me your addy i'll send a card
13:08.38tzangerKatty: that's right... Manxpower could have stayed with you :-)
13:08.49KattyHmmhesays: i don't want a card you goofball :P
13:09.01Hmmhesaysfine
13:09.14Kattyi'd rather have a hug anyway (=
13:09.25shimi00:0a.0 Network controller: Unknown device e159:0001 < can this be the digium card ?
13:10.12Simon-possibly
13:10.16*** part/#asterisk firestorm-voip (n=firestor@62-181-86-226.skbbip.com)
13:10.27Simon-google says: Class 0780: e159:0001   Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
13:10.34Hmmhesaysoh wow, foxnews had a wonderfully spot on article against MADD yesterday
13:10.36SpaceBassIs there a way to have a zap channel ignore a 1/2 ring?
13:10.40shimi<PROTECTED>
13:10.45Hmmhesayss/against/about
13:10.49shimiis from /proc/pci
13:11.07tzangerSpaceBass: callwaiting=asreceived?
13:11.26shimihow can I "communicate" with the card to check if it works before trying to mess with asterisk on it? :)
13:11.43SpaceBasstzafrir i'll try that... whats happening is that I've forward my zap/1 to my BV account (temp fix) but I still get a 1/2 ring on incoming calls which triggers zap/1 to pick up
13:11.45tzangershimi: asterisk is what communicates with it.
13:12.00shimibut anything in /proc, etc ?
13:12.05tzanger1.25mil files so far
13:12.11tzangershimi: not to communicate with it, no
13:12.21shimito see status, etc ?
13:13.55tzanger/proc/zap/
13:14.01tzangerbut you need the drivers loaded obviously
13:14.24shimiI have "zaptel" and "wcte11xp" and "wcusb" loaded. is that enough ?
13:14.30*** part/#asterisk samyl (n=samyl@194.167.18.244)
13:14.58Kattycold :<
13:15.03shimioh, I can see /proc/zaptel/1. Guess it's OK :)
13:15.26*** join/#asterisk musical_Duck (n=kvirc@wblv-146-236-254.telkomadsl.co.za)
13:15.31musical_Ducklo
13:15.44Kattyquack.
13:15.52tzanger1572333 files on my hard drive :-)
13:15.54tzangerfilelight is cool
13:15.57musical_Duckyou should go meow :)
13:16.02shimiI see there what appears to be 31 "WCT1/0" channels
13:16.09tzangeryup
13:16.12Kattymusical_Duck: actually, i mew.
13:16.48Kattyfile: :<
13:16.53KattyHmmhesays: :>
13:16.53file[laptop]:(
13:16.55musical_Ducksay anyone here you tx/rx fax?
13:17.01musical_Duckuse even
13:17.05file[laptop]this woman says her number isn't working, yet I can't get it to not work
13:17.36musical_DuckYou have the right number? :)
13:18.03Kattymister file always has the right number
13:18.11file[laptop]I pulled it up by the account
13:18.13Kattyhe practically is the number.
13:18.21*** join/#asterisk jeffik (n=Jeff@CPE0020ed8494b8-CM0012c999ca4e.cpe.net.cable.rogers.com)
13:18.33*** join/#asterisk brad_mssw (n=brad@shop.monetra.com)
13:18.36file[laptop]I've called directly, through a third party toll-free termination provider, and through our own toll-free termination provider
13:18.40file[laptop]works in every case
13:18.45musical_DuckMy faxes over plain pstn break up for some reason
13:19.13Kattyfile: i don't suppose she told you exactly why she thought it wasn't working, did she?
13:19.25file[laptop]Katty: she said it just did nothing
13:19.53file[laptop]"That DID is not working, we get nothing when we call it"
13:19.59Kattyfile[laptop]: did you talk to her on that line when you called?
13:20.07file[laptop]no it's an automated system
13:20.14file[laptop]it's been intermittent since yesterday
13:20.18Connor-What the frack is going on with asterlink switches today.. they're bouncing up and down like mad.
13:20.19Kattyhmmm.
13:20.34Kattyfile[laptop]: it clearly requires hugs.
13:20.35SpaceBassOk... profress
13:20.35bjohnsonI see your frack and raise you a frick
13:20.35file[laptop]bloody hell
13:21.42Kattyfile[laptop]: mew?
13:21.59musical_DuckThey stole some of my telco's copper the other day, network was dodgy all day
13:22.23Kattymaybe they needed it for something more important
13:22.30protien[macro-stdexten];
13:22.33protienall these macros, are they needed
13:22.38protienor can i remove them?
13:22.45file[laptop]if you use them, they're needed
13:22.51protienwhat are they for mate
13:22.55musical_DuckNah, was probably just being shiny ;)
13:23.47musical_DuckSo bout them faxs programs, anyone use rx/tx fax?
13:24.19musical_Ducksilent s BTW :)
13:26.24musical_Duckmmm Friday evening slump heh?
13:29.09Kattymusical_Duck: ((=
13:29.16Hmmhesaysand I never, what to do this again, heartbreaker
13:29.21*** part/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek)
13:32.09*** join/#asterisk MikeJ[Laptop] (n=ircatjer@mi.origenfinancial.com)
13:39.14*** join/#asterisk bon (n=bon@fux.wnet.sk)
13:39.41protienim having problems with incoming calls, outgoing work fine but when i recieve a call on this iax interface i get
13:39.42protienOct  7 22:50:58 NOTICE[24120]: chan_iax2.c:6690 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@'
13:39.46protienhow can i recieve calls at s@
13:40.30*** join/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk)
13:40.34bonwhat kind of billing solution wouuld you suggest?
13:40.39bonor managing solution..
13:40.45bon(adding users/extensions/cdr..)
13:41.45bjohnsonprotien: I don't think you can use '@' as an exten name
13:42.06musical_Duckchan_zap.c:4543 zt_write: Cannot handle frames in 2 format <- what does this mean? Was recving fax when this happend
13:42.10protienhm i dont understand how to recieve these calls then
13:42.28ret28This a SIP problem ooi?
13:42.30bjohnsontry getting them to not specify an exten
13:42.43ret28(just been dealing with '@'s myself ... )
13:42.46KattyHmmhesays: i have to hand it to work.......they tried.
13:42.50protienits a public system
13:42.54protieni cant get them to do anything bjohnson
13:42.55bjohnsonprotien: if no exten is specified, * will default to use the 's' exten
13:43.09KattyHmmhesays: halfway expected a cake this morning...with milk and eggs in it...but instead, they brought in fruit salad and a brownie recipe of mine.
13:43.11protienhm
13:43.18protienhow can i forward the s exten to a number
13:43.18bjohnsonret28: his error said iax
13:43.25bjohnsongoto()
13:43.27KattyHmmhesays: sadly, they didn't follow the recipe in such a way that would make it vegan (wrong margarine and the wrong chocolate chips) but they did try.
13:43.31ret28Oh ok, this is what I get for leaping in halfway through :)
13:44.18file[laptop]are you feeling old now?
13:44.23Kattyfile[laptop]: not yet.
13:44.23kpettitAnybody had any luck using t.38 or t.38 for faxing?  I haven't seen alot of information out there on how to get it working with asterisk
13:44.28Kattyfile[laptop]: sorta depressed actually.
13:44.35*** join/#asterisk marc324 (n=marc3234@206-248-159-4.dsl.teksavvy.com)
13:44.51ret28Though for anyone clued up about SIP ... how to get asterisk to stop demanding authentication from guest calls where the From username just happens to match one of my own SIP users? (even though the from domain is entirely different, and not one of my own)
13:44.58file[laptop]kpettit: coppice did something to make passthrough work, but I dunno if it really works... besides that you should have read tons of posts about how it doesn't work and isn't supported
13:45.11Kattyfile[laptop]: mamma called me around 6:30 this morning, but didn't say a thing about me being 21
13:45.13marc324for sending incoming calls to xlite running on a pc... where do I set the ip address of xlite?
13:45.26file[laptop]Katty: scary
13:45.34kpettitfile[laptop], ugggh sounds wonderfull
13:45.48file[laptop]marc324: it either registers to your Asterisk, or you specify it's IP in the host entry for the peer
13:45.48bjohnsonhow does margerine and chocolate chips (both from plants) get to be non-vegan?
13:45.49Kattyfile[laptop]: i didn't expect her to say Happy birthday...she's a JW... but she usually says something along the lines of how does it feel to be $age
13:45.50tzangerKatty: your mom calls you at 6:30am regularly?
13:46.03ret28marc324: In a host=<ip> for a user in sip.conf
13:46.06file[laptop]it was my birthday yesterday, and today it is Katty's birthday!
13:46.06tzangerbjohnson: milk chocolate?
13:46.07Kattytzanger: she was returning my call from yesterday aftenroon
13:46.09kpettitThe impressing I've been getting through readying about problems with fax is t.37 and t.38 will be about the only thing that can work reliably
13:46.14tzangerKatty: ahh.
13:46.19Kattybjohnson: milk
13:46.39kpettitI'm using ulaw right now with a upstream sip provider and it seems to work in most cases, but of course there are some that we just can't get to work
13:46.58file[laptop]oh beautiful
13:47.04file[laptop]my copy of the O'Reilly book is out for delivery!
13:47.26bjohnson(non-vegan) birthday brownie rejects
13:47.30file[laptop]yay brownies
13:47.31Kattyfile[laptop]: hippo birdie two ewe.
13:47.34*** join/#asterisk jimmy_deanPB (n=jhodapp@adsl-70-228-242-126.dsl.ipltin.ameritech.net)
13:47.44Kattyfile[laptop]: hippo birdie two ewe. hippo birdie deer ewe.
13:47.52Kattyfile[laptop]: hippo birdie two ewe.
13:49.34*** join/#asterisk Godsey (i=lanny@pdpc/supporter/sustaining/Godsey)
13:50.11*** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca)
13:50.59marc324in [xlite] section in sip.conf , the line "host=" is what?
13:51.23bjohnsonprobably the ip of the host you want to connect to
13:51.23file[laptop]dynamic if X-Lite registers to your asterisk box, or the ip/host of where the X-Lite is
13:51.40ret28Yeah, you could just set host=dynamic, and have X-Lite register
13:51.52ret28(then asterisk will do magic to send the call to the SIP/xlite channel)
13:51.54*** join/#asterisk dalabera (n=Dalabera@pmr.pmrtechnologies.com)
13:51.57queuetueWhat causes 503, service unavailable errors?  The user is on a rt31p2-na, picks up the line, hears the stuttered dialtone (message waiting), hits *98 and gets a fast busy - the logs show "503 service unavailable".  Same happens if he tries to dial an extension or dial out.  The rt31p2 *is* the firewall, so there should not be a NAT issue, and my firewall here is forwading 5060-5069 and 10000-20000 to asterisk...  What could be cau
13:52.28queuetue(Connections from inside my firewall work just fine...)
13:52.54Kattyanyone hear about that python that ate a 6ft alligator?
13:53.01*** join/#asterisk funxion (n=nunya@mtnuser.icgws.com)
13:53.23ret28The one that blew up? Yeah.
13:53.33ret28Would make a good motivational poster, "Ambition"
13:54.24Kattyret28: yes, it would
13:54.57ret28Or whatever the opposite of motivational is anyway, considering how it turned out ...
13:55.17Kattyambition fits nicely
13:55.19*** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it)
13:56.22ret28http://despair.com/ambition.html ... hahah, already done ... sort of
13:57.04SpaceBassthe CLI is showing that i have a failure in a context and then it starts over, how can I get more detail?
13:57.16SpaceBassi've increased to vvvvv but it doesnt tell me more
13:57.21FABRIZIOxxxhello all .. is it possible to send the called party ID on the display of a SIP phone?
13:57.22SpaceBasslike what line is failing
13:58.14*** join/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp)
13:59.05funxionwhat is the way to read the results of a system() to a variable?
13:59.06protienhow do i forward any incoming calls that dont have a defined extension to a certain number?
13:59.21tzangerDonald Rumsfeld rushes into Bush's office.  "Mr. President, I have terrible news.  I've just received a report that three Brasillian solders were killed!"
13:59.26tzangerBush replied "That's a tragedy!... wait...  How many's in a brasillian again?"
13:59.31SpaceBasslol
13:59.39SpaceBassheard this the other day
13:59.45tzangeryeah it's old but funny
13:59.46SpaceBasss/this/that
14:00.13Ariel_So any word on the L3/Cogent problems???
14:00.49FABRIZIOxxxprotien, you could try with the gotoif() command .. maybe
14:01.05ret28protien: Something like "exten => _.,1,Dial(<whatever>)" , make sure it's the last one in the context of valid numbers though. (. is a wildcard). Disclaimer; I think I might be wrong, try it and see.
14:01.11SpaceBassI'm getting Spawn extension (from-pstn, s, 2) exited non-zero on 'Zap/1-1'
14:01.24SpaceBasshow can I pinpoint which line in the dialplan is causing it?
14:01.45SpaceBassunfortunatly there are some includes, etc in the context... there is no exten=> s,2...
14:01.52protienlemme try that ret28
14:02.46protiennop
14:02.47protienOct  7 23:29:37 NOTICE[24116]: chan_sip.c:10154 handle_request_invite: Failed to authenticate user "2025560000" <sip:2025560000@66.54.140.46>;tag=as12b501b7
14:02.53protienstill get these
14:03.02ret28That's not a dial plan problem ...
14:03.18tzangerSpaceBass: unfortunately that's not able to be done from my experience
14:03.33tzangeryou need to sprinkle NoOps throughout and follow the breadcrumbs
14:03.34SpaceBasstzafrir mack to the manual parsing :)
14:03.37ret28Is 2025560000 one of your users?
14:03.46protienits an incoming call
14:03.54protienits a remote user
14:03.57SpaceBasstzafrir thanks... always forget about noops
14:04.08tzangernoops ist WUNDERBAR!
14:04.18*** part/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net)
14:04.38Kattyanyone have a recipe for rum cake?
14:04.50SpaceBasswould a noops in zapata.conf show up when a call comes in on a zap channel
14:05.00SpaceBassor are they all parsed at startup only?
14:05.01tzangerKatty: take cake and soak in rum
14:05.03iDunnotake cake mix, add run, cook.
14:05.08tzangerSpaceBass: no
14:05.14tzangerSpaceBass: noop is a dialplan command
14:05.16iDunnoor tzanger's way :)
14:05.22Kattyhmm
14:05.29Kattycake mix is not vegan
14:05.30tzangeralthough I don't think soy cake is all that tasty
14:05.34Kattynor is most cake you buy on the market.
14:05.52Kattytzanger: soy cake? the non vegan parts of cake are milk and eggs
14:06.04Kattytzanger: which are easily replaced with corstarch, water, and soy milk
14:06.04tzangersoy milk and soy eggs, of course
14:06.09Kattyhaha soy eggs
14:06.17tzangerfrom those soy chickens you were talking about earlier
14:06.19Kattycornstarch, silly
14:06.27iDunnoheh
14:06.35ret28protien: Hmm, does that user have an entry in sip.conf ? Wondering why asterisk is insisting on authenticating it.
14:06.36tzangerdammit now I have to watch chicken run again
14:06.38tzangerI love that movie
14:06.48ret28(entry as peer, user or friend)
14:06.48SpaceBassi haven't seen it yet.... good?
14:06.50iDunnochicken run is good :)
14:06.59tzangerNOOOOOOOOOOOOOOOO CHICKEN ESCAAAAAAAAAAAPES FROM TWEEDY'S FAAAAAAAAAARM!!!
14:07.01iDunnoSpaceBass: how on earth have you not seen it?!
14:07.07tzangerI say that to my kids when I put them to bed sometimes :-)
14:07.27SpaceBassduno... I usually like animated stuff like that
14:07.32tzangerwallace and gromit rocks
14:08.44tzangerjust don't ask where the soy nuts come from
14:10.29SpaceBasscan I check a caller ID and drop a certian caller right into DISA with out a password?
14:10.46file[laptop]yes.
14:10.50tzangeralthough I also had the kids going around the supermarket saying in low growley voices "mmm.... beef is goooooood..." this summer
14:11.06tzangerwhich is rather funny to hear a 4 year old boy in as low a voice as he can muster
14:11.08file[laptop]exten => 8777811208/5068780147,1,Playback(muffins)
14:11.18tzangerSpaceBass: yes, but be careful, that's security through obscurity
14:11.19file[laptop]8777811208 is the DID, 5068780147 is the callerid number
14:11.34SpaceBasstzafrir yeah, not toooo worried... home setup
14:12.06SpaceBassgoing to Ireland, got an irish sim card and cell, want to call my irish voip DID and have it drop me into DISA so I can get dialtone and make us calls
14:12.21iDunnofile[laptop]: you'd probably want to Answer() it before trying to play back stuff at them.
14:12.38file[laptop]iDunno: it'll automatically Answer
14:12.45mutilatorwhats a good way to get cron to run on the last day of the month every month?
14:13.00file[laptop]there's an option to disable that too... *G* so you can play back audio as inband session progress
14:13.11marc324how do I restrict xlite users from registering?
14:13.17Kattytzanger: save a chicken, eat an oreo.
14:13.25iDunnomutilator: can't you just run it on the first day of the month instead?
14:13.31ret28marc324: With a username= and a secret= for the [xlite] entry
14:13.36protiensorry ret28
14:13.38protieni pasted you the wrong line
14:13.39protienOct  7 23:40:22 NOTICE[24120]: chan_iax2.c:6690 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@'
14:13.40file[laptop]marc324: if they're peer/friend entry in sip.conf is not dynamic, ie not host=dynamic, they can't register....
14:13.46protienthis is the reject attempt, that im trying to filter
14:13.50mutilatoriDunno: no it has to log the AR at the end of the month
14:13.56protienthis is an incoming call from goiax network
14:14.04tzangerKatty: I actually really dislike oreos, I never ever understood why people liked them so
14:14.10tzangerpirate cookies though... now that's good
14:14.18iDunnomutilator: hmm, that's a bit of a pain then.
14:14.29mutilatornope found it
14:14.31Kattyhttp://us.news3.yimg.com/us.i2.yimg.com/p/ap/20051005/capt.mh10310051654.gator_python__mh103.jpg <- why you shouldn't have an alligator for breakfast.
14:14.33ret28Hmmm ... I think if that dialplan was wrong though, there'd still be an error unrelated to the incoming channel
14:14.36mutilatorthere is a L character
14:14.52marc324right now, any xlite can register....  where do I specify the password in sip.conf?
14:14.52mutilator* * L * *
14:14.54SpaceBassarrruuuggg why are these calls in the wrong context! I hate zapata.conf and all its friends
14:14.57ret28Still, I'm not certain, I've only been playing with asterisk for a few days :)
14:14.58mutilatorlast day of the month
14:15.29ret28And wondering how to stop it being dumb with insisting that remote SIP calls authenticate just because the username matches one of my own
14:15.41ret28(regardless of the from address domain)
14:16.43*** join/#asterisk DarthClue (i=user76@wsip-68-99-73-32.tu.ok.cox.net)
14:17.11iDunnomutilator: the only way I can think of getting cron to do that would be to write a script that checked wether it was the last day of the month and call the script instead... and make the cronjob run between the 28-31
14:17.28mutilatori jus said..
14:17.28*** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com)
14:17.28mutilator* * L * *
14:17.31iDunnomutilator: coo, when did that get there?
14:17.39mutilatorruns on the "last" of that type
14:17.44mutilatorhttp://wiki.opensymphony.com/display/QRTZ1/CronTriggers+Tutorial?decorator=printable
14:17.54*** join/#asterisk PMantis (n=Miranda@66.251.89.34)
14:18.07mutilatori just have to get that
14:18.22iDunnodoesn't appear to be documented in my crontab page :(
14:18.51PMantisMy IAXy's recently started giving me one-way audio. What should I check?
14:18.54DarthClueis there a reference on how to set channel variables within c code?  i need to hack the originate command to let me set some channel variables when i originate a call?
14:19.27fileDarthClue: I'll give you a hint.
14:19.47Kattyneat! a blue whale is bigger than a tyrannosaurus
14:19.48filein a sec
14:20.03filepbx_builtin_setvar_helper(channel, "name", "value");
14:20.15filewhere channel is a pointer to the channel structure
14:20.21iDunnoKatty: surely only larger than the largest set of bones they've found of a tyrannosaurus?
14:20.28Kattyhttp://www.enchantedlearning.com/sgifs/Sizecomparisons.GIF
14:20.43fileKatty: don't you have works-like stuff?
14:20.54*** join/#asterisk Abbas (n=Abbas@203.81.225.91)
14:20.55KattyiDunno: uhh....much bigger. i dont' think it would matter.
14:21.01Kattyfile: nah. i'm on call though!
14:21.08fileKatty: ooooooh
14:21.22Kattymew :<
14:21.46Abbashello
14:21.48AbbasOct  7 10:24:43 WARNING[990]: rtp.c:1425 ast_rtp_bridge: codec0 = 13 is not codec1 = 256, cannot native bridge.
14:21.51iDunnoKatty: coo - yes, hmm. interesting.
14:21.56Abbasy this warning comes
14:22.11iDunnoKatty: then again, I think I'd be more worried by a tyrannasaurus chasing me than a blue whale...
14:22.13DarthCluesize doesn't matter.  at least not when you are a bowl of petunias materializing in mid-air
14:22.37iDunnofor one, if it was a blue whale I'd probably more be worried about the fact that I was in the middle of a cold sea type thing.
14:22.46KattyiDunno: you silly humans and your fears of being eaten by extinct creatures.
14:23.02iDunnoheh
14:23.09KattyiDunno: i'd be worried about a sperm whale
14:23.18KattyiDunno: they're the largest carnivore
14:23.46Kattyhttp://www.oceanwanderers.com/SpermWhale.7342.JPG
14:24.08iDunnoKatty: I'd still be more worried by the whole being in the cold sea thing, probably ;)
14:24.19Kattyhttp://www.oceanwanderers.com/NYSpermWhale.html <- poor thing :<
14:24.33fileI'm hungry :\
14:25.28Kattyfile: muffins.
14:25.29iDunnoit was a bit of a careless spermwhale...
14:25.39KattyiDunno: it got beached by a storm
14:25.42iDunnodya reckon that it was out getting drunk with it's mates before that?
14:25.42*** join/#asterisk hotgrits (n=hotgrits@192.160.238.156)
14:25.47iDunnoahh. that'd do it.
14:25.53KattyiDunno: they live in deep water, obviously
14:25.58fileKatty: the downstairs freezer is inaccessible right now
14:26.07Kattyfile: :<
14:26.14fileit's sad :(
14:26.29*** join/#asterisk jonathh (n=asd@host81-154-159-222.range81-154.btcentralplus.com)
14:26.42christoI wonder if they removed it
14:26.51christoor if it's still there in some advanced stage of rot
14:28.03KattyiDunno: that whale was a calf too
14:28.14KattyiDunno: just a little baby :<
14:28.22iDunnoKatty: indeed :(
14:28.49Kattyfile: don't breathe that. eat it.
14:28.57fileohhhh
14:29.05iDunnoheh
14:29.15Kattyyou son of a silly person.
14:29.55christoI'm having the same problem as this bloke: http://asterisk.linkx.net/asteriskusers/200401/msg00627.html
14:29.59christodamned if I can fix it :(
14:30.39fileKatty: I force you to listen to the soundtrack for The Phantom of the Opera
14:30.39Kattyyay!
14:30.40filechristo: no chan_iax2.so
14:31.38christofile - chan_iax2.so is there
14:31.44fileis it loaded?
14:31.55christooh poop
14:32.00christono
14:32.01znoGi think the guys at openpbx should put something on their page stating whether they do or do not have anything to do with Asterisk, cause it has created confusion.
14:32.42christocan I reload modules from the CLI?
14:32.49FuriousGeorgewhats the range for the rxgain setting in zapata.conf?  10 - -10 gb?
14:32.51christoie without dropping *
14:32.55FuriousGeorgedb***
14:32.59filechristo: yes, load <module name> reload <module name>
14:33.03fileie: load chan_iax2.so
14:33.06christothanks file :)
14:33.14Dr_Ray100%-100%
14:33.36FuriousGeorgeDr_Ray: u sure thats a percentage?  i heard it was db
14:34.02Dr_Raymy asterisk book  says it's a percentage.. but it could be wrong
14:34.03Hmmhesaysso strange
14:34.41KattyHmmhesays: a dolphin's iq is only slightly lower than the average human's iq
14:35.30FuriousGeorgeKatty: i find that difficult to believe, you'd think they'd be able to teach'em morris code or something to communicate if that were the case
14:35.33ian_kKatty: did it take the supervised WISC-R test?
14:35.47HmmhesaysHmmm yes very strange
14:35.54Hmmhesaysmy meetme is acting up
14:36.15FuriousGeorgeKatty: not that they arent intelligent, they recognize themselves in mirrors. which is more than you can say for almost any animal
14:36.17Dr_Rayrxgain: Adjusts receive gain. This can be used to raise or lower the incoming volume to compensate for hardware differences. Takes a percentage of capacity, from -100% to +100%rxgain: Adjusts receive gain. This can be used to raise or lower the incoming volume to compensate for hardware differences. Takes a percentage of capacity, from -100% to +100%
14:36.24Dr_Rayoops
14:36.26KattyFuriousGeorge: google it (=
14:36.33Hmmhesaysi initiate a call via the manager, it calls out my vonage account, I answer my cell and it should dump me into meetme
14:36.38Hmmhesaysit does, but then I hear nothing
14:36.48FuriousGeorgeDr_Ray: thanks i believe you
14:37.02FuriousGeorgedid you guys here about the parrot who understands the conzep of zero
14:37.04*** join/#asterisk SuPrSluG (n=SuPrSluG@pool-141-149-242-183.buff.east.verizon.net)
14:37.06FuriousGeorge*concept
14:37.07ian_kDr_Ray: gain should not be adjusted beyond 10-15 in either direction, or hardware damage may result
14:37.18fileKatty: masquerade!
14:37.29Dr_Raywell, I double checked for my memory too..
14:37.46filemřřse attack!!!
14:37.58RoyKmøø
14:38.47protienis there a way i can set a callback script, so if someone dials an extension, it gets their number and calls them back?
14:38.56Hmmhesaysyes
14:39.00fileprotien: you can do anything
14:39.08Hmmhesaysfile, fix it
14:39.09fileyou just have to research how, learn, and find examples
14:39.13DarthCluefile: does originate use a channel structure?
14:39.17protienyeah im trying to find an example
14:39.27fileHmmhesays: mmm don't remember
14:39.28FuriousGeorgeso how come meetme keeps prompting me for a conference pin when i specified none in metme.conf?
14:39.31fileer that was for DarthClue
14:39.43fileDarthClue: show me source code.
14:39.51*** join/#asterisk mkrufky (n=mk@68.160.103.77)
14:39.51Hmmhesaysprotien: check out the wakeup call script
14:40.05Hmmhesayson the wiki wiki
14:40.20Hmmhesaysas for my problem, its an odd one
14:40.34fileI hear the cops! they're coming for me
14:40.37DarthCluehttp://www.pbxfreeware.org/app_changrab.c ... looking at the originate_cli section
14:40.41FuriousGeorgeHmmhesays: apply ointment, it should subside
14:40.45filekk
14:40.48*** join/#asterisk Cresl1n (n=matt@gateway.digium.com)
14:40.59christois the iax.conf now called iaxprov.conf, or is that something else?
14:41.05christoI've never needed that before on older versions of *
14:41.13Dr_Rayiaxprov is for the iaxy, i think
14:41.14*** part/#asterisk ret28 (i=rt@82-71-120-246.dsl.in-addr.zen.co.uk)
14:41.22HmmhesaysFuriousGeorge my wonderfully delightful nick'ed friend... i don't think so
14:41.24christohmm
14:41.29fileDarthClue: I see what it does
14:41.32christoI don't care about the IAXy, I just wanna trunk :S
14:41.37FuriousGeorgeHmmhesays: lol
14:41.39Dr_Rayiax.conf
14:41.46KattyHmmhesays: i think the dolphins are plotting
14:41.53SuPrSluGDarthClue:they use it in the callme.php script
14:41.53fileDarthClue: look at the function static void *originate(void *arg)
14:42.09fileDarthClue: you can probably put in the setvar stuff in that if statement
14:42.34*** join/#asterisk rudi_ (n=rudi@karfire.megabit.net)
14:42.40rudi_hey there
14:42.43Hmmhesays-- Executing MeetMe("SIP/17327905389-2694", "1337|dM") in new stack
14:42.43Hmmhesays<PROTECTED>
14:42.47Hmmhesaysseriously wtf
14:43.39christoDr_Ray - the odd thing is that I have an iax.conf, but when I reload chan_iax2.so it complains that there's no iaxprov.conf
14:43.47*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
14:44.10Dr_Raythat is odd, I thought iaxprov.conf was just for firmware upgrades
14:44.28christowell I copied one over from the samples directory
14:44.54christothis is a PITA actually.. I'm still getting this error http://asterisk.linkx.net/asteriskusers/200401/msg00627.html
14:44.57malverian[work]Hmmm...
14:45.20Dr_RayI Was wrong, it's not for the iaxy
14:45.39christoI have checked the module is there, reloaded it at the CLI, created the iax.conf, copied over an iaxprov.conf just to keep it happy, but still it doesn't seem to know what IAX2 is
14:45.43christocrazy
14:45.55malverian[work]SNOM doesn't resend it's SUBSCRIPTIONs when you reload Asterisk...
14:45.58malverian[work]That kinda sucks.
14:46.21*** join/#asterisk Blazint (n=blazin@cm225.epsilon203.maxonline.com.sg)
14:46.27FuriousGeorgedoes anyone care to speculate as to whymeetme prompts me for a PIN when there is none and complains my pin is wrong when i specify one?
14:46.40FuriousGeorgeDTMF is fine
14:46.41malverian[work]FuriousGeorge, There's a flag to not prompt for pin.
14:47.01rudi_does anyone know how a good web-interface to the asterisk-config?
14:47.05FuriousGeorgemalverian[work]:  ILL LOOK INTO THAT, THINKS
14:47.12FuriousGeorge*thanks
14:47.22FuriousGeorgesorry 4 caps
14:49.06FuriousGeorge*of
14:49.11rudi_anyone? :>
14:49.13*** part/#asterisk Guus_ (n=g@139.63.241.162)
14:49.36rudi_(i'm talking about one that supports using existing config files without any kind of databases, templates, profiles etc.)
14:50.03christopoopah
14:50.30FuriousGeorgerudi_: by gui for the asterisk config you mean "a way of getting around manually editing the configs"?
14:51.08Hmmhesaysmeetme hates me today, as soon as I enter it stops the music on hold
14:51.10rudi_yeah...i actually prefer doing it the 'manual way' but not everbudy else does..
14:51.23rudi_*everybody
14:51.54iDunnopeople like guis?! woah.
14:52.51iDunnois good for graphical manipulation, and looking at pretty rather than functional websites...
14:53.03iDunnobut what else do you actually want or need a GUI for?
14:53.26rudi_apparantly my company wants to offer asterisk-based solutions to their customers
14:53.38PMantisAnyone know why all my IAXy's would just start to receive audio, but not send? DTMF works, but not voice!
14:53.48*** join/#asterisk MicC_ (n=sum1@CPE000c419ce901-CM000a7363f92c.cpe.net.cable.rogers.com)
14:53.50MicC_hey guys
14:53.55*** join/#asterisk AlivesWrk (n=email@static-70-19-114-50.ny325.east.verizon.net)
14:54.20AlivesWrkwhat is the most used windows popup call manager software?
14:54.21rudi_and they're asking for something that is also managable by not-extremly-experienced admins
14:54.35AlivesWrki need something with caller id and possibly transfer/on hold for calls
14:54.40*** join/#asterisk gandhijee (n=gandhije@host-66-202-34-162.spr.choiceone.net)
14:54.52Hmmhesaysok this is the strangest thing evar
14:54.58Hmmhesayssome of the conferences work, some don't
14:55.00KattyAlivesWrk: mew?
14:55.08KattyAlivesWrk: popup call manager software?
14:55.13Beirdomorning once again, it seems
14:55.17KattyBeirdo: rehi
14:55.27BeirdoHeya, Katty.  Good morning
14:56.15rudi_hm no ideas at all? :>
14:56.29AlivesWrki am new to asterisk, yes
14:56.45KattyHmmhesays: mew? what is this popup call manager software stuff?
14:56.48rudi_i checked some of the links from the voip-info.org wiki
14:56.50AlivesWrkkatty: yes, just somethign that would interface with calls on a windows machine
14:57.07Hmmhesayscan I kill a conference from the clie?
14:57.09Hmmhesays*cli even
14:57.25MicC_hmmmhesays: do yu have al ot of pseudo channels not disappearing ?
14:57.32Hmmhesaysjust one
14:57.34MicC_yah..you can soft hangup it
14:57.43MicC_if it will let you
14:58.15MicC_is there a way to specify and external IP on the polycom IP501s? for Nat'ing ...etc?
14:58.44Hmmhesayshey file
14:58.51Hmmhesaysi've come back to the world of voip for awhile
14:58.57KattyMicC_: why would you want a phone on an external ip?
14:59.16fileHmmhesays: yay!
14:59.44MicC_Katty: umm....I wanna bring a polycom home with me :P
14:59.51*** join/#asterisk shido6 (n=shido6@d221-68-210.commercial.cgocable.net)
14:59.55MicC_I have firewall at home.
15:00.01*** join/#asterisk gvag11 (n=g@ppp30-adsl-61.ath.forthnet.gr)
15:00.03filenat=yes is your friend
15:00.08MicC_I got it working with Xlite...just don't see the option in the Polycomms.
15:00.22MicC_file that simple eh?
15:00.26fileyup
15:00.34KattyMicC_: try it. you may see. you may like it in a tree.
15:00.46fileKatty: excellent! gold star for you!
15:00.46KattyMicC_: or in a house with a mouse.
15:00.54Kattyfile: mew?
15:01.01MicC_I don't it all, you will surely have a mysterious "fall"
15:01.20MicC_err I screwed that up
15:01.46MicC_file: here is the fun part, my * is nat'd as well :P
15:01.58fileexternip and localnet
15:02.11MikeJ[Laptop]yes!
15:02.16Kattyis that something like a parsnip?
15:02.19MikeJ[Laptop]good morning mr. file.
15:02.24MikeJ[Laptop]Katty, yes... errr.. no
15:02.26filehi MikeJ!!!
15:02.36MikeJ[Laptop]FUN DAY!
15:02.38KattyMikeJ[Laptop]: you sure are chipper.
15:02.40MikeJ[Laptop]not really :(
15:02.44Kattyoh :<
15:02.50fileMikeJ[Laptop] is death!
15:02.50Kattyi'm mostly chipper!
15:02.55KattyMikeJ[Laptop]: i'm all grown up today.
15:02.59malverian[work]Is "Event: -" for a NOTIFY actually RFC compliant?
15:03.14malverian[work]My SNOM phone seems to see it as a bad event.
15:03.45malverian[work]The server sends this when doing a "reload" to tell the phone to resend SUBSCRIBEs.
15:04.01MicC_file: yah...it works great...just haven't done the polycom yet
15:04.34fileMicC_: nat=yes is pretty simple, it ignores the IP information in the SIP messages and uses the received IP and port
15:04.55MicC_kewlio...thanks file.
15:05.07MicC_so basically I would have no problem when I go home this weekend :P
15:05.13Kattyhmm.
15:05.17file...yeah
15:05.41Hmmhesaysso dynamic conferences work like shiat in an old version of aah
15:06.34MikeJ[Laptop]Katty, all grown up?
15:06.44MikeJ[Laptop]file, umm.. I'm a bastard arn't I?
15:06.50MikeJ[Laptop]it was yesterday wasn't it?
15:06.52BeirdoI don't ever wanna grow up
15:07.07Hmmhesaysi'm a toys r mine brat, I go running through the store yelling give me this and that
15:07.07mutilatori hope i'm never born
15:07.19Hmmhesaysi hope i'm never birthed again
15:07.26fileMikeJ[Laptop]: nah you're all peachy!
15:07.34MikeJ[Laptop]but the 6th right?
15:07.37BeirdoHmmhesays: I'm sure your mother would appreciate that
15:07.48fileMikeJ[Laptop]: the 6th what
15:08.38*** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com)
15:09.29e3ganyone who could help me in Linux commands???? :( learning Asterisk is not enough....
15:09.49AlivesWrkwhat do you need?
15:09.51AlivesWrk##linux
15:09.55PMantisI thought one-way audio was a SIP problem! Now, my IAXy's suddenly have this problem. Ideas?
15:11.14e3gneed to change the Primary IP address of my machine through SSH
15:11.17mishehuPMantis: an iax2 bug.  I had the same problem when I was running an earlier version of head (prior to 1.2.0b1)
15:11.40Beirdomishehu: sure it's not just missing firewall rules?
15:11.59marc324xlite --> asterisk --> pstn --  is this outgoing?
15:12.15*** join/#asterisk Alives (i=loot@cpe-68-173-215-206.nyc.res.rr.com)
15:12.24e3gmarc324: yep
15:12.34PMantismishehu: Weird, I was running this version of Asterisk for some time... (1.0.9)
15:12.35KattyMikeJ[Laptop]: today's my 21st.
15:12.35DarthClueoh file...
15:12.36fileTECHNICALLY it's both ;)
15:12.45MikeJ[Laptop]file can drink legally now too
15:12.47fileDarthClue: did you break it?
15:12.54Hmmhesaysin canadia
15:13.06MikeJ[Laptop]which just happens to be where he is
15:13.13filecrazyness!
15:13.22MikeJ[Laptop]yes
15:13.37DarthCluefile: remind me to buy you dinner next time i see ya, k?
15:13.39MikeJ[Laptop]carefull they'll stick
15:13.48bkw_file
15:13.50bkw_oh file
15:13.51bkw_call 42
15:13.56filebkw_: there you are!
15:14.04Kattyfile: bkw_ was hiding.
15:14.08Kattyfile: if by hiding i mean having breakfast.
15:14.18marc324I get "Cannot find extension context default" when attempting to make calls from xlite.
15:14.22PMantismishehu: And... it was working on the 3rd... I havne't even restarted * since then!
15:14.23Kattyfile: in his better homes and gardens kitchen.
15:15.47*** join/#asterisk olivier_ (n=olivier_@obs92-4-82-239-116-113.fbx.proxad.net)
15:16.04fileit's networking day
15:16.17Kattyhttp://kat.mandriva.com/ <- kat goes public.
15:16.29protienhm im trying to run this yet another wakepu script
15:16.31protienive got it all setup and
15:16.32malverian[work]I'm concerned about this NOTIFY stuff in chan_sip.
15:16.32protien<PROTECTED>
15:16.32protien<PROTECTED>
15:16.37protienis all i get when i dial it
15:16.42protienit just finishes before it starts
15:16.46malverian[work]protien, "agi debug" in the cli.
15:17.13DarthCluenew question...where are the c code functions documented?  and...how do i set the cdr field values from c code
15:17.19gandhijeei know this is a retarded question
15:17.44gandhijeebut how come when i include my local extentsion under default
15:17.49gandhijeei can't ring them
15:17.53*** join/#asterisk ManxPower (n=eric@adsl-70-247-221-174.dsl.lgvwtx.swbell.net)
15:18.08*** join/#asterisk fiber0pti (n=johndoe@pcp01876618pcs.sandia01.nm.comcast.net)
15:18.14protienmalverian[work], http://pastebin.com/386239
15:18.46ManxPowerIf I ever find the cisco programmer that decided to default async interfaces to 8 data bits, no parity, and TWO stop bits, I will kill them slowly.
15:18.47malverian[work]protien, It's apparently exiting immediately upon invocation.
15:18.55protienyeah man
15:18.56malverian[work]Open the script and add some debugging.
15:19.08protienthats above me
15:20.42marc324how do you make outbound call?  xlite -> asterisk    --what should I put in extensions.conf ?
15:20.54ManxPower~docs
15:20.56jbotfrom memory, 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
15:20.58KattyDarthClue: how's the bus?
15:21.02PMantismishehu: I just figured it out!  I have my IAXy provisioned with "codec: adpcm" as well as pcm. Created tons of CLI errors! Ugh.
15:21.49DarthCluehell, living hell.  but i'm managing almost 90 hours every 2 weeks which makes for a nice check and a really easy job.  kids suck.  well teenagers do.
15:21.59*** join/#asterisk groogs (n=greg@d221-73-237.commercial.cgocable.net)
15:22.08KattyDarthClue: whyfor?
15:22.16*** part/#asterisk PMantis (n=Miranda@66.251.89.34)
15:22.24groogsanyone here done the linksys PAP2 unlock thing?
15:22.37gandhijeemarc: you make your sips context in the extensions
15:22.42gandhijeeie [sip_peers]
15:22.43Kattygroogs: no
15:23.01KattyHmmhesays: have you even said hi to me this morning yet? :P
15:23.07protienim still having this annoy as problem, when i recieve calls via IAX from goiax.com i get this
15:23.10protienOct  8 00:49:42 NOTICE[24120]: chan_iax2.c:6690 socket_read: Rejected connect attempt from 204.13.233.114, who was trying to reach 's@'
15:23.13DarthCluethey are just bad kids.  the parents don't care, and the kids are just stupid.  they have no respect for authority (spitting on cops, throwing trash at the principal) and i can't do anything but drive em around.
15:23.14groogsdoesnt look very hard. $65cdn for 2 fxs...
15:23.19*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
15:23.20protienhow can i forward those calls to my extension? i cant work that out
15:23.25gandhijeethe do exten => exten#,priorty,Dial(SIP/<user>
15:23.28KattyDarthClue: ewww
15:23.44ManxPowerDarthClue, You are now a school bus driver?
15:23.50DarthCluekatty, it's better than flipping burgers, and pays better too.
15:23.50marc324gandhi -- yes, what do i put in it...  I have Dial(zap/1)
15:24.03KattyDarthClue: oh, right....i never did the burger flipping thing
15:24.14ManxPowerIsn't being a school bus driver the 3rd level of Dante's hell?
15:24.15BeirdoDarthClue: take corners extra hard?
15:24.17DarthClueManxPower: it's a part time gig that allows me to do consulting on the side.
15:24.32Beirdooh look, half the brats are on the floor... oooops
15:24.39DarthClueBeirdo: as hard as possible, but have to limit it to about 20 mph so the bus doesn't flip/roll.
15:24.42KattyBeirdo: now, now
15:24.49Beirdotrue enough, don't want to crash
15:24.53KattyBeirdo: someone's little girl is on that bus..and she's a good kid.
15:25.05BeirdoKatty: true.
15:25.07DarthClueManxPower: could be, but if that is true, then I must be king.
15:25.11KattyBeirdo: a quiet little 6 year old with pigtails.
15:25.25Beirdowon't somebody think of the children
15:25.27Beirdo:)
15:25.33Kattyexactly
15:25.39Kattyfile: mew?
15:25.40DarthClueKatty: no girls, just stupid little boys.  that's how bad these kids are, they have 2 buses, one for the girls (who are worse) and one for the boys.
15:25.45Beirdoyou are right of course, Katty, but that won't stop us from thinking of punishing the bratty ones
15:26.03Katty:<
15:26.06*** join/#asterisk gabb0 (n=gabb0@131.202.90.23)
15:26.10gabb0hello all
15:26.18Kattyif i ever adopt, maybe i'll send them to a private school
15:26.35DarthCluefile: how do i set the cdr user field?  any quick ideas on that?
15:26.42gabb0are the DS3000P out at all
15:26.43Beirdonot all public shools are that bad
15:26.51Beirdobut there seem to be a fair number that are
15:26.51*** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net)
15:26.57KattyBeirdo: but how would you know the difference?
15:27.05Beirdotrue
15:27.07KattyBeirdo: in my first day of school...
15:27.13KattyBeirdo: i thought my mother had abandoned me
15:27.16Beirdoand some private schools are hellish too
15:27.19KattyBeirdo: to a much of hideous monsters
15:27.27Beirdoawww.  Poor Katty.
15:27.31Kattys/much/bunch/
15:27.35tzangeryeah my kids get a good healthy dose of discipline.  I won't tolerate that kind of thing from my kids
15:27.38tzangerand they know it
15:27.40BeirdoI had my cousin there to help me not feel that way
15:27.42DarthCluethat is one advantage of driving, i can see which schools are bad/good.
15:27.45fileDarthClue: not particularly
15:28.09Kattytzanger: there's a fine like between discipline and overboard. i hope you're not crossing it.
15:28.11DarthCluefile: if you want dessert with that lunch, you need to think harder.
15:28.23Kattygoshdangitanyhow i can't type today.
15:28.29tzangerKatty: discipline means teaching self respect and others-respect.  overboard usually only happens when you do it angry
15:28.35tzangerI don't discipline my kids angry, ever
15:28.41Kattygood
15:28.46Beirdogood choice
15:28.47Kattyhopefully they won't turn out like me then :P
15:28.54ManxPowerI'll never have kids
15:28.54tzangerand it's very very very rare that I ever strike them, and if so it's usually on the hand (when they're little)
15:29.12Kattyi would never strike a child
15:29.13Kattyever
15:29.18Kattyfor /any/ reason.
15:29.19tzangerchildren are NOT little adults, that was one of the biggest fallacies in recent parenting doctrine
15:29.22tzangerKatty: that's wrong
15:29.30Kattytzanger: what's wrong?
15:29.35ManxPowertzafrir, exactly!
15:29.44ManxPowerChildren are unsocialized animals.
15:29.46tzangerwhen a child does something that puts them in immediate danger they need immediate and very clear correction
15:29.54Abbashello ManxPower
15:29.59Kattytzanger: hitting a child is not the answer.
15:30.07Kattytzanger: i am living proof of that statement.
15:30.09ManxPowerOne of the major jobs of a parent is to socialize them.
15:30.11tzangera smack on the hand gets their attention, does NOT hurt and snaps them out of whatever spinlock they have
15:30.21Abbascan u pls guide me about this  warning
15:30.31AbbasOct  7 10:24:43 WARNING[990]: rtp.c:1425 ast_rtp_bridge: codec0 = 13 is not codec1 = 256,
15:30.33Beirdoand if they are about to grab a pot of boiling water on the stove...
15:30.41Kattytzanger: if it doesn't hurt, then it's not a strike.
15:30.46tzangerAbbas: it can't transcode between codec 13 and codec 256
15:31.01tzangerKatty: they ACT like it's the most painful thing ever but that's normal :-)
15:31.14Abbastzanger    i am testing intel's 723
15:31.31ManxPowerYou don't talk to a dog when it does something bad.  Before a child learns language skills physical things are the only way to communicate
15:31.34protienis there any examples of how to forward all unused/bad numbers to a recoding
15:31.41Kattytzanger: i wouldn't recommend ignoring a child if they think it hurts.
15:31.53tzangersometimes a child's mind is in such a tight little loop that the only thing you can do to get them out of it is to do something physical.  (screaming their head off, inconsolable) -- you give 'em a little smack and their brain snaps out of it because it has to respond to the stimulus.  THEN you can begin to talk them down or reason with them if they're older
15:32.00Kattytzanger: but do what you want. it's their psychology you're playing with.
15:32.05tzangerKatty: you're overexaggerating
15:32.10Kattytzanger: i'm not.
15:32.14tzangeryou are.
15:32.16Kattytzanger: i lived it
15:32.20*** join/#asterisk zpn (n=xpn@gateway.digium.com)
15:32.22Kattytzanger: i'm not exaggerating
15:32.23tzangersmacking them on the hand is not "playing with their psychology"
15:32.28nfi|ermesi have some problems with voicebox
15:32.35tzangerKatty: if you were physically abused that is *much* different than what I'm describing
15:32.35Abbastzanger:  how can i avoid this warning
15:32.38Kattytzanger: striking them, then ignoring that they're in pain...
15:32.41protienmanxpower, is your example callback script, included in that tar of all your website?
15:32.44BeirdoI got spanked may times (deservedly) as a child, and I turned out just fine
15:32.53Kattytzanger: if it hurts the child, why do you do it?
15:32.54nfi|ermes-- Playing 'vm-review' (language 'it')   <---- but i listen in english
15:33.07tzangerKatty: you give them a swat on the hand and they recoil as if you branded them -- that's not hurting them.
15:33.07Kattytzanger: i don't understand that.
15:33.25Kattytzanger: did you bother to ask them if it hurt?
15:33.31ManxPowerprotien, Yes, but I don't provide ANY help for thost scripts
15:33.41protienthats fine, i just want to know which one it is
15:33.44protienso i can try it out
15:33.58tzangerI agree with corporal punishment for small children.  They do not understand words or reason because they themselves are not capable of the level of reason that is required.  Corporal punishment is NOT beatin gthe shit out of them because you're drunk and they're annoying
15:34.15*** join/#asterisk paryl (n=paryl@209.236.78.59)
15:34.19tzangerKatty: I know what hurts and what doesn't, and I am very much aware of the level of force I use, which is NOT a lot
15:34.24Kattytzanger: well if you don't think they can comprehend reason, then do you think they can comprehend why you're smacking them?
15:34.32tzangerthere is no mark left, no redness, no swelling, nothing.
15:34.43paryli can't find any working sites that have drivers for the tdm400p
15:34.44Kattytzanger: or just that you hit them, for no reason they could understand?
15:34.46ManxPowerKatty, They can't.  Neither can a dog, but it works for dogs.
15:35.01tzangerKatty: because animals (people included) respond to it.  It's not a way to teach them, it's a way to train them.
15:35.01ManxPowerMy cats know that if they get on the counter top they will get smacked.
15:35.12Kattytzanger: children are NOT dogs.
15:35.13tzangerKatty: you're again exaggerating.  i'm not beating them
15:35.16tzangerKatty: you're absolutely right
15:35.18protienmanxpower, which script should i be looking at?
15:35.19Kattytzanger: dogs should not be trained by hitting them
15:35.25Kattytzanger: and neither should children.
15:35.36tzangerbut before they understand reason they must understand basic rules and the most basic rule is cause and effect
15:35.37ManxPowerprotien, I don't recall.  It's been a year since I even looked at the script.
15:35.40Kattytzanger: dogs are trained by positive reinforcement.
15:35.40mutilatorKatty: sorry but yes, they do
15:35.50tzangeryou do something wrong, I will be upset.
15:35.59Kattyyou know what, we'll just agree to disagree
15:36.09Kattybecause there's nothing that you can say that will change what i believe
15:36.15Kattyand i will NEVER strike a child for any reason.
15:36.15Kattynone
15:36.17Kattyever
15:36.19Kattyend of story
15:36.25tzangerbut as I said, now that they understand reason I very very very rarely ever have to resort to corporal punishment
15:36.33KattyNEXT
15:36.35mutilatorheh
15:36.46tzangerbecause the rules are enforced and showing them how upset you are is FAR more painful to them than any smack
15:37.04Hmmhesaysanyone ever heard of "nailing" in the cisco voip world?
15:37.15mutilatornope
15:37.51tzangerKatty: you look at my children: very well behaved, very sociable, funny, friendly, happy but more importnatly: respectful to themselves and respectful to others.  This is a 5 and 4 year old too.  I think that my ex and I have done very very well so far
15:37.53Hmmhesaysyeah some guy just called me "cisco has a new technology called nailing, it doesn't require any digital signal processing, it just reroutes calls IP to IP"
15:38.06Hmmhesaysi'm like uhhhhhh dude, that sure sounds like a proxy
15:38.18tzangerthese children are not social misfits, they don't cower when someone raises their hand, they are not afraid of expressing their opinions or disagreeing with their parents
15:38.30Beirdosounds like what asterisk does if there's no codec change
15:38.31tzangerKatty: what you're describing is *very* different from what I am trying to describe
15:38.39paryloxymoron? "I think that my ex and I have done very very well so far"
15:38.44mutilatorbeating a child senseless is dumb
15:38.45Hmmhesaysstateful or stateless, he described a proxy
15:39.23parylwhere can i find tdm400p drivers?
15:39.38gandhijeecompile them... when you compile zaptel
15:39.45protienwhere are Asterisk::AGI and Asterisk::Outgoing available, i cant find them on google
15:39.52tzangernever striking your children is a laudable goal but I don't agree with it being the best solution ever
15:40.10*** join/#asterisk wunderkin (i=kev@12-219-162-233.client.mchsi.com)
15:40.18*** join/#asterisk Veto_laptop (i=mdkuser@cpe-66-69-38-192.satx.res.rr.com)
15:41.00iDunnotzanger: the best solution is to just not have children. works quite well ;)
15:41.03tzangerI will not argue with my children, I will not bargain with them.  They know the rules and they know that their parents have the final say, especially at these ages.  As they get older (it's already happened, but it'll happen much more as they age) they will be given more freedoms and more opportunities ot execercise their minds and their decision making abilities and ability to listen ot their inner voices
15:41.13parylgandhijee: i can't find the source to compile... is it packaged with the asterisk source?
15:41.26tzangerthat's the big one right onw, especially with my daughter.  She KNOWS she's doing things that aren't good but she still does them, I'm trying to teach her to listen to that inner voice because I know she has it
15:41.34gandhijeehttp://ftp.digium.com/pub/zaptel/zaptel-1.0.9.2.tar.gz
15:41.37gandhijeedownload that
15:41.45gandhijeethen once its compiled and installed
15:41.50gandhijeemodprobe wcfxs
15:41.54mutilatorya
15:42.00mutilatorelse your child will end up like my gf
15:42.07parylawesome, thats what i was looking for ;)
15:42.11nfi|ermes-- Playing 'vm-review' (language 'it')   <---- but i listen in english
15:42.19mutilatorpissed off at the world, doesn't do anything for herself, and doesn't care about much else but her
15:42.25mutilatorand not social at all
15:43.42*** join/#asterisk apardo (n=apardo@200.Red-83-50-234.dynamicIP.rima-tde.net)
15:43.49*** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1)
15:43.59tzangermy daughter is far more hurt when her daddy's upset with her than if her daddy ever hit her.  Guilt is a much bigger punishment, and wielded incorrectly it would be far worse than any kind of physical abuse I could give to her, but katty I don't think you agree
15:44.59Beirdofor sure.  the mental pain is way harder to judge too
15:45.12mutilatori don't think it is..
15:45.40mutilatoryou have to know a person for a bit, but everyone feels the same to an extent
15:45.54mutilatorless they're just plain nuts
15:46.39Beirdoheh
15:46.45Beirdohave a good sleep
15:46.45gandhijeeyep
15:47.21mutilatoractaully i have to get back to work
15:47.28mutilatormy fsckin other engineer quit
15:47.40filemutilator: well that bites
15:47.41mutilatorso i have like 2x the work and i was already too busy as it was
15:48.11mutilatorthis weekend is going to be heaven
15:48.22mutilatormy gf promised me bowling and sex tomorrow
15:48.24mutilator:P
15:48.45*** join/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp)
15:48.56mutilatorthink i'll turn the pager off and dissapear
15:48.57Beirdoway to go, mutilator
15:49.41mutilatori still havn't figured out my callerid problem
15:49.56mutilatorfristrates me
15:50.00mutilator*u
15:50.54tzangerbowling and sex?
15:50.58tzangerthat is a VERY odd combination
15:51.00iDunnoat the same time?
15:51.04iDunnothat's going to hurt.
15:51.19mutilatorwill be interesting tho no?
15:51.36mmmToop...has anyone here tried to install gnudialer?
15:51.52nfi|ermesanyone can help me with some problems with voicemailbox ?
15:54.51*** join/#asterisk miztic (n=gerard@rarcoa.com)
15:57.14*** join/#asterisk kaushal (i=kaushal@202.159.244.45)
15:57.40*** part/#asterisk kaushal (i=kaushal@202.159.244.45)
16:01.07*** join/#asterisk AsterNov (i=richard_@82-45-239-141.cable.ubr01.enfi.blueyonder.co.uk)
16:03.05*** join/#asterisk jdv79 (n=jdv79@u1057064.ul.warwick.net)
16:03.47jdv79if Monitor() a call and i hold down a DTMF tone constant, should i hear that or should it be reduced to a blip instead?
16:05.32*** join/#asterisk Qwell (n=north@24-50-66-194.vnnyca.adelphia.net)
16:11.44marc324i get Unable to create channel of type 'Zap' (cause 0)
16:12.40Hmmhesaysokrut?
16:12.47Kattyorkut
16:12.53Kattyi'm feeling dyslexic apparently
16:13.11iDunnoorkut, the fad of 2 years ago...
16:13.12Kattyand hungry. bye now.
16:13.20KattyiDunno: heh, i still have icq
16:13.24iDunnogoogle sponsored, wasn't it? kinda sucked arse ;)
16:13.33KattyiDunno: my 5 digit number FROM THE 5TH GRADE
16:13.35Beirdoit's still around
16:13.41iDunno3442444
16:13.42Beirdolinked closer to gmail now
16:13.49iDunnonope, I was in the 6 digit group.
16:13.54Kattybye now.
16:13.58iDunno5 digits is just scary ;)
16:13.59fileKatty: noooooooooo
16:14.01file:(
16:14.03iDunnobye bye Katty ;)
16:14.11iDunnohave a good rest-of-ya birthday
16:15.00*** join/#asterisk DeeJayTwo (i=deejay2@215-238.sh.cgocable.ca)
16:15.27jdv79anyone else have trouble passing audio like DTMF and stuff
16:15.31AsterNovI cant start asterisk@home manually using asterisk -vvvg, it scrolls some stuff and doesn't start up.
16:15.32Hmmhesaysbeverly hills
16:15.35Hmmhesaysthat's where I want to be
16:15.42HmmhesaysASterNov that is right
16:15.45Hmmhesaysdon't do that
16:15.53Hmmhesaysreboot your machine
16:15.53AsterNovwhy?
16:16.04Hmmhesaysstuff in aah has to be started in order
16:16.04*** join/#asterisk queuetue (n=queuetue@toronto-HSE-ppp4122670.sympatico.ca)
16:16.25AsterNovI dont want to use the asterisk_safe script
16:17.01AsterNovI want asterisk to fork on the 2.6 Kernel
16:18.04paryli'm trying to follow http://www.asteriskguru.com/tutorials/wildcard_tdm400p.html, but the setup in zaptel.conf is confusing me, and only refers to 1 card (i have 2)... is there a better tutorial out there?
16:18.07*** join/#asterisk iCEBrkr (i=icebrkr@rrcs-24-129-130-158.se.biz.rr.com)
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16:23.40opus__why does DTMF fail with SIP?
16:23.53filehave you specified a dtmfmode?
16:23.57opus__yes
16:24.05filedo they match on both sides?
16:24.09opus__yes
16:24.12*** join/#asterisk zetabug (n=zetabug@207.44.170.12)
16:24.12filehave you checked to see if DTMF is being received?
16:24.22rayvdHave you placed a middle-aged goat near a box of feathers?
16:24.42opus__yes its being received.
16:24.46opus__according to RTP debug
16:24.58opus__and Oct  7 09:24:11 DEBUG[32725]: rtp.c:251 process_rfc2833: - RTP 2833 Event: 00000002 (len = 4)
16:25.02fileso what are you trying to do that uses DTMF?
16:25.08*** join/#asterisk mhnoyes (n=mhnoyes@user-2ivfl1c.dialup.mindspring.com)
16:25.13opus__Dial an extension.
16:25.19opus__Works sometimes..
16:25.19fileusing what?
16:25.22fileDISA?
16:25.37opus__Nope, just WaitExten after my * box picks up from my provider
16:25.58opus__what do you use?
16:26.02fileBackground
16:26.29opus__it works sometimes, but not all the time
16:26.45opus__do you use 'ztdummy' ?
16:27.12filewhy would a timing device matter?
16:27.28opus__I'm thinking maybe
16:27.34opus__I've been trying to trace this problem for months
16:27.36nfi|ermesanyone can help me with some problems with voicemailbox ?
16:28.10*** join/#asterisk mcreedjr (n=mcreedjr@72.240.172.15)
16:28.15opus__For example, I just dialed out.
16:28.20opus__It dialed out correctly.
16:28.24opus__Now DTMF works fine
16:29.07mcreedjrDoes anyone have any ideas on troubleshooting choppy VoIP using a cable modem on one end and a T1 on the asterisk side of things. I'm not sure where to start with identifying where the congestion(?) lies.
16:29.20opus__mcreedjr - you need QoS
16:29.28*** join/#asterisk leszq (n=leszq@82.177.97.254)
16:29.34leszqhiiiiii allll
16:29.38nfi|ermeshi
16:29.42mcreedjrI have it setup on both ends..
16:29.47iCEBrkropus__: I don't use QoS.. But QoS would help :)
16:30.21mcreedjrOpus_: The T1 side prioritizes traffic by IP address in the edge router and the Cable side of things does layer 2 QoS by MAC address.
16:30.45opus__mcreedjr, double check everything. then tripple check it.
16:31.27filesilly people who don't answer the phone
16:32.22InfraRedphones are so 19,99
16:33.36jdv79where's iaxy prov stuff again?
16:34.45iDunnothis one was 50,00 (ish)
16:35.18*** join/#asterisk myiagy (n=myiagy@200.138.215.78)
16:35.21*** join/#asterisk viLeR (i=1000@66.128.47.232)
16:37.24*** join/#asterisk ke4qqq (n=chatzill@68-115-212-158.static.spbg.sc.charter.com)
16:42.13nfi|ermesanyone knows how to set language for voicemail registered messages ?
16:42.31christonunnite all
16:42.38nfi|ermes-- Playing 'vm-login' (language 'en')
16:42.48nfi|ermesen--->it
16:43.06InfraReddid you check the wiki?
16:43.11InfraRed~docs
16:43.15jbot[docs] Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk
16:44.41InfraRed~google prompt language site:www.voip-info.org
16:44.49nfi|ermesthx
16:44.50nfi|ermes:|
16:45.00InfraRedcheck that
16:45.06InfraRedit's 3rd result
16:45.10InfraRed+on
16:46.46*** join/#asterisk dalfry (n=dalfry@gateway.ishisystems.com)
16:48.52*** join/#asterisk tsetane (n=tsetane@87.252.68.0)
16:50.30marc324a context [ccc] in extensions.conf has to match a context in zapata.conf ?
16:51.08iCEBrkrSo where can I find a changelog or whatsup file for 1.2.0 beta1?
16:51.18iCEBrkrIt'd be nice if there were a link to a changelog on asterisk.org
16:54.09*** join/#asterisk fordvoice (n=chrisf0r@rrcs-70-61-133-91.central.biz.rr.com)
17:00.56jdv79Oct  7 12:59:31 WARNING[20788]: chan_iax2.c:9434 load_module: Unable to open IAX timing interface: No such file or directory
17:01.07jdv79is that a real issue?
17:01.53jdv79Oct  7 13:00:22 WARNING[20808]: res_musiconhold.c:827 moh_register: Unable to open pseudo channel for timing...  Sound may be choppy.
17:01.56jdv79or that one:)
17:01.59opus__yes
17:02.01opus__it can be
17:02.14jdv79how can i fix it - there is no digium hw in the box
17:02.43bkw_ztdummy
17:03.07jdv79thanks
17:03.22*** join/#asterisk justinu (n=j2@72.18.13.40)
17:03.27*** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net)
17:05.07*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
17:07.22protiencan an extension, call a number from another context
17:08.10enderprotien: depends on what context you list in sip.conf
17:08.22enderprotien: they'll only be able to dial contexts reachable by that context.
17:08.26protienive got a sip on default context
17:08.34protienand i want it to call another context
17:08.35protienpossible?
17:08.43protienvia an extension?
17:08.43enderprotien: so then they'll be able to reach all contexts that are included in default
17:08.51justinuinclude => othercontext
17:08.55protienoh
17:08.56justinuin default
17:08.58pifiuhow do you set music on hold up in extensions.conf?
17:08.58enderyep
17:09.10enderprotien: it's on by default, you'd have to turn it off IIRC
17:09.10protienbut how do i then
17:09.19enderprotien: you just dial it.
17:09.30protienif ive got two sets of s,
17:09.33enderprotien: if the digits dialed match any ext that is in a context you i nclude, i twill dial it.
17:09.34protienwhich one is used
17:09.40enderprotien: none.
17:09.47enderprotien: s is special, for entering that context.
17:10.04SwK[Work]is opencall.org b0rked?
17:10.06protienlet me tell you what i want to do
17:10.08*** join/#asterisk sahafeez (n=sahafeez@67.109.14.227.ptr.us.xo.net)
17:10.10protienand maybe you can give me the best solution
17:10.15protienim building a basic menu system
17:10.19enderprotien: you could have an extension, which has 'Goto(Context,s,1)
17:10.21protieni want to know the best way to call it up
17:10.32protienoh thats possible is it
17:10.41enderprotien: exten => 5555,1,Goto(mainmenu,s,1)
17:11.15protiendo i still need to include => mainmenuu
17:11.19protienin default context then
17:11.23enderno
17:11.31enderGotos can jump contexts.
17:11.40protienah then thats a good solution
17:11.41protienlemme try it
17:12.10protien:>
17:12.12protienvery nice ender
17:12.25enderyour welcome
17:12.25protieni can call park into that aswell right?
17:12.34enderhrm.
17:12.37enderpark is different.
17:12.47enderpark will take an unused park ext in a range.
17:12.56enderprotien: you could blind transfer somebody to that ext.
17:13.01protienah okay cool
17:13.12protienanother thing i wanted to know is
17:13.19protienif i have a list of phones numbers
17:13.25shido6I recommend, dentibone!
17:13.29protieni want banned, like sent to a certain extension
17:13.35protienhow would i go about that, for say 10 numbers
17:14.05enderar ethey numbers you own, or are they numbers being called from?
17:14.18protienwell numbers called from
17:14.23protienthat are calling into my pbx
17:14.24marc324the context=aaaa  in zapata.conf has to match a corresponding [aaaa] in extensions.conf ??
17:14.26protienweither it be from voip or pstn
17:14.54justinuexten => did/callerid,1,Goto(banned,s,1)
17:14.56endermarc324: the context in zapata is where incoming calls will start in your extensions.conf
17:15.14protieni have to enter that for each number?
17:15.30justinuyou could create a macro that checks all inbound calls
17:15.37marc324ender-- therefore a [context] has to be present in extensions.conf ?
17:15.47endermarc324: yes, it should be.
17:15.57enderprotien: 10 numbers isn't that much.
17:15.57marc324how about for outgoing calls?
17:16.03protientru ender
17:16.38endermarc324: outgoing calls care not about the context listed in zapata.conf
17:16.51enderprotien: plus if htey are in a specific range, you can do wildcard matching.
17:17.09ender_206295426X
17:17.23protienexten => did/_8270XXXX,1,goto(banned,s,1) ?
17:17.33enderthat about does it.
17:17.35justinuyep
17:18.00enderalthough thats a really odd phone number
17:18.18protienthats australian
17:18.22enderoh ok.
17:18.29protienim just doing a quick test
17:18.37protienmy local sip is 1001, i tried banning that
17:18.40*** join/#asterisk mcreedjr (n=mcreedjr@72.240.172.15)
17:18.44protienor doesnt it work local?
17:18.51enderit works local
17:19.20enderthe CallerID if your Sip is most often the [foo] part of the sip.conf entry
17:19.57mcreedjrCan anyone point me on where to change the SIP jitter buffer in asterisk?
17:19.57protienyeah
17:19.59protien1001 is mine
17:20.14endermcreedjr: searched the wiki?
17:20.15protienexten => did/1001,1,goto(menusys,87,1)
17:20.16protienis my entry
17:20.28enderhrm, the 'did' is the target extension.
17:20.33enderor hte Direct Dial
17:20.44justinudirect INWARD dial ;)
17:20.55mcreedjrender: Searched for "jitter buffer" in the Wiki and found articles about what the jitter buffer is, but not where to change it.
17:20.56enderso if you wanted to block 1001 from ever calling 1005, you would put 1005/1001,1,goto(
17:20.57protienoh
17:21.07protiencan i go
17:21.25protien_./1001,1,goto
17:21.29protienfor *
17:21.39enderyep
17:21.46QwellYou probably want _X.
17:22.16endermcreedjr: hrm, isn't sip jitter buffer a phone level thing?  I know you can adjust IAX2 jitter buffer...
17:22.35mcreedjrEnder: I don't know, thats why I'm asking :)
17:22.46mcreedjrEnder: It's beginning to look that way, just trying to verify that for sure.
17:22.54enderah.
17:26.27*** part/#asterisk joelsolanki (i=joelsola@202.160.161.93)
17:28.08*** join/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
17:28.30redder86~seen coppice
17:28.37jbotcoppice <n=chatzill@127.202.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 4d 17h 2m 59s ago, saying: 'dudes: ghostscript is the usual suspect'.
17:28.45redder86still in India
17:29.14*** part/#asterisk redder86 (n=lee@gateway.howardsilvan.com)
17:31.19harryvvin india for what
17:32.38justinuno kitty, that's mah pot pie!
17:33.03Kattykats do not eat average pot pies.
17:33.08Kattythey have meat and cheeses and things
17:33.09*** join/#asterisk paulc (n=Paul@216.187.75.190.novuscom.net)
17:33.13justinuno kitty, that's a bad kitty!
17:34.19*** join/#asterisk Hupe (n=Hupe@iD4CC1003.versanet.de)
17:34.48Kattyactually, i have an interesting little issue.
17:35.01Beirdojustinu: no need to say the next line :)
17:35.04Kattyafter x ammount of calls, these polycom 500s cannot hear anything.
17:35.06*** join/#asterisk [Jedi] (n=hhgds4@154.Red-217-127-168.staticIP.rima-tde.net)
17:35.08[Jedi]Hello
17:35.12justinubeirdo: damn...
17:35.23Kattythey can call in....the person on the other end can hear us.....and after rebooting the phone......it's fine.
17:35.25[Jedi]Anyone can help me with SpanDSP and TxFax please?
17:35.30harryvvkatty, since when do 500s hear?
17:35.40Kattyharryvv: since katty was born, obviously.
17:35.41Qwellharryvv: latest firmware
17:35.49gandhijeehey ingorepat => number is supposta give a tone
17:36.00gandhijeeafter you hit the number right?
17:36.01Kattymaybe a routing issue?
17:36.02Qwellgandhijee: no, its supposed to not remove the tone, iirc
17:36.05Kattya network hiccup?
17:36.06justinuRTP send on out out of range udp port?
17:36.16gandhijeeQwell: thats what i meant
17:36.24gandhijeeso is it broken in 1.0.9??
17:36.28KattyHmmhesays: i bet you would know (=
17:36.32Qwellgandhijee: don't think so
17:36.36[Jedi]anyone with TxFax ?
17:36.56harryvvso has the fax issue with voip been corrected?
17:36.56justinucan you capture an invite/200OK exchange with SDP on a call that doesn't get RX audio?
17:37.10paulcJerJer: You around?
17:37.27KattyHmmhesays: right. when you come back. these polycom 500s are ruining my 21st birthday.
17:37.35KattyHmmhesays: i demand you stabity them with your hair.
17:37.51BeirdoKatty:  happy birthday, BTW
17:37.56KattyBeirdo: thanks (=
17:38.17Beirdohehe
17:38.25Kattymew :<
17:39.05Beirdouse the claws on it, maybe it will smarten up
17:40.20justinuis it possible that the polycom's have an upper RTP port limit, and that asterisk eventually starts sending the RTP to the polycom above that limit?
17:40.26endergandhijee: your phone is more likely to handle the tone.
17:40.29gandhijeethen is there something wrong with my inore pat
17:40.35endergandhijee: the ignorepat is more for analog phones.
17:40.39gandhijeehttp://pastebin.com/386351
17:40.54gandhijeeender: i know, i have it setup and it seems right to me
17:40.56endergandhijee: are you using a sip phone?
17:41.10gandhijeeshow dialplan shows that its there
17:41.17gandhijeea buncha dif phones
17:41.27[Jedi]anyone with TxFax ?
17:41.35gandhijeei have 3 analogs off a TDM400, an IAX softclient and SIPs softclient
17:41.50endergandhijee: well, w/ sip phones, the digitmap on the phone itself will take care of the tone generation.
17:41.55gandhijeeok
17:42.08gandhijeewhen i hit 9 or 8 on my analogs
17:42.15gandhijeei lose the tone, but i can still dial
17:42.23endergandhijee: and your analog phones, do you have the right context= in zapata.conf so that they are by default sent to the context that has the ignorepat ?
17:44.08KattyBeirdo: nothing in the cli about my errrr :<
17:44.19Beirdohmmm, that sucks
17:44.31Kattyi really think it's a routing issue
17:44.33Beirdodon't let it ruin your day :)
17:44.40Kattypfft
17:44.53Kattyi have to reboot my phone 8 times a day
17:44.55justinui gave you some suggestions, but you don't listen
17:45.23Kattyjustinu: if you're talking to me, you need to let me know.
17:45.37Kattyjustinu: with as many conversations as go on in here...heh, it's hard telling who you're talking to.
17:45.38gandhijeeender: i thikn so
17:45.46justinuwho else was asking about polycoms not receiving audio?
17:45.47justinuder
17:46.08Kattyjustinu: i don't read every line in this channel.
17:46.13[Jedi]I'm the only one in the world trying to use txfax in asterisk 1.1 or 1.2?
17:46.20Kattyjustinu: in fact, i'm not even in here most of the time (=
17:46.23justinuoh well
17:46.25Beirdo:)
17:46.34gandhijeeender: i have a dial_pstn context
17:46.35Kattyjustinu: i'll just wait for Hmmhesays or anthm
17:46.41Kattyjustinu: they speak kat well.
17:46.43justinulol
17:46.45Kattyjustinu: you do not.
17:46.52justinuwhatever you say
17:46.56Kattykthx
17:46.59*** join/#asterisk jeffgus (n=jeffgus@2002:d856:c704:0:0:0:0:1)
17:47.02gandhijeeender: that i include in my analog_handsets context, which is what is defined in my zapata.conf
17:47.05endergandhijee: and in zapta.conf for the lines that you have analongs plugged into, they have 'context=dial_pstn' ?
17:47.22*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
17:47.33endergandhijee: try shortcutting the include and go straight to the dial_pstn
17:48.11*** join/#asterisk bsd3 (n=bsd@203.134.193.168)
17:48.32gandhijeeender: no in zapata.conf the handset lines are def a zap_lines
17:49.02gandhijeeender: then there is a dial_pstn context to access theat
17:49.19gandhijeewhich is included in under a section called dialables
17:49.28gandhijeethen dialables is included in zaplines
17:50.24*** part/#asterisk bsd3 (n=bsd@203.134.193.168)
17:50.37*** join/#asterisk FABRIZIOxxx (n=FABRIZIO@81-208-26-86.ip.fastwebnet.it)
17:50.47mutilatoranyone ever admined a teamspeak server?
17:51.08marc324~docs
17:51.09jbotdocs is, like, 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
17:51.42endergandhijee: so again, try shortcutting all that including mishmach and go straight to the context that has the ignorepat
17:51.51FABRIZIOxxxhi guys .. when i call through a zap channel i'm getting a very soft sizzle on the background .. everything else is ok .. could this be caused by bad and/or long cables ..?? I couldn't find anything on voip-info.org regarding this problem ..
17:52.03enderanybody smell bacon?
17:52.07justinulol
17:52.07[Jedi]anyone with TxFax ?
17:52.11[Jedi]:(((((
17:52.12*** join/#asterisk Godsey (n=lanny@pdpc/supporter/sustaining/Godsey)
17:52.18justinuender: it's sizzlean
17:52.26Katty[Jedi]: sorry, i don't know anything about that.
17:52.27bjohnsonFABRIZIOxxx: maybe try adjusting gains to turn down volume
17:52.34bjohnson[Jedi]: yes.  but not me
17:52.35gandhijeeender: ok
17:52.44[Jedi]I'm going crazy
17:52.46[Jedi]with it
17:52.54bjohnsonsounds very appealing
17:52.56gandhijeeender: but like i said, under the show dialplan they ingnore pat shows up
17:53.13FABRIZIOxxxbjohnson, already tried .. but i'm getting complaints about low voice volume .. and at the moment the tx and rx are set to 3.5 .. i don't think its very hig ..
17:53.18[Jedi]TxFax gets executed but does nothing
17:53.38bjohnsonlow voice at which end?
17:53.50bjohnsonthe rx end?
17:53.57gandhijeeanyone also know where i can get a hold of some decent sips phones for kinda cheap
17:54.05gandhijeenone of those grandstream craps
17:54.09bjohnsonsip?  never heard of it
17:54.10FABRIZIOxxxyes .. people from outside my * server
17:54.23gandhijeewell how about an IP phone then
17:54.29bjohnsonFABRIZIOxxx: increase tx and decrease rx
17:54.41bjohnsongandhijee: you pee?
17:54.46FABRIZIOxxxok .. i'll try straight away ... thks
17:54.50endergandhijee: voipsupply/cdw Polycom phones.
17:55.41gandhijeeso i guess voipsupply is the cheapest, nuts
17:55.45bjohnsongandhijee: all the cheap variants are reported to feel cheap .. what is "decent phone" to you
17:56.10justinuum, atacomm sells polycom 501s for cheaper than voipsupply
17:56.11justinu169
17:56.47gandhijeei have one of those phones from iareaphone. doesn't feel cheap
17:56.52paulcHas anyone had problems with failed call attempts through NuFone in the past couple of hours? I've had a customer complaining about congestion + audio quality problems but have been unable to replicate myself.
17:56.55*** join/#asterisk halogen8 (n=halogen8@66-146-190-146.skyriver.net)
17:57.53halogen8I'm looking at the OEM X100P Digium cards on ebay, and wanted to know what people thought of them?
17:58.05halogen8are they absolute garbage?  do they work ok, or great?
17:58.16gandhijeei found a place on the net that sold them for 10 bux
17:58.20gandhijeelike 15 with shipping
17:58.30gandhijeeworked fine for me
17:58.41halogen8gandhijee: you have a link to that place?
17:58.45*** part/#asterisk n0where (n=kc@q041140.ppp.asahi-net.or.jp)
17:58.50gandhijeei can try and google it back up
17:58.58gandhijeei bought it about a year ago
17:59.08gandhijeegive me a min
17:59.19gandhijeei gotta order some ip phones and ata'
17:59.21gandhijees first
18:01.29FABRIZIOxxxbjohnson, actually its quite better .. but i can hear still a little bit ... i'll check the cables and buy some new ones since the ones i have now are ahdn made .. and about 25 meters long..
18:02.17Kattyhttp://groups.google.com/group/Asterisk-users/browse_thread/thread/91e6002e0ed66504/704f829272d6d75f?hl=en#704f829272d6d75f <- mine issue.
18:02.24halogen8anyone know where i can find the diff between asterisk@home 1.5 and the 2.0 betas?
18:02.43[Jedi]bkw_ are you there? got a min?
18:03.12gandhijeeFABRIZIZOxxx: make sure your not running your phone line parallel to any power lines
18:03.30gandhijeeand a ferite core to wrap your line around wouldn't hurt either
18:03.43Qwellhalogen8: They are absolute garbage - and they ARE NOT Digium cards.
18:03.55Kattyjustinu: looks like the port is 2226 on rtp debug
18:04.07FABRIZIOxxxgandhijee, thks for the info .. i'll definately check that
18:04.27znoGQwell: not sure about "absolute garbage", I have one and it works fine.
18:04.32justinukatty: is that what port the pbx is sending audio to you on?
18:04.44QwellznoG: Would you use one in a production system?
18:04.49halogen8Qwell: why do you say they are garbage?  what problems have you had with them?
18:04.51QwellI most certainly would not.
18:04.52Kattyjustinu: both directions
18:05.14justinukatty: is this a call you didn't receive audio on?
18:05.19Qwellhalogen8: They're unsupported, for one.
18:05.30Kattyjustinu: no
18:05.31Qwelland they're just...cheap
18:05.35Kattyjustinu: i enabled rtp debug
18:05.35[Jedi]Zaphfc bri's are unsupported too
18:05.39Qwell"you get what you pay for"
18:05.40[Jedi]and work perfect
18:05.44[Jedi]and cost 15 EUR
18:05.48justinukatty: my guess is that subsequent calls will incrment the port number
18:05.55Kattynow it's 2228
18:06.01Kattyjustinu: mew?
18:06.13halogen8Qwell: yeah, but right now, i can't really justify spending alot on a FXO card......unless those OEM cards just won't work
18:06.23gandhijeeQwell: from the hardware standpoint they are the exact same thing
18:06.25Qwellhalogen8: it's hit or miss
18:06.32halogen8Qwell: I've heard they shouldn't be used for production, but I just wanna know what problems people have had with them
18:06.36gandhijeeQwll: on production, yeah don't use them
18:06.40znoGQwell: no, hell no, this is for my home system. You never asked halogen8 what kind of system he was putting it in, did you?
18:06.41*** join/#asterisk MnDBnDr (n=MnDBnDr@198.234.224.6)
18:06.50gandhijeebut if you are just tryin to get your hands wet, its the way to go
18:07.09gandhijee10 bux v 100 bux....
18:07.16halogen8Yeah, I'm just using this as a learning experiment.....can't justify spending alot on it yet
18:07.18Kattytoo confusing. not sinking in.
18:07.24gandhijeeyeap
18:07.30znoGhalogen8: call quality, echo problems, volume issues.. mine seems to work fine apart from maybe a slight volume problem and occasional caller ID issues, but I can live with that.
18:07.31gandhijeei did the same thing halogen
18:07.38justinukatty: my guess is that eventually when the port eventually gets above a certain number, you will stop getting audio
18:07.50MnDBnDrcan someone point me in the right direction for a good doc to configure oh323 on AAH?
18:07.54gandhijeethe asterisk card just has a resistor moved to change its ID number. thats all
18:07.56halogen8znoG: thanks for the examples......
18:08.09iCEBrkrznoG: Yea, same here, I don't get consistant volume with calls.. but that's my only problem.
18:08.21justinukatty: when you reset your phone, you're causing the phone to start at the lower part of the port range again, thus making it work
18:08.22gandhijeehalogen8: i don't seem to have a problem with mine
18:08.49halogen8it just seems, it would be a good card to learn on, then when it goes to production, use the $100 real digium card
18:08.51gandhijeenone of the issues these guys are talkin about, then again my line might be "cleaner" than theres
18:09.14halogen8gandhijee: thanks.....i think i'll try it.....cus its just so cheap
18:09.24gandhijeehalogen8: if you go in to production, get the TDM400P, that way you can expand if u just use POTS lines
18:09.27iCEBrkrgandhijee: I used mine for years.  It was fine.. But I did have an few volume problems.  Just had to tweak my gain
18:09.40Kattyjustinu: i see.
18:09.51Kattyjustinu: there's no port range specifications on the phone's settings though
18:09.51gandhijeeiCEBrkr: i've only had mine for about a year
18:10.08iCEBrkrIt worked well enough that ALL my calls went through my asterisk box.
18:10.09justinukatty: hmm... you can change the rtp ports asterisk uses in rtp.conf, but I'm not sure if that'll help you.
18:10.14iCEBrkrSo I really guess I can't complain
18:10.14halogen8gandhijee: yes, that is the card I would like......but its just too costly right now......looks like a great card though
18:10.25Kattyjustinu: i don't even know it's rtp causing it.
18:10.31Kattyjustinu: i'm not going to go poking about my conf files.
18:10.42justinuwell, one way audio is usually an RTP problem
18:10.52Kattyjustinu: i'll ask elsewhere first.
18:10.57justinuif you can't hear, you're not receiving the RTP stream correctly.
18:10.58Kattyjustinu: second opinions good.
18:11.16gandhijeehalogen8: it is =) i just got mine yesterday
18:12.09halogen8gandhijee: how much did you pay?  I found this:  http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-45624523776.htm
18:12.18halogen8looks pretty cheap without the adapter cards
18:12.30MnDBnDrcan someone point me in the right direction for a good doc to configure oh323.  It is compiled and installed.
18:12.39gandhijeeyeah it is, add bout 70 bux i think for each module
18:12.52gandhijeemy card cost 274, 1 FXO 3 FXS
18:13.10MnDBnDrjust got my hands on 3 symbol h323 phones.  802.11b
18:13.45halogen8yup, thats looks like a good price
18:13.52halogen8i think i just want one with 2 FXO ports
18:13.57gandhijeeanyone know of a decent ATA too?
18:14.01halogen8with a way to add 2 more FXO cards
18:14.22gandhijeehalogen8: if you are just playing, get the 2 clones
18:14.38gandhijeehalogen8: and just use softphones for now. till you are comfortable
18:14.54halogen8gandhijee: yes.....I will get the clones to learn, then if they ever go to production, I will get the good one
18:15.16gandhijeehalogen8: good idea
18:15.37halogen8gandhijee: I have 5 Linksys PAP2 adapters that I can use with it.....so don't need softphones
18:16.00halogen8gandhijee: staples just had a great deal on the PAP2's....they were $50/ea with a $50 rebate
18:16.14*** join/#asterisk DagMoller (n=DagMolle@2001:5c0:8fff:ffff:0:0:0:2b)
18:16.23Qwellhalogen8: locked for vonage
18:16.27gandhijeehalogen8: aren't they tied to like vonage though
18:16.34halogen8gandhijee: easily unlocked
18:16.41gandhijeesipath??
18:16.42halogen8all of mine are unlocked now
18:16.46justinui wish I could get the rtp300 unlocked
18:16.47Qwellsomebody needs to toss me one of those pap2's, heh
18:16.49gandhijeewhat you do to unlock them?
18:16.49DagMollerasterisk -r without colors in CLI, why?
18:17.16gandhijeecuz if thats the case, i'll run to worstbuy right now and buy 2
18:17.21halogen8gandhijee: lemme find you the forum that talks about it......its actually quite easy.....if you get it before VONAGE updated the firmware
18:18.06halogen8http://www.broadbandreports.com/forum/remark,14450684
18:18.11syledagmoller: cause your most likely not starting asterisk with -vvvgc options
18:18.51gandhijeesweeet
18:18.52gandhijeethanx man
18:18.55halogen8gandhijee: I dunno if they still have the rebates or what, but they are only about $50 without the rebate.......
18:19.06gandhijeeyeah, that better than the other prices.
18:19.08halogen8gandhijee: sure, enjoy.....thanks for the info......gotta get back to work....ttyl
18:19.14gandhijeelater
18:20.41DagMollersyle, -c if for console, and i dont want console at start time!!
18:20.51sylewhy not
18:20.54DagMollersyle, sorrry my english!
18:20.56sylei just start it in a screen session
18:22.16DagMollerbecause asterisk can be run as a daemon.
18:22.37syleup to you man
18:23.49*** join/#asterisk syle2 (n=blag@unaffiliated/syle)
18:24.17DagMollersyle, its not make sense!
18:26.48*** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn)
18:27.33syle2dagmoller: /usr/bin/screen -L -d -m -S asterisk /bin/nice -n -19 /usr/sbin/asterisk -U asterisk -vvvgc ...that make more sense to you?
18:28.09DagMollerlol
18:28.11*** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net)
18:30.30MnDBnDrcan someone point me in the right direction for a good doc to configure oh323
18:37.39*** join/#asterisk azzie (n=az@azzie.net)
18:38.17enderum.
18:38.23enderwhy put asterisk into screen?
18:38.28marc324what port does sip use?
18:38.41enderasterisk has a console you can attach to w/ -r
18:39.16endermarc324: a bunch of them.
18:39.25marc3245560?
18:39.45ender5060 is the main one.
18:39.46enderudp
18:40.41MicC_anyone try backing up and restoring voicemails to a new system?
18:40.50MicC_does some strange stuff.
18:41.13MicC_won't let you record a voice mail on the new system.
18:41.32jontowmicc; ?
18:41.46MicC_its hard to explain.
18:41.56MicC_I have 2 * boxes...1 primary and 1 backup
18:42.00jontowshouldn't be.. the format is pretty..reasonable
18:42.01groogsMicC_: permissions
18:42.10groogsMicC_: vm files are just that .. files
18:42.22DagMollerender, why in -r mode dont have colors?
18:42.23enderperms on the directories for VMs
18:42.24MicC_groogs: that was my first thoguht....I tar''d them and verified perms
18:42.30jontowthe files won't differ from box to box, there isn't anything unique about 'em..
18:42.30enderDagMoller: hrm, dunno.
18:42.52jontowwhen you ran tar, did you give it the 'p' option? (both when creating the archive and when extracting it?)
18:43.11groogsMicC_: well, there's nothing else to it. if the permissions are right and you can't record vm, it has nothing to do with backing up / restoring .. you probably can't record ANY voicemails, for whatever reason
18:43.33MicC_jontow: to keep the permissions. No not really. I have never had a problem having perms changed on an extract
18:43.44jontowdouble check it, then
18:43.47groogswell, what ARE the permissions?
18:43.48jontowall files, all directories
18:43.55enderMicC_: specifically user and group ownership.
18:44.01MicC_groogs: if I blow away all the voicemails it works.
18:44.09groogs(ownership, and mode) .. and what user/group is asterisk running as?
18:44.16MicC_ender: boxes are exact mirrors of each other UID and GUID :P
18:44.45groogsi guarantee its a permission problem because well, it can't be anything else
18:44.57*** join/#asterisk brookshire (n=matt@gateway.digium.com)
18:44.59groogscheck permisions on the voicemail directories and their parents
18:45.07jontowjust isn't much to fuck up with the voicemail structure.. its solid
18:45.10MicC_I came to the conclusion it must keep a counter (msg.001) somewhere in the databas(mysql backend)
18:45.37MicC_jontow: it seems like a no brainer...backup...then restore.
18:46.20MicC_my goal is to eventually store the voicemail from both the primary and backup on the same SAN device.
18:46.37MicC_that will help me make a seamless failover from one box to the other.
18:51.44MicC_I will find the culprit and expose it...If its DFU (me) you might now hear anything :P
18:51.53MicC_now=not
18:54.17*** join/#asterisk cio (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net)
18:54.57cioHi all.  When a remote sip phone registers on say port 63686, is that an RTP port?
18:55.12*** join/#asterisk arguile (i=user224@66.38.201.234)
18:56.15zetabugso I got my extensions working, to get behind a NAT.  Now how do I make it so I hear a ring ring, until they pick up the line on their end?
18:56.17*** join/#asterisk kue (n=Administ@CPE-70-94-56-196.wi.res.rr.com)
18:56.22zetabug(sip)
18:56.56zetabugthis is what I have: 'exten => 6666,1,Dial(SIP/bob,,rm)'
18:57.10zetabugand it works. but no ringing :(
18:59.15*** join/#asterisk pc2 (n=pc@209.151.52.81)
18:59.36*** part/#asterisk kue (n=Administ@CPE-70-94-56-196.wi.res.rr.com)
18:59.49pc2Ok -- what does an "IXC interconnect" mean as far as telephone lines go?
19:03.28*** join/#asterisk cio_flood (n=na@adsl-072-149-159-016.sip.bhm.bellsouth.net)
19:03.54cio_floodWhen a sip phone registers on a high port, like 62823 or something, is that a Dynamic Nat port or a RTP port?
19:05.06cio_floodHere's what I'm getting: NOTICE[1524]: chan_sip.c:8034 sip_poke_noanswer: Peer '804' is now UNREACHABLE!
19:05.17*** join/#asterisk harryvv (n=none@S010600a0c93f6f7e.vs.shawcable.net)
19:05.18cio_floodI get this immediately after it registers.
19:06.49cio_floodAnyone even here?
19:09.27marc324what port does xlite use?
19:09.31*** join/#asterisk wolfson (n=hehe@usr-kdh-208-6-58-26.beachlink.com)
19:10.03*** join/#asterisk kue (n=Administ@CPE-70-94-56-196.wi.res.rr.com)
19:10.13paulcHas anyone had problems with failed call attempts through NuFone in the past couple of hours? I've had a customer complaining about congestion + audio quality problems but have been unable to replicate myself.
19:10.47hardwiredo you qualify your customers?
19:11.25*** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com)
19:12.56*** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn)
19:13.18*** join/#asterisk razu_ (n=razu@213-35-173-39-dsl.prn.estpak.ee)
20:44.09*** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt)
20:44.09*** topic/#asterisk is Asterisk 1.2.0 Beta1 - http://www.asterisk.org || Astricon 2005 - Anaheim, CA - Oct 12-14 - http://www.astricon.net/2005 - Sign Up Now!
20:44.10QwellKatty: I was mostly just joking, until you asked. :p
20:44.17Kattywhat a weirdo
20:44.36Qwellpsuedo-meat is like...that fake sausage they put on pizza
20:44.44Qwellsoy protein or whatever
20:44.49Kattyit's real sausage
20:44.51Kattyi checked.
20:44.58Qwellnah, veggie pizza
20:44.59Kattyjust not the grade you're used to
20:45.05Qwellmy friend bought some one day
20:45.14Kattymew?
20:45.20Kattyveggie pizza shouldn't have meat on it
20:45.22Qwelltasted just like sausage and pepperoni...but it wasn't
20:45.25Kattymost pizzas still aren't vegan.
20:45.48file[laptop]TastyKatty(tm)
20:45.59Kattyfile[laptop]: mew? ^_-
20:46.05QwellKatty: So, what don't you eat, exactly?
20:46.17KattyQwell: don't eat meat, dairy, honey, etc.
20:46.22Qwellhoney?  wow
20:46.29KattyQwell: no animals or animal biproducts.
20:46.37Qwelldo you wear cloth?
20:46.39KattyQwell: some animal biproducts, if free range, i will eat
20:46.51Aliveshow do you place a call on hold in asterisk?
20:46.54KattyQwell: no wool or angora fur or leather, etc.
20:47.37KattyQwell: and no, my shoes are not made of leather.
20:47.58KattyQwell: i do not wear makeup, and my conditioners are not tested on animals, nor do they contain animal.
20:47.58Qwellcroc?
20:48.18QwellKatty: examples of free range animal biproducts?
20:48.20shido6the hold button, Alives, or offhook quickly
20:48.26KattyQwell: free range honey
20:48.36KattyQwell: it is illegal to buy free range milk
20:49.43QwellKatty: Do they actually market and sell free range honey?
20:49.59*** join/#asterisk marc324 (n=marc3234@206-248-159-253.dsl.teksavvy.com)
20:50.12nestArPIZZA!
20:50.17nestAri could never be vegan
20:50.21Qwellhmm...what about mushrooms?  Do you eat mushrooms?
20:50.24QwellnestAr: yeah, totally
20:50.31marc324is the te411p worth an upgrade from te410p?
20:50.35nestAri did the whole ovo lacto veggie thing for a while.
20:50.39nestArbut i <3 cheese
20:51.03QwellI know vegetarians who eat cheese and eggs...kinda weird
20:51.06scspongerhey does anyone what the headset command is to put someone on hold?
20:51.22nestArQwell: that's ovo lacto
20:51.30Qwelloh
20:51.35Alivesshido6: its not working
20:51.39QwellnestAr: and fish?
20:51.42Alivesdo i have to dial a * command?
20:51.46Qwellco-worker eats fish too
20:52.11nestAri ate fish occasionally
20:52.28nestArthis was when i was a teenager
20:52.44QwellnestAr: glad to hear you grew out of it :P
20:52.55nestArI LOVE LAMP
20:52.57nestArerr steak.
20:54.51KattyQwell: of course.
20:54.59QwellKatty: to which?
20:55.00KattyQwell: are mushrooms animals?
20:55.08Qwellno, but the way some of them are grown...
20:55.14Qwellnot sure that would constitute a biproduct
20:55.19KattyQwell: mew?
20:55.32Qwellyou know...the fun kind?
20:55.36Kattymrow?
20:55.39Qwellnevermind
20:55.42Kattyk
20:56.52QwellI bet fungi have souls though
20:58.00iDunnomouldy souls.
20:59.42Qwellalright, I'll shut up now
21:02.01*** part/#asterisk colinm_ (n=colol@VDSL-130-13-9-157.PHNX.QWEST.NET)
21:03.24*** join/#asterisk mcf3782 (n=mcf3782@adsl-19-83-37.asm.bellsouth.net)
21:05.33mcf3782Anyone in here had any luck getting kphone to work with Asterisk? I could use a little debugging help, as I've done everything I can think of.
21:06.53*** join/#asterisk abcbooze (i=abcbooze@70.153.216.92)
21:06.55abcboozesup
21:07.10abcboozeanyone have a cisco 12SP+ phone?
21:09.34mcf3782Asterisk seems to think that my kphone client is connecting and registering, as it prints messages to the console like: "-- Registered SIP 'kphone' at 200.200.200.123 port 5060 expires 900"
21:09.43gandhijeeumm so can my IAX local phones not call my SIP local phones?
21:11.24mcf3782When I try and call an extension on my Asterisk server from kphone, I see messages in the Asterisk console like: "Oct  7 16:14:49 WARNING[4222]: chan_sip.c:695 retrans_pkt: Maximum retries exceeded on call 1116616469@200.200.200.123 for seqno 3362 (Non-critical Response)"
21:12.27ManxPowerCan anyone help me debug a jitterbuffer issue?
21:12.46*** join/#asterisk Gibster (n=hnd@paradise.baysider.com)
21:13.07*** part/#asterisk mkrufky (n=mk@68.160.103.77)
21:14.27harryvvsixtel sound quality seems to have really improved since thay made some changes.
21:15.06harryvvDoes anyone know if netzero is selling voip services? I dont see it on there page yet my perents say thay pay 16 dollars a month for long distance though netzero.
21:15.14GibsterHi. I'm battling to get asterisk to listen on port 5060. I cant find anything in the logs and starting asterisk with -vvv shows starting sip and listening on interface fine. I have set listenaddress and port in sip.conf and I'm finally running out of ideas. Could someone please point me in the right direction.
21:15.45ManxPowerGibster, what makes you think Asterisk is NOT listening on port 5060?
21:16.32gandhijeehey can one not dial IAX to SIP and SIP to IAX?
21:16.33GibsterI get a timeout when trying to use a sip client and netstat doesnt show anything on port 5060
21:16.51ManxPowerGibster, sounds like a NAT problem.
21:17.12ian_kiptables -F
21:17.33*** join/#asterisk Rav1974 (n=r@static-70-19-119-112.ny325.east.verizon.net)
21:17.41ian_kgandhijee: no, unless you have asterisk between them
21:17.58Rav1974does amp support asterisk 1.2
21:18.04Primerian_k: heh, that's not A Good Thing™
21:18.05Gibsterhmmm. I'm not doing any NAT on this box. should I try setting the nat address anyway?
21:18.13ManxPowergandhijee, you can call between protocols just fine.
21:18.18Primerian_k: especially if you're remote, and your default policy is DROP
21:18.19ian_kPrimer: what isn't?
21:18.25Primer<ian_k> iptables -F
21:18.51ManxPowerGibster, in sip.conf [general] set port=5060 and do NOT have a bindaddr= line
21:18.54syle2gibster: it should be a udp port
21:18.55Primerdefault INPUT policy, that is
21:19.05PrimerManxPower: why not?
21:19.13ian_kPrimer: true.. most people that can't determine why a port isn't open won't have an overly complex firewall though. Most distros have some lame default that is easily disposed of.
21:19.15Primerwhy not have a bindaddr? I use it...works fine
21:19.18ManxPowerPrimer, older versions od Asterisk had problems with it.
21:19.28ManxPowerand really, most people don't need it.
21:19.33PrimerManxPower: ahhh
21:19.52ManxPowerwithout a bindaddr, asterisk should bind to all IPs.
21:20.05ian_kdoes it bind to yours?
21:20.09Primerian_k: all I'm saying is I've been bitten by that once when a dude with sudo on my box ran iptables -F when I was on vacation
21:20.16Primerand he was supposed to know what he was doing
21:20.36Primerasterisk  27133     root   15u  IPv4 4037522       UDP 216.239.132.121:5060
21:21.03Primerworks for me
21:21.03ian_kPrimer: I agree. I should have probably not put that in command-line-ready form.
21:21.28Primerian_k: this dude got said command from a friend on IRC ;)
21:22.38ian_kPrimer: Ah..
21:24.13*** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) [NETSPLIT VICTIM]
21:24.27gandhijee*CLI>     -- Registered SIP '2003' at 201.1.1.116 port 5060 expires 1800
21:24.27gandhijee<PROTECTED>
21:24.27gandhijee<PROTECTED>
21:24.27gandhijeeOct  7 21:07:03 WARNING[682]: channel.c:1891 ast_request: No translator path exists for channel type IAX2 (native 63502) to 512
21:24.27gandhijeeOct  7 21:07:03 NOTICE[682]: app_dial.c:764 dial_exec: Unable to create channel of type 'IAX2'
21:24.52Rav1974does amp support asterisk 1.2
21:25.43ManxPowergandhijee, you have a CODEC issue.
21:26.09gandhijeeok
21:26.10GibsterManxPower: thanks. seems to be binding now :)
21:26.12ian_kgandhijee - Does your iax phone work at all? ever?
21:26.15gandhijeeyeah
21:26.26gandhijeei had speex as the first on
21:26.29gandhijeei just changed it
21:26.32gandhijeegonna try it now
21:26.41ManxPowergandhijee, One side is trying to use SpeeX and one side is trying to use something else.
21:26.42ian_kgandhijee - use ulaw on both and see what happens
21:28.19gandhijeejust did
21:28.19gandhijeeworks
21:28.23gandhijeegotta go
21:28.24gandhijeethanx
21:32.58*** join/#asterisk Chotaire (i=chotaire@chotaire.net) [NETSPLIT VICTIM]
21:36.12*** join/#asterisk viLeR (i=1000@66.128.47.232)
21:36.44*** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) [NETSPLIT VICTIM]
21:41.06*** join/#asterisk Corydon-w (i=purple@pdpc/supporter/sustaining/Corydon76-home) [NETSPLIT VICTIM]
21:41.12*** join/#asterisk A-Tuin|work (n=A-Tuin@nat.office.legend.net.uk) [NETSPLIT VICTIM]
21:43.03*** join/#asterisk Hupe (n=Hupe@iD4CC1003.versanet.de)
21:43.39Kattyhmm
21:44.50*** join/#asterisk zeedo (n=zeedo@obsidis.org)
21:45.25Kattyfile[laptop]: are you in montreal?
21:46.05abcboozeanyone have a cisco 12SP+ phone?
21:46.12abcboozeand have config'd it for use with *
21:47.21HupeHi, is it a known Bug that Asterisk-System-Load  grows to 100% when sip register  fails for a longer time?
21:48.04*** part/#asterisk halogen8 (n=halogen8@66-146-190-146.skyriver.net)
21:48.36ManxPowerHupe, no.
21:49.37Nyvarabc, yuk no.  you look at the tiki page for 12SP+ phones?
21:49.51lanceyhi all
21:49.56Kattywikiwikiwikiwiki mushroom mushroom
21:51.05*** join/#asterisk flynux (i=hmj4hun@cl-8.bru-01.be.sixxs.net) [NETSPLIT VICTIM]
21:51.35uuuppzanyone here done much using BRI cards in NT mode?
21:51.44uuuppzI'd really appreciate some advice :)
21:53.16*** join/#asterisk SkramX (n=mark@mark-s.net)
21:53.22Kattyskram.
21:53.31SkramXHello, Katty ...?
21:53.43KattySkramX: that's what oscar says.
21:53.48KattySkramX: skram.
21:54.03SkramXOh. Sorry.
21:54.35SkramXWhatever.
21:54.55KattySkramX: are you having a bad day?
21:55.22SkramXKatty: No.
21:55.41KattySkramX: k
21:56.53SkramXI am alright.. you do know I am not Mark Spencer or anything, though I have been using the alias MarkS and Skram for a while now.
21:58.13abcboozehook me up skram ;)
21:59.00*** join/#asterisk Corydon-w (i=cinnamon@pdpc/supporter/sustaining/Corydon76-home)
21:59.43Kattybyebye
21:59.56KattySkramX: heh
22:02.38file[laptop]paulc: !!!
22:02.51*** join/#asterisk brc_ (n=Brian@pdpc/supporter/basic/brc)
22:03.08paulcI AM I AM.. kinda..
22:03.19anglerpaulc, whats up!
22:03.28paulcHey handsome :)
22:03.41paulcNot much.. Friday afternoon.. testing RMA devices.. packing and shipping others.. waiting for the weekend to begin! And you?
22:03.59jake1932paulc - how come when you leave and go you have this message come up
22:04.02anglerwaiting for 6pm myself
22:04.13jake1932leave and return
22:04.31jake1932now if you were elvis we'd have a tired joke
22:04.42pc2What's the command to automagically update asterisk from cvs head?  Isn't it like make update?
22:05.17paulcangler: can't come soon enough eh... weekend plans?
22:07.21file[laptop]feisty
22:08.54brc_FILE!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!11
22:08.56brc_WHAT
22:08.58brc_IS
22:08.59brc_UP!
22:09.05brc_mishehu, hey man
22:09.17mishehubrc_: hey
22:09.30mishehubeen what...  a few months already
22:09.33brc_ha
22:09.35brc_been busy
22:09.51brc_nat suckorz
22:09.57mishehusame here.  3 classes at college, a company to run, a parrot to train, and a world to conquer...
22:10.40*** join/#asterisk FuriousGeorge (n=brian@ool-44c5a9b8.dyn.optonline.net)
22:10.43FuriousGeorgehi all
22:11.24marc324is it possible to create a web interface that would update the extensions.conf file (without shutting down asterisk)
22:11.40brc_hahahaha
22:11.45brc_you kill me man
22:11.50mishehumarc324: if you build it, they will come.
22:11.57mishehuif you want them to build it, dial 1
22:12.30pc2make update says this.  hmm?
22:12.30pc2The following files have conflicts:
22:12.30pc2apps/Makefile
22:13.16FuriousGeorgevote:  usb vs. soundcard headset for softphone
22:13.21FuriousGeorge?
22:13.51Nyvarusb
22:14.03FuriousGeorgeNyvar: its unanimous
22:14.06Nyvarlol
22:14.58Nyvaralthough there is that cool hack of only plugging in a headset halfway :)
22:15.15marc324is answer yes or no?
22:16.00file[laptop]the answer will always be yes
22:16.13file[laptop]you could make Asterisk make toast for you if you wanted
22:16.59pc2FuriousGeorge - ata adapter :)
22:17.15mishehufile[laptop]: that's when it crashes and burns
22:17.19mishehuthen you get toast.
22:19.29FuriousGeorgefile[laptop]: toasts sounds good.  do you expect that to make it into 1.2?  i forgot if you were working on sip messaging or breakfast...
22:19.43FuriousGeorgeit was breakfast, right?
22:20.02file[laptop]app_breakfast
22:20.02file[laptop]duh
22:20.07lancey:))))
22:20.10FuriousGeorgeof course
22:20.14lanceyapp_antiheadache
22:20.18lanceyis what i need now :)
22:20.22Nyvarapp_chicken_n_waffles
22:20.33file[laptop]lancey: that's deprecated, it's app_advil now
22:20.35FuriousGeorgeNyvar: you beat me to it
22:22.51*** join/#asterisk nagl (n=nagl@213.235.241.6)
22:23.10lancey:>>>
22:23.33FuriousGeorgem@rch81
22:23.38FuriousGeorgethere goes my aim pw
22:23.57ender*snicker*
22:24.22pauldyanyone know an elegant way to give permissions to a user for the audio device when using devfs
22:24.24FuriousGeorgedamn sloppy focus
22:24.38pauldytrying to get console audio working with asterisk and this part has me a bit stumpet
22:24.45pauldystumped
22:25.02enderpauldy: I know a good way when using udev..
22:25.17pauldyhow than ender?
22:25.22shido6:(
22:25.29pauldyhows that ender?
22:25.46enderpauldy: there are permissions.d files for udev that you can define who gets ownership and what permission levels they have.
22:26.01pauldyhrm wonder if there is something similar for devfs
22:26.08enderpauldy: but devfs is deprecated and inherrently broken, so I don't know how to do it there.
22:27.34*** join/#asterisk buddah (n=djbrianc@67.110.253.129)
22:27.49buddahcan you upload .wav's to polycom phones to use as ring tones for distinctive ring?
22:28.18*** join/#asterisk pifiu (n=myassisb@208.205.181.170)
22:28.30pifiuhey what are some good sites for royalty free music?
22:28.42Nyvarbittorrent
22:28.48buddahyup
22:28.59pifiuwhich bit-torrent
22:29.07Nyvarj/k
22:31.36pauldyender, if I upgrade to udev where would I find the permission files?
22:32.39enderpauldy: /etc/udev/rules.d/
22:32.56enderI think the main one is 50-udev.rules
22:33.48pauldykewl thanks
22:34.10endernp
22:34.17pifiuso no one knows?
22:35.45FuriousGeorgepifiu: aussievoip-org
22:35.53mcf3782pifiu: I just did a google search for "royalty free music" and found quite a few sites. Have you tried that?
22:36.04FuriousGeorgewhats google?
22:36.19FuriousGeorgei come here whenever i need to know anything
22:36.23Nyvarthe key is finding 'free' royalty free music
22:36.34marc324can the extensions.conf be from a database?
22:36.42marc324built from db.
22:37.20marc324for large extensions, it becomes impossible to edit the file
22:37.38endermarc324: it can be done from include files as well
22:37.57enderuse many small files to create your big dial plan
22:37.59lanceyand from db as well
22:38.05enderyep
22:38.16lanceyres_config or something like that, it was, i recall
22:44.10pifiuyeah i have tried google
22:44.14pifiubut i was wondering if there were any free ones?
22:45.23Nyvarmarc, why not use 'realtime' asterisk and just everything from a DB?
22:45.24pauldypifiu, googleis your best bet they are out there but they come and go so there is no defacto site
22:47.00paulcJerJer: You around? I've got a quick question on nufone/remote party ID for presentation CLI
22:47.32mcf3782I just downloaded several mp3 music files from http://www.incompetech.com/m/c/royalty-free/
22:47.58mcf3782Their FAQ seems to indicate that all their stuff is royalty free and licensed under CCL
22:48.20Nyvarohhh.. space opera... my fav
22:48.32syle2PRI's are free inbound and outbound dialing right
22:48.48Nyvarall free, all the time
22:48.54Qwellsyle2: not always
22:49.02Nyvarassuming that someone else is paying the LD bill
22:49.09Qwellmy work pays like 2c/min for both local and long distance
22:49.11syle2assuming no long distance
22:49.19Qwellstill no
22:49.45Nyvardepends on the carrier, in the US its easy to find flat-rate-per-month PRI's, around the world some providers still charge per-minute fees
22:49.59syle2on inbound or outbound?
22:50.06Nyvarhow's that workin out for you aussies?
22:50.28syle2i;m not from US so just wondering
22:51.25opus__does anyone here use T38-bits?
22:52.00Nyvaris that a device?
22:52.21mcf3782Well, I have to say that thus far; I am very unimpressed by any of the softphones I've tried.  They either don't work at all, or I can only get audio to go in one direction.
22:52.35Nyvari use t.38, but not familiar with 't38-bits'
22:53.10Nyvarmcf, sounds like nat to me
22:53.29marc324how do you make an extension automatically execute once the context is called?
22:53.55lancey`busychan_sip rulz
22:53.56lancey`busy<PROTECTED>
22:53.58lancey`busy:)))
22:54.01syle2mcf3782: lots of sites like that, i just want a site where i can download gigs of mp3 files in some compressed format
22:54.19mcf3782If I'm talking across the same class-c lan from one machine to another; I wouldn't think nat would be involved.  But maybe I don't yet know enough to know what it is that I don't yet know.
22:54.28syle2i got about 15 thousand songs , always willing to trade
22:59.19mcf3782syle2: don't know what to tell you there. I was replying to pifiu's question with what I found.  I'm not yet at the point of needing music for MoH. I'm still trying to get some more basic stuff working like 2-way audio from a client (windows or Linux) and a softphone to my Asterisk server.
23:00.26mcf3782Best I've been able to do thus far is get the 4.1.0 version of kphone to register, call, connect, and receive audio from the server; but it can't/won't send anything back to it.
23:01.18endermcf3782: sounds like you don't have your dtmf settings right.
23:01.36enderwait, n/m
23:01.40mcf3782dtmf?
23:01.46mcf3782:)
23:01.48enderdtmf is for sending digits down the line.
23:02.05mcf3782yea.. I was wondering how that was related to sending audio. :)
23:02.07enderlike passcode for voicemail or something.  My co-worker had dtmf issues w/ kphone.  You're talking about audio though.
23:02.26enderHe had to use a specifc audio codec too....
23:02.32endersure your mic is working and all that rot?
23:04.12mcf3782I can hear the audio from my mic (tried the built-in one and one plugged into the mic jack) in the speakers on the client. I can start up a sound recorder on the client and record my voice from both the built-in and the external mic; I just don't seem to be having any luck getting kphone to send it to the server.
23:04.48mcf3782Watching the lan with tcpdump, there's no indication that there's any audio traffic leaving the client, headed toward the server... or anywhere else for that matter.
23:05.10enderah
23:06.41mcf3782I've tried several versions of kphone, starting with the most recent version posted on their web site, and walking down through older versions. the 4.1.0 version is the first one that will even connect to the point that it receives audio being sent from the asterisk server to the client.
23:08.00mcf3782Maybe I just haven't had enough caffeine yet for it to make sense..
23:08.17Nyvarmaybe kphone has a mic volume contorl
23:08.28opus__mcf3782 check externip, make sure you define localnet. check nat=yes
23:08.52*** join/#asterisk nvrs (i=RUR@Toronto-HSE-ppp3866734.sympatico.ca)
23:09.05mcf3782opus: are those kphone settings or asterisk settings?
23:09.22opus__asterisk settings
23:10.17Nyvarhttp://www.linuxjournal.com/node/8165/print  -- search for Turn on the microphone
23:10.21*** join/#asterisk ZeMMad (n=blah@72.252.15.246)
23:10.23mcf3782Ahh.. ok. I will look at both of those.
23:10.38*** join/#asterisk zotz (n=zotz@24.231.36.100)
23:11.19mcf3782Nyvar - good thought. The mic is definitely turned on though according to alsamixer.
23:12.10mcf3782I'll check the externip and localnet settings that opus suggested, but first...more caffeine. ;)
23:16.11opus__finally I have some dvds to burn all my shit off to!
23:27.02*** join/#asterisk lters (n=lters@mrtcdsl-034.mis.net)
23:29.18*** join/#asterisk asterisk99 (n=dunoowhy@modemcable169.194-130-66.mc.videotron.ca)
23:29.44asterisk99anyone here using PRIs? I have a really dumb question!!!!! ;)
23:30.11denonjust ask it
23:33.01asterisk99denon: I configed a context called pri in my zapata.conf... In extensions.conf I created a siple Dial(zapata/1/4165551234) to try an outbound call... I get 'no channel type registered for pri'     ideas?
23:33.17asterisk99denon: I configed a context called pri in my zapata.conf... In extensions.conf I created a simple Dial(zapata/1/4165551234) to try an outbound call... I get 'no channel type registered for pri'     ideas?
23:34.00ltersZap/1/12345
23:34.01asterisk99denon: I don;t see a type= entry in zapata.conf
23:34.09ManxPowerasterisk99, contexts are normally only use for INCOMING calls.
23:34.19ManxPowerAnd it would be Dial(Zap/1/12345)
23:34.31asterisk99ManxPower: Zap???  OK I'll try
23:34.55ManxPowercontext=pri in zapata.conf would cause incoming calls from the PSTN into your PRI to land in the [pri] section of extensions.conf
23:35.52asterisk99ManxPower: OK tried Dial(Zap/1/4165551234) I get no channel type registered for 'Zap'
23:36.35asterisk99ManxPower: It's a PRI
23:36.45ManxPowerasterisk99, then 1) you don't have the zap drivers installed, 2) you have noload => chan_zap.so in /etc/asterisk/zapata.conf, or 3) you didn't rebuild Asterisk after installing zapata
23:37.04ManxPowerAll digium cards are Zap, regardless of the interface
23:38.58*** join/#asterisk stkn (i=nobody@gentoo/developer/pdpc.active.stkn)
23:39.29asterisk99ManxPower: Oh oh... ok    Zaptel is loaded (lsmod sees it)... no Noload in zapata.conf... I dodn;t rebuild Asterisk  (that's definately required??)
23:39.40asterisk99ManxPower: Oh oh... ok    Zaptel is loaded (lsmod sees it)... no Noload in zapata.conf... I didn't rebuild Asterisk  (that's definately required??)
23:39.59ManxPowerasterisk99, If asterisk doesn't see zaptel installed when you build Asterisk, it won't build zap support
23:40.13ManxPowerasterisk99, and if you keep repeating every line you type nobody will help you
23:40.50asterisk99ManxPower: Hmmmmmm.  (Sorry. I repeated the lines to correct typos. I'll let the typos alnoe from now on.)
23:41.10mcf3782the 'gotta rebuild asterisk after installing zap' part got me the first time around.
23:41.17ManxPowerI hate IPSec
23:42.05asterisk99ManxPower: Do I need set idledial, idleext, minunused for PRIs or is that optional?
23:42.21ManxPowerasterisk99, I've never used them
23:44.08opus__damn it
23:44.16opus__Anyone here use a T.38 provider?
23:44.55*** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net)
23:45.05generalhanwhats up everyone ?>
23:45.36generalhanwho in here is pretty good with the FOP ? rather is anyone in here using the FOP ?
23:47.50*** join/#asterisk _zemmad (n=blah@72.252.15.246)
23:48.09_zemmadhi is there an asterisk software for windows??
23:48.57*** join/#asterisk hypnox (n=dan@cornelyn.force9.co.uk)
23:49.13mmlj4_zemmad: you might be able to compile it under cygwin
23:49.33*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
23:49.36mmlj4but no, there is not and will not ever be an official asterisk release for windows
23:50.16_zemmado
23:50.24_zemmadgood stuff
23:53.42mmlj4well, hrm, i was sorta wrong: http://www.voip-info.org/tiki-index.php?page=AsteriskWin32
23:53.51ManxPowermmlj4, it's not supported
23:54.30ManxPowermmlj4, and it's really a linux emulator running under windows, which Asterisk runs inside of, and it only supports VoIP, and people will point and laugh at you if you try to use it.
23:54.57mmlj4notice i said "sorta"?
23:55.07ManxPoweropus__, Since Asterisk does not support T.38, not many people use it in this channe
23:55.41ManxPowermmlj4, my "isp" replaces one of their boxes and my internet connection seems more stable.
23:56.08mmlj4hmm... they couldn't take anymore complaining from you? heh
23:56.25ManxPowerOf course, jitter is still pretty horrible
23:56.28hypnoxhow popular is call routing by srv record?
23:57.12FuriousGeorgewhat do i need besides this http://pastebin.ca/24897 to detect fax?
23:57.21ManxPowerI hate IPSec
23:57.23FuriousGeorgei cant find anything on the wiki besides that, but that doesnt seem to do it
23:57.42ManxPowerI'm starting to think that IPSec doesn't actually encrypt anything, it's just so complicated nobody can figure out how to read the data.
23:57.54hardwireboob
23:58.01*** join/#asterisk toddf (n=toddf@adsl-67-65-251-106.dsl.okcyok.swbell.net)
23:58.16mmlj4ok, so why do you hate it?
23:58.29ManxPowermmlj4, because it's so complicated
23:59.34mmlj4heh, well, have you looked at openswan? it's tons easier than freeswan was, i'm told
23:59.44ManxPowerHuh?  I use Cisco
23:59.54mmlj4mmm, ok

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