IRC log for #asterisk on 20091110

00:01.04*** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl)
00:04.29superbeefI'm playing with make menuconfig...... module embedding is just adding functions directly into the build rather than loadable modules right?
00:06.07Get_The_Fishthats my understanding
00:06.48superbeefword
00:07.14superbeefi'm recompiling now that i have libcap-dev installed so that tos will work... i was hoping i could see some sort of indicator that the support would be compiled in, but i havnet realy found it
00:07.14carrary0
00:08.10carrartakes LDAP and throws it on the ground!
00:08.22Get_The_Fishdaddy likes ldap
00:08.40manxpowersuperbeef: config.log should indicate it was found.
00:09.12superbeefmanxpower: what should i search for in the log.. libcap?
00:09.25manxpowersuperbeef: yup
00:09.35*** join/#asterisk mythicalbox (n=mythical@rrcs-64-183-110-250.west.biz.rr.com)
00:09.37superbeefmanxpower: no dice
00:10.00manxpowersuperbeef: what version of Asterisk?
00:10.06superbeef1.4.26.2
00:10.34*** join/#asterisk rpm (n=russell@S0106000c29898b7e.cg.shawcable.net)
00:10.39superbeefmanxpower: maybe it would help if i install libcap-bin
00:10.45Get_The_Fishso, in config/scripts, does anyone know the difference between asterisk.ldif and asterisk.ldap-schema, with regard to usage on fds?
00:10.55manxpowersuperbeef: I don't have a 1.4 machine handy, let me boot up my 1.6 machine a min
00:11.56manxpowersuperbeef: usually you have to run a make command I can never find or just delete the source dir and unpack it again when you install libraries
00:12.18superbeefmanxpower: something beyond make clean and ./configure ?
00:12.27manxpowersuperbeef: correct
00:13.46Godfather_manxpower, im reading examples of parkedcalls and transfer calls. In these examples speaks about "Transfer button", wich button is it on a analog phone connected to a ata?
00:14.03carrar#
00:14.23carrarand ading a variable to the dial command
00:14.23manxpowersuperbeef: "checking for cap_from_text in -lcap" is what I see in my config.log in 1.6
00:14.44superbeefmanxpower: awesome thanks i'll sure for it after its done recompling
00:14.53Godfather_ok
00:14.55superbeefsure=search
00:15.14manxpowerActually, just below that line is what you should be looking for.
00:15.41*** join/#asterisk mchou (n=mchou@unaffiliated/mchou)
00:15.46superbeefbreaking the 3 line rule
00:15.47superbeefconfigure:17641: checking for cap_from_text in -lcap
00:15.47superbeefconfigure:17676: gcc -o conftest -g -O2   conftest.c -lcap    >&5
00:15.47superbeefconfigure:17683: $? = 0
00:15.47superbeefconfigure:17704: result: yes
00:15.55superbeeflooks like i'm gravy
00:15.57superbeefthanks
00:16.46Godfather_carrar, this example is wrong then?
00:16.48Godfather_http://www.asteriskguru.com/tutorials/transfer_image272184.jpg
00:17.10Godfather_i dont understand SIP/1212
00:17.26Godfather_i think it must be Dial(SIP/115)
00:18.13[TK]D-FenderGodfather_: Ignore that.... odds are they have no clue what they are doing.
00:19.02*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
00:19.16[TK]D-FenderGodfather_: And no, you do a transfer with teh PHONE to the parking extension defined in features.conf
00:19.57Godfather_[TK]D-Fender, i've read this with Callparking
00:20.06Godfather_but it doesnt work for me
00:20.57Godfather_[TK]D-Fender, did you see the example i posted?
00:21.43[TK]D-FenderGodfather_: You have to include [parkedcalls] and transfer the call to the parking ext # you defined
00:26.16Godfather_[TK]D-Fender, with Transfer() you can call somebody and after pressing # destination call to the desired destination nop?
00:27.49Godfather_assuming you have exten => user1,1,Dial(SIP/101,tT)
00:28.38leifmadsenjblack: where do I send the $1000 to in order to prove my worth and have you kiss my pinky ring and call me Lord?
00:30.49carraronly $1000?
00:31.27leifmadsenthat was the price to prove you make a million dollars
00:32.16leifmadsen(please note that I do not actually make a million dollars a year, and will not actually be sending any money to jblack)
00:32.24leifmadsen(but don't tell him that)
00:34.31Godfather_[TK]D-Fender, http://pastebin.com/m40a39927
00:35.15Godfather_this is a good configuration of transfer?
00:36.14[TK]D-FenderGodfather_: ... FORGET THE TRANSFER APP
00:36.24Godfather_O_O
00:36.30[TK]D-FenderGodfather_: Am I clear now?  If a guide mentioned it to you then it is WRONG
00:36.43Godfather_lol
00:36.59[TK]D-FenderGodfather_: What are you calling with?
00:37.03Godfather_[TK]D-Fender, then, how do you pass a call to another sip phone?
00:37.24Godfather_imagine A calls 101, then 101 want to tranfer the call to 102
00:37.27[TK]D-FenderGodfather_: What are you calling with?
00:37.41Godfather_with another softphone on the network
00:37.46Godfather_103..
00:37.52mythicalboxdoes anyone know why putting a caller on hold and then coming back to the call sound would be gone from the caller but they can still hear us?
00:38.00[TK]D-FenderGodfather_: What are you calling with?
00:38.29Godfather_[TK]D-Fender, with a phone?
00:39.16*** join/#asterisk TecR0c (i=tecr0c@never.met.a.leet-hacker.net)
00:39.28TecR0cwhat is a good free VOIP client for the mac ?
00:39.50Godfather_[TK]D-Fender ?
00:40.01Godfather_http://pastebin.com/m3a68583a
00:40.44[TK]D-FenderGodfather_: What are you calling with?
00:41.02Godfather_i dont know what you are asking
00:42.41[TK]D-Fenderheads off for a while
00:42.45Godfather_:/
00:43.17TecR0ci want a good VOIP client that works well with MAC 10.6 and trixbox
00:43.28TecR0cx-lite doesn't work
00:44.03rpmCan realtime support retrieving an accountcode from a database with IAX, I attempted this earlier but I'm taking another shot at it. It seems if I statically set an entry in /etc/asterisk/iax.conf it works, when I use the realtime iax_friends table it fails. Where should I begin?
00:44.21*** join/#asterisk smokie (n=efnet@27.9.202.62.fix.bluewin.ch)
00:45.39smokiehey guys, i installed asterisk on ubuntu and used x-lite on 2 windows machines to test 2 users i added, they connect fine but cannot make any calls
00:45.55*** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri)
00:45.55smokieany idea why its not making any calls?
00:46.03smokiethis is for a local network only
00:47.23*** join/#asterisk timmyd (n=timmyd@pool-173-79-13-149.washdc.fios.verizon.net)
00:48.03timmydhey i noticed something in my asterisk prompt, someone else is registering to my extensions: e.g. Registered SIP '105' at 72.54.32.26 port 19169 expires 120
00:48.45timmydhow can i figure out more if it's a bug in asterisk or if my box is compromised or something?
00:49.52manxpowertimmyd: change the password for that account
00:49.54carrarhaha
00:50.29timmydi did that a few weeks ago to all random pws since they were easily guessable but it just happened again today
00:52.28*** join/#asterisk retentiveboy (n=pdugas@74-95-28-37-Atlanta.hfc.comcastbusiness.net)
00:52.59*** join/#asterisk voipmonk (n=voipmonk@dsl-67-55-17-41.acanac.net)
00:53.36*** join/#asterisk chendy (n=chatzill@119.122.163.43)
00:53.39timmydand the weird thing is that the bot(?) registered to like 5 accounts
00:56.03*** join/#asterisk tuxcrafter (n=jelle@84-245-3-195.dsl.cambrium.nl)
00:56.23TecR0csmokie: i have the same issue with my mac
00:58.38rpmnasty. accountcode is only processed by iax.conf when it is defined in the [general] section, if you define it under a subscriber, type=friend/peer or user it is always null.
01:00.09smokieTecR0c, so i assume you didnt find a solution yet?
01:02.25TecR0cnope i also tried the beta
01:02.30TecR0cno help
01:02.37TecR0cim looking for another application to do the job
01:02.41TecR0cbut no luck yet
01:03.49russellbhave you tried zoiper?
01:04.23russellbAlso, you should try AsteriskNOW instead of trixbox.
01:05.03carrardoesn't russellb use freepbx?
01:05.08carrarruns
01:05.11carrarruns faster
01:05.41russellbcommits a patch into Asterisk ... if (admin == carrar) { eat_brainz(); }
01:05.46carrarhaha
01:05.52carrarnoooooooooooooo!!
01:09.01smokieQwell, its the first time i install it asterisk and i have it installed in ubuntu 9.04
01:09.03*** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
01:09.03smokiewell*
01:10.02*** part/#asterisk timmyd (n=timmyd@pool-173-79-13-149.washdc.fios.verizon.net)
01:15.11*** join/#asterisk SaiSoma (n=SaiSoma@adsl-074-167-136-030.sip.mob.bellsouth.net)
01:16.53*** part/#asterisk ruben23 (n=RPL@122.55.48.243)
01:20.31*** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim)
01:26.11*** join/#asterisk Kumbang (n=kumbang@rusnas.paume.itb.ac.id)
01:28.39*** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber)
01:35.30*** join/#asterisk dan__t (n=dant@vpn.withparity.net)
01:35.39dan__t'evening.
01:36.15dan__tb14ck, you around by chance?  Don't mean to call you out by name like this but I had some questions regarding a "shadow" channel, like attached to another channel, in a bridge.
01:36.26dan__tOr better yet, placing one channel on more than one bridge at the same time.
01:48.47*** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
02:00.01*** join/#asterisk riddlebox (n=user@75-132-225-75.dhcp.stls.mo.charter.com)
02:00.36*** join/#asterisk geneticx (n=geneticx@adsl-10-104-182.mia.bellsouth.net)
02:02.19geneticxhi !
02:06.04b14ckdan__t, ?
02:07.50*** join/#asterisk Trupsalms (n=Shawndel@adsl-68-20-35-251.dsl.chcgil.ameritech.net)
02:07.59Trupsalmshelp
02:08.01Trupsalmsplease
02:08.15Trupsalmstrying to install asterisk
02:08.33b14ckhelp with what?
02:08.34Trupsalmsinstalling all pre tools before the install
02:08.35b14ckyou didnt ask a question?
02:08.39Trupsalmsok
02:08.40Trupsalmssorry
02:08.53Trupsalmsdahdi tool install
02:09.05b14ckare you installing on centos?
02:09.05Trupsalmsi receive error
02:09.57Trupsalmscompilation failed in require at perl_modules
02:10.04Trupsalmsubuntu server
02:10.16b14ckso are you compiling from source/
02:10.20Trupsalmsyes
02:10.23*** join/#asterisk chendy (n=chatzill@119.122.163.43)
02:10.26b14ckinstall dahdi-linux first
02:10.35b14ck./configure && make && make install
02:10.44b14ckthen dahdi-tools: ./configure && make && make install && make config
02:10.56Trupsalmsis that apt-get install dahdi-linux
02:10.58b14ckIf the dependencies are not met--install them.
02:11.15Trupsalms?
02:11.18b14ckNo... You just said you were compiling from source... apt-get is a package manager
02:11.26b14ckpackage are not sourcecode, they are binaries.
02:11.27*** join/#asterisk mchou (n=quassel@unaffiliated/mchou)
02:11.53Trupsalmswhat i'm i running from command prompt
02:12.52b14ckdies
02:31.24*** join/#asterisk bbt (n=sam@180.189.138.55)
02:36.36hardwiredies too
02:37.27*** join/#asterisk jaytee (n=jforde05@unaffiliated/jaytee)
02:40.52*** join/#asterisk OrNix (n=ornix@host89-251-107-21.hnet.ru)
02:41.14dan__thi
02:41.26dan__tMan my laptop battery died right after I asked that question.  Anyone mind letting me know if someone replied please?
02:50.32hardwirehi is not a question.
02:50.40hardwire:P
02:51.41p3nguinThe old 'gaim' FAQ said   Q: Hello?   A: We don't know how to answer that; stop asking.
02:52.12carrarhi?
02:52.19p3nguinno.
02:52.27p3nguinor yes, depending on if you want to.
02:52.31carrarhika
02:52.54*** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber)
02:53.53dan__tI didn't want to repeat :)
02:54.11dan__tCan I have one channel on two different bridges at the same time?
02:55.07carrarCan I have fries with that
02:55.13dan__tYou may.
02:57.17drmessanoBridges generally go over channels
02:57.24drmessanoYes, one bridge may span two channels
02:58.01drmessanoThe channels would need to be close together
02:58.23dan__theh
02:58.48drmessanoTrick is, too close and they're one channel, too far apart and technically its two bridges
02:58.53drmessanoFine line there
02:59.03dan__tright.. i'll set something up and test it.
02:59.14drmessanoIts not that easy
02:59.31drmessanoFirst we need to start off with the environmental impact assessments
02:59.37drmessanoThose take months, if not years
02:59.51Kattypeeks in
02:59.52drmessanoThen we must solicit designs
02:59.55Kattyewwo :>
02:59.59Kattyi've come to crash the party.
03:00.06drmessanoThen we solicit bids on said projects
03:00.27*** join/#asterisk chendy (n=chatzill@119.139.171.193)
03:00.30dan__tbureaucrats and all.
03:01.23drmessanoFollowed by lots of underhanded dealing, where in the process we give the bid to some political allies' cousin, who will subsequently spec out the project with 1/3 less bolts than the original design, and 12x more likely to result in a catastrophic failure
03:01.52drmessanoBut.. Not until you've moved on to Washington and long forgotten those local peasants and they're stupid bridge
03:01.52Kattyi showed up at a bad time, didn' ti
03:01.55drmessanotheir*
03:02.01dan__tYou did.
03:02.47jblackdrmessano: You didn't respond to my tweet, you twit
03:03.01drmessanoSorry, my canary is sick
03:03.08jblacktwit, as defined as something that tweets. :)
03:03.12drmessanoand by that, I mean, tweetdeck was turned down
03:03.39jblackturned down huh?
03:04.01jblackOh, silenced or quiet or whatever
03:04.14drmessanoyeah
03:04.36Kattydoesn't want to know.
03:04.39drmessanoHavent been tweeting from my phone, seeing as how I am a short timer and it's a work phone
03:04.41jblackThat's fine. I just had to jump at the one barely-legitimate reason I've ever had to call someone a twit without being insulting. ;)
03:04.43drmessano9 more days!
03:05.03Kattymister black, are you on facebook?
03:05.03jblackthough sometimes I think it's a complement to some
03:05.05Kattyi can't recall
03:05.20jblackMs atty, I technically am.
03:05.32jblackI have an account. I don't remember the acount name or password.
03:05.36*** join/#asterisk cyyaw (n=cyyaw@201.240.31.76)
03:05.40jblackso if you find me, point me out to me?
03:05.50Kattysure. uhh... /query me your state again.
03:05.56jblackwhere I live?
03:06.01jblackWilkes-Barre, PA
03:06.02Kattyjust the state
03:06.11jblackPennsylvania
03:06.33jblackI think I have facebook. Maybe it's myspace.
03:06.39jblackmight be both.
03:06.46Kattyi'll ask pipl
03:06.49jblackI'll make a fb
03:06.53drmessanoDid I mention 9 more days
03:07.10Kattyjblack: if you have an account, pipl will find you
03:07.27Kattyinteresting, you have an amazon profile
03:07.31drmessanoI had a pipl once, but I popdit
03:07.47Kattyah,, that's a good novel you have on your wish list.
03:07.52Kattyi liked brave new world (=
03:08.22drmessanoMy amazon wish list is full of romance novels and grandstream ATAs
03:08.28drmessanoI suck :(
03:08.34Kattyromance novels are boring.
03:08.38carrarHOT
03:08.53Kattyboring.
03:08.55drmessanoRomance novels are NOT boring
03:08.57carrarA romance over a grandstream ATA phone call
03:09.24Kattyi found you on friendster.
03:09.39jblackOk. I have facebook now
03:09.56Kattyk
03:10.13drmessano"He slowly lowered his overbaked creampuff and brushed her gelatinous bowl of love pudding...."  <-- LOVE THAT STUFF
03:10.28Kattyare you part of the Pittsburgh area?
03:10.30carrarfood channel romance?
03:10.42Kattyi'll just link you my page.
03:10.47carrarthats dangerous to watch late at night
03:11.05Kattydrmessano: that sounsd like Xanth
03:11.08Kattydrmessano: also, boring.
03:11.13Kattydrmessano: romance novels are so vanilla.
03:11.51drmessanocarrar: Sandra Lee is HAWT
03:12.18dan__tSorry, this is worth sharing:  http://twitpic.com/oye5r
03:12.19drmessanohttp://www.foodnetwork.com/chefs/sandra-lee/index.html  <--- Nobody doesn't like Sandra Lee
03:12.48carraryeah
03:13.06carrarI think she was adultism
03:13.08carrarwas on
03:13.14carrarsomeone told me that
03:13.18carrarheh
03:13.35*** join/#asterisk maxagaz (n=maxagaz@soho2.i-xanadu.com)
03:13.55drmessanoShe was?
03:14.10Kattythere sure are a lot of mister blacks on facebook
03:14.28jblackWoot! I have a facebook friend!
03:14.50jblackbreaks into sobbing when he realizes he has a friend.
03:14.57Kattypats jblack
03:14.58jblackOk. I'm over it. I'm better now. =)
03:15.01Kattyk
03:15.06Kattynow it's time for PHOTO!
03:15.16jblackoh god. I'm frigging ugly
03:15.23Kattywho cares! it's you
03:15.38Kattyonward with the photos!
03:15.40jblacksmall children care!
03:15.44jblackgallery.linuxguru.net
03:15.51Kattypoints jblack back to facebook
03:15.53jblackI'll find something
03:16.06jblackhttp://gallery.linuxguru.net/main.php?g2_itemId=3871
03:16.09jblackI love this pic. :)
03:16.23p3nguinI have no friends, I guess, since I don't do the facebook nor myspace stuff.
03:16.38jblackp3nguin: Do some facebook. We can all be friends.
03:16.39Kattyi understand why she's an ex.
03:16.44p3nguinIs that vrsa?
03:16.44Kattyno offense meant ;)
03:17.14jblackThen, we can get a VW bus, and paint flowers on it, and make a commune where we a love one another, toast marshmellows, and play beatles songs.
03:17.31Kattyi like 0294
03:17.41Kattyoh wait, no, 0500 is better
03:17.53jblackI was thining 4017
03:17.59jblacklemme look at 0500
03:18.51rpmdoes anyone use iax and realtime here?
03:19.07Kattyis mister hoodie on the left a relative of yours?
03:21.06*** join/#asterisk jblack (n=jblack@pool-96-243-97-134.sctnpa.east.verizon.net)
03:21.13jblackam I back?
03:21.18Kattyyes
03:21.31jblackyeah, 500 is about as good as it gets.
03:21.37Kattyk (=
03:21.48rpmis it me, or is iax accountcodes broken in 1.6.x?
03:23.31jblackActually, I'm not even gona crop it
03:23.57jblackI don't have that turkey thing under my neck any more.
03:24.06*** join/#asterisk saxa (n=sasa@host242-95-static.223-217-b.business.telecomitalia.it) [NETSPLIT VICTIM]
03:24.12jblackwell, not so bad, anyways
03:24.18jblackI should gimp that out
03:24.32Kattyleave it
03:25.26jblackwhile I'm at it...
03:27.05Kattyi sent you some more friends.
03:27.26jblackI killed twitter.
03:27.36Kattynow if only p3nguin had a facebook :<
03:27.55jblackTried uploading my pic, they came back with a picture of a big fat whale holding down a swarm of birds.
03:28.26*** join/#asterisk Gokee2 (n=gokee2@24-113-159-168.wavecable.com)
03:28.27*** join/#asterisk mahlon (i=mahlon@martini.nu) [NETSPLIT VICTIM]
03:28.27*** join/#asterisk _Raptor_ (i=raptorbl@andariel.informatik.uni-erlangen.de) [NETSPLIT VICTIM]
03:28.59jblackIs that the Suggestions on the right?
03:29.07Kattyprobably
03:29.12jblackNo.
03:32.38jayteeI just accepted Katty's friend suggestion for you
03:34.41Katty:>
03:34.44KattyTWO FRIENDS! HORAY!
03:37.55jblackI haven't found the suggetsions yet
03:38.11jblackI see lots of suggetsions at the bottom.
03:38.24Kattyi think the friend suggests actually send YOU to the other people
03:38.26jblackBut I think those are automatic.
03:38.29p3nguinkatty: If I did have one... ?
03:38.48Kattyp3nguin: facebook
03:39.03jblackJaytee, how do I find you?
03:39.08p3nguinRight.  I'm saying, if I did have one, what would happen?
03:39.31jayteejblack, if you're James Blackwell then I already sent you a friend request
03:39.41jblackBecause when I search for "Jaytee" I'm seeing anyone I Think is you. :)
03:39.43jayteeit should show up in the upper right corner of your Wall page
03:39.51jayteeJohn Forde
03:39.57jblackOh, there it is, in front of my face.
03:40.05Kattyhe looks like a penguin
03:40.11jblackThe green penlantern?
03:40.18jayteethat's the one
03:40.19jblackYAY!
03:40.35Kattyp3nguin: that much is obvious.
03:40.44jblackpeng, where's your page?
03:40.50Kattyhe doesn't have one
03:40.58jblackp3nguin: Go get your page!
03:40.59Kattyeven tho he's just afew hours away from me
03:41.08Kattyp3nguin: we have another friend, nearby, in Marion
03:41.13Kattyp3nguin: well, close to marion
03:41.20jaytee"In brightest day, in blackest night, no evil shall escape my sight, let those who worship evil's might beware my power, Green Lantern's light."
03:41.39Kattyp3nguin: Evansville
03:44.30jblackBut green lantern wasn't a penguin.
03:44.46jblackI never got deep into superfriends, but I'm prtty sure of that.
03:44.51jblack[TK]D-Fender: Where's your fb?
03:44.53jayteeit's a linux avatar penguin, there's a website with thousands of them
03:45.09jayteethere's even a Judge Dredd penguin
03:45.27jblackHow about wondertwin penguins? I can't live without a penguin in the shape of a puddle of water!
03:45.47jayteenot sure about wondertwins
03:46.07*** join/#asterisk Corydon76-dig (i=black@c-68-52-33-133.hsd1.tn.comcast.net)
03:46.07*** mode/#asterisk [+o Corydon76-dig] by ChanServ
03:46.14carrarcan I get a waterfall!
03:47.20jblack[TK]D-Fender: ping
03:47.57*** join/#asterisk bmoraca (n=chatzill@66.242.174.254)
03:47.57*** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr)
03:47.57*** join/#asterisk Chinorro (n=Chino@91.117.226.19)
03:50.40rpmis there a list of testcases performed on asterisk before it is released?
03:51.55jblackis having a ball
03:53.20jblackrpm: getting the sources should indicate that easily enough. I'll check real quick
03:53.40Kattyhmm.
03:54.13rpmaccountcode= does not work on iax peer/user or friend statements, accountcode= only works when it is set within the [general] section.
03:54.24Kattythere's this odd feeling in my stomach
03:54.39jblackrpm: are you sure?
03:54.42jblackdouble checks
03:55.06Kattyi think i'm hungry
03:55.17jblacktry setvar=accountcode=whatever
03:55.29jblackkatty: Hrmmm. There might be a solution for that.
03:55.32Kattyyes.
03:55.34Kattywalnuts!
03:55.45rpmjblack: i've been trying to get it working all day today. when i set the accountcode= under the general section, i see the correct accountcode. when i set it under a subscriber account it is always null.
03:55.47*** join/#asterisk hatoff (n=hatline@unaffiliated/hatoff)
03:55.47jblackOhh, better than the mcdoubles I ate. :)
03:55.48jayteeI have smokehouse almonds
03:55.57rpmrotten ronny's?
03:56.02jblackrpm: try setvar=accountcode=whatever ?
03:56.11hatoffwhich is the simplest way to check if asterisk sees the modem?
03:56.15jblackI like buying raw almonds and roasting them myself.
03:56.18rpmjblack: in the iax.conf?
03:56.26jblackhttp://www.mail-archive.com/asterisk-users@lists.digium.com/msg227066.html
03:57.47*** join/#asterisk dkirker (n=dkirker@pcp063416pcs.wireless.calpoly.edu)
03:57.50Kattywas it anumber 2?
03:58.01Kattya number two sounds awfully good
03:58.17jblackgives katty a weird look
03:58.31*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
03:58.35Kattymaybe it's a number 7
03:58.46Kattyit's the two cheeseburger thingy
03:58.52jblackOh.
03:59.05jblackNah. It's a double hamburger with a slice of cheese.
03:59.11hatoffomg asterisk doesn't work with modems!?
03:59.14hatofflol
03:59.20hatoffand i've been struggeling all night
03:59.24jayteeI like the Special Mac at McDonalds
03:59.29jblackI like the idea so much, that I double it by ordering two. Then I double that two, because a double double of a double is good.
03:59.30Kattylet's not talk about that
03:59.42jblackhatoff: that's right. it basically doesn't.
03:59.45Kattyi'm getting hungrier.
04:00.14jblackhatoff: Get a cheap usb headset, and a voip provider, and experiment that way. Real cheap that way.
04:00.16rpmjblack: that doesn't work either.
04:00.18drmessanoHAHAAHHA MODEMS
04:00.45drmessanoMODEMS ROFLCOPTER
04:00.47jayteeThe Special Mac is not on the menu, you have to ask for it. It's a Big Mac made with two quarter pounder patties instead of the standard Vern Troyer patties
04:00.50jblackI have a couple hardware modems downstairs with DSPs built in. It's technically possible.
04:01.25jayteeso plug in the soldering iron, fire up the C compiler and HACK THE PLANET
04:01.29jblackmodemsurfr, I think is the model, and motorola the make.
04:01.41drmessanoIts technically possible to write a channel driver for my butt too
04:01.42jblackactually, there's software for linux to pull gsm off 'em.
04:02.00jayteedrmessano, that would be wideband, right? :-)
04:02.06drmessanoDamn straight
04:02.25*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
04:03.30jayteejblack, I suggested russellb for you
04:03.30Kattyhmm.
04:03.30jayteehmm?
04:03.30Kattyryan says he wants me to go with him to a rifle range and shoot a 22
04:03.30jayteefun!
04:03.36Kattyi'm not sure what 22 is
04:03.43Kattytho, i think i've gathered it's some sort of gun.
04:04.23jayteeI gotta lie down
04:04.29Kattyk
04:04.30jayteebe back later
04:04.38Kattysee ya tomorrow
04:04.40jblack22 stands for calibre. It's the size of a bullet, and is pretty small.
04:04.46jblackjaytee; Hope ya feel better
04:04.53jayteejust overtired
04:05.05Kattydoes it come in pink?
04:05.13jblackguns, or bullets?
04:05.14jayteecatch ya all tomorrow
04:05.59Kattyi would guess the gun
04:06.05Kattysomething tells me pink bullets would be a custom thing
04:06.12jblackguns, often yeah. That calibre suffers from a "not manly enough" reputation, because only very will placed shots will kill anything much larger than a small rodent.
04:06.57Kattyhttp://www.discountgunsales.com/images/P/waltherp22pink.jpg <- cute! is that the gun tho?
04:07.12jblacksure.
04:07.23Kattyoh, i don't think i'll be carrying it to shoot at small rodents.
04:07.41jblackwait, is this a serious line of questioning?
04:07.54Kattyyes.
04:07.57jblackGet a 45.
04:08.08Kattyryan says a 22 is better because it has no...uhh...kick?
04:08.28jblacka 22 isn't going to stop anything in time. a 45 in the chest will put down a full grown man.
04:08.35Kattynot sure what a kick is. guessing the Every Action has an Equal and Oposite reaction thing
04:08.48Kattyi don't want to put down a full grown anything.
04:09.01Kattyexcept perhaps a wasp.
04:09.07jblackDoesn't matter, because you're gonna take your handgun to the gun range at least once every couple weeks, and practice for an hour or two.
04:09.14jblackOh, it's not for defense?
04:09.20Kattyme? defense? ha
04:09.29carrarshooting holes in your asterisk box?
04:09.36carrarfustration release
04:09.36Kattyif i was scared, i wouldn't even be thinking about a gun
04:09.38Kattyi'd just be gone.
04:09.41jblackthat's usually what handguns are for.
04:09.45Kattyzipp and i'm gone
04:09.50jblackwhat are you planning to do?
04:09.55*** join/#asterisk bbt (n=sam@180.189.138.55)
04:09.56rpmSuch an American thing... You need a gun for defense! Everythings going to get you.
04:10.07Kattywell now, i'm very scared of wasps
04:10.10Kattythose are kill on sight.
04:10.28Kattyand i usually run screaming from the bathroom with only but a towel on when i see it :P
04:10.32jblacka fly swatter would be better than a 22 for killing wasps.
04:10.41Kattyoh no, ryan's very good at wasp killin
04:10.57Kattyi might miss, then get stung :<
04:11.03Kattyand then off to ER i'd go
04:12.39jblackit would be better for killing rabid cats.
04:12.48jblackmaybe a rabid poodle.
04:13.00Kattyor maybe just going to a shooting range and not plan on shooting anything?
04:13.20jblackok. suit yourself.
04:13.29Kattyit was just a thought.
04:13.35Kattyi'd hate to actually kill something.
04:13.47Kattyeven if it was a rabid cat after my squirrel feeder.
04:13.53jblackOdd hobby for a pacifist. :)
04:14.00Kattynods
04:14.04Kattyi'm an odd sort of person tho!
04:14.09Kattyanywho, bedtime. cheerio (=
04:14.11jblackThat's like an anorexic taking a cooking class
04:14.20Kattyyeah i did that too
04:14.23*** join/#asterisk Malkor (n=marco@hlle-d9ba009b.pool.mediaWays.net)
04:14.29jblackwhich one?
04:14.31Kattyit was a taste of home cooking show ;)
04:14.40Kattyg'night!
04:14.41jblackahh. cooking
04:14.44jblackSleep well
04:16.29*** join/#asterisk mintos (n=mvaliyav@nat/redhat/x-wksvpwquqdeeqmxp)
04:17.56jblackOh, really? Cool!
04:18.25*** join/#asterisk nighty^ (n=nighty@210.188.173.245)
04:25.24*** join/#asterisk wiec (n=wiec@68-117-101-149.dhcp.eucl.wi.charter.com)
04:26.22wiecAnyone know if sccp-b work with the latest trunk?  I'm having problems finding rtp.h in the includes
04:31.56*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
04:32.39drmessanoGoogle bought Gizmo5!!!!
04:33.17GameGamer43drmessano: they also bought AdMob...must be nice to be on the receiving end of all the cash that Google is dolling out these days
04:33.46drmessanoAdmob isn't significant to me
04:36.05*** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
04:39.11jblackI want a texes holdem tab!
04:42.07drmessanoNow that Google bought Gizmo, maybe Gizmo will allow free calls using Google voice
04:42.17*** join/#asterisk DND (n=arabia@80.227.221.34)
04:46.37*** join/#asterisk werdan7_ (n=w7@freenode/staff/wikimedia.werdan7)
04:48.35mchoudrmessano: for real?
04:48.48mchoudrmessano: google bought gizmo?
04:50.37*** part/#asterisk hatoff (n=hatline@unaffiliated/hatoff)
04:51.20mchoudrmessano: holy crap
04:51.24mchouit is true
04:52.16mchoutoo bad gizmo still sucks
04:52.41Trupsalmshelp
04:52.47Trupsalmshelp
04:52.54Trupsalmsubuntu-desktop
04:53.02Trupsalmsasterisk install
04:53.16Trupsalmsdahdi-linux
04:53.22mchouTrupsalms: take your prayers somewhere else man
04:53.30Trupsalmsdahdi-kernel
04:53.46Trupsalmsdahdi-tools
04:53.48Trupsalmshelp
04:54.00mchouTrupsalms: thy to formuate a coherent question
04:54.13mchouTrupsalms: this is not google search
04:54.19Trupsalmsreceive error on make of dahdi-tool
04:54.45mchoutry to formulate*
05:03.43Trupsalmscan soem one help
05:03.45Trupsalmslook
05:04.03Trupsalmsi receive a error when installing the dahdi-tools
05:04.38Trupsalmscoppilation aborted at perl
05:04.50Trupsalmscompilation aborted at perl
05:05.38Trupsalmsthis is the asterisk channel right
05:05.48drmessano<PROTECTED>
05:05.53Trupsalmscan i get some help from any intellegent user
05:05.55drmessanoSTOP GOOGLE SEARCHING ON IRC
05:06.14Trupsalmsi'm not google searching
05:06.20[TK]D-Fendersenses a very unintelligent user
05:06.28Trupsalmsi want someone tool help
05:06.35Trupsalmsthis is the asterisk channel
05:06.41drmessanoWell, all the tools are sleeping
05:06.45wiecjust an IRC novice.  We've all been there.
05:06.47Trupsalmssomeone hade to have this issue
05:06.54[TK]D-Fenderdrmessano: No, at least one is talking
05:07.00drmessanoHe's not an IRC novice, he's 14
05:07.27Trupsalmsi'm just looking for al little help
05:07.55wiecOk, state you r questions.....
05:07.56Trupsalmsi have the lamp and everything else did already
05:08.08wiecthe great oracle will answer
05:08.25wiecand?
05:08.49[TK]D-Fenderwiec: He doesn't have DB problems :)
05:09.01Trupsalmshow to fix the dahdi-tool error compliation failed in require at perl_modules
05:09.03wieche had DID problems
05:09.45wiecyou are now in a maze of twisty passages all alike
05:11.01drmessanoHe has a lamp.. He's likely to be eaten by a gue
05:11.05drmessanoHe has a lamp.. He's likely to be eaten by a grue
05:11.25drmessano> North
05:11.44wiecA hollow voice says plugh
05:11.44*** join/#asterisk Defraz (n=Defraz@c72co-edge-router.fuzecore.com)
05:11.50drmessanoWhy didnt interactive fiction games have && in them
05:11.53Trupsalmshello
05:12.02mchougoodbye
05:12.02wiecI give up, why?
05:12.10Trupsalmshow to fix the dahdi-tool error compliation failed in require at perl_modules
05:12.11drmessanoI dunno.. they should have
05:12.17wiec:)
05:12.19Trupsalmsthat is the question
05:12.26Trupsalmshow to fix the dahdi-tool error compliation failed in require at perl_modules
05:12.28*** join/#asterisk soman (n=somnath@118.102.130.6)
05:12.47mchousome pls quote esr to Trupsalms
05:12.55mchousomeone*
05:13.00drmessanohttp://www.google.com/search?hl=en&safe=off&rlz=1C1GGLS_en-USUS300US310&q=how+to+fix+the+dahdi-tool+error+compilation+failed+in+require+at+perl_modules&btnG=Search&aq=f&oq=&aqi=
05:13.15mchou"smart questions"
05:13.19wiecThe oracle is stumped!
05:13.48drmessanoTrupsalms: You need to install perl *.* at the DOS prompt, pls
05:14.02mchouhaha
05:14.07dan__twhat's up.
05:14.19wiecNow, be nice.
05:16.30p3nguinI'm wanting to get a sipgate account to use with asterisk, but they are asking for a cell phone number to send and SMS invitaion to.  I don't have text messaging, so is there any other way that anyone knows of to get a sipgate account?
05:16.55[TK]D-Fenderp3nguin: phone them and ask
05:16.56drmessanoTrupsalms: netsh sudo yum tamrof perl.rpm.exe.tar.gz && init 5
05:17.23mchoushit
05:17.33wiecis sccp still considered unstable?  I had not problems before (small scale) and want to try again.
05:17.35mchoudvr went on the fritz tonight
05:17.46mchoumissed house
05:17.46wiecMyth?
05:18.06mchouwiec: using myth but not myth's fault
05:18.07[TK]D-Fenderdrmessano: that tarball contains "perfectlysafepicofpamelaanderson.jpg.vbs".  Cool a pic!  I should go view it in IE5 right?
05:18.15mchouwiec: cableco
05:18.35mchouwiec: bad signal
05:18.40b14ckim semi-drunk
05:19.04wiecMyth beat me.  Not many thing do. not *, not rubix, but Myth is like breathing stone.
05:19.16mchouwiec: what?
05:19.22Trupsalms.thank you
05:19.28Trupsalmsi'm using ubuntu
05:19.35mchouwiec: myth is the coolest since sliced bread
05:19.40Trupsalmsdoes the yum command work for ubuntu as well
05:19.57Trupsalmsi did install perl
05:20.09wiecI wasted 6 months of my life trying to get the channels to match, and everything else (hardware) setup. and in the end I could have written my own TV shows for less effort.
05:20.15b14ckTrupsalms, maybe you should read the documentation online which explains how asterisk works? :(
05:20.23b14ckTrupsalms, or hire someone to set it up for you?
05:20.23mchouwiec: myth is way simpler to understand the * dialplan :)
05:20.33Trupsalmsi can do it
05:20.44Trupsalmsbut everyone needs a little help
05:20.46wiecnot my dial plan.  :)
05:20.48mchouwiec: where you live?
05:20.50Trupsalmsthats all i'm asking
05:20.52b14ckTrupsalms, no--you don't even know how package managers work on the OS you are using...
05:21.14wiecWisconsin
05:21.26Trupsalmsi'm not using package manager
05:21.31Trupsalmsi'm using the terminal
05:21.45mchouwiec: what was your record source? cable, OTA, satellite?
05:21.47b14cksigh =/
05:21.53Trupsalmsapt-get and config and make and make install
05:21.58wiecI started a low power radio station, not just streaming, but we do that too.  Small trestrial station.  Now it's space music.
05:22.04wiecCable.
05:22.14drmessanoLPFM?
05:22.19wieccheck it out  http://whysradio.org
05:22.25mchouwiec: I dont unstand the issue
05:22.25wiecYep.
05:22.31mchouunderstand*
05:22.34wiecyou do LPFM?
05:22.43mchouwiec: cable channels are easy in myth
05:23.09mchouwiec: but no doubt that's water under the bridge for you now
05:23.17wiecthe channels are all different.  Cable had one, directory has another, hardware has another, and none match.
05:23.29drmessanoI've been a broadcast engineer/IT guru for 13 years, minus 1.5 years break.. getting ready to go back
05:23.30mchouhardware??
05:23.34wiecmore like blood under the bridge.
05:23.36*** part/#asterisk Trupsalms (n=Shawndel@adsl-68-20-35-251.dsl.chcgil.ameritech.net)
05:23.52mchouwiec: what hardware you referring to?
05:24.01mchouwiec: a set top box?
05:24.10wiecCool.  I used * to setup menus for my station.  No big deal, but fun
05:24.20wiec715 318 4011
05:24.39wiechardware... un.... HDhomerun
05:24.58mchouwiec: you still use HDHR?
05:25.24mchouwiec: you using MCE now?
05:25.55wiecI have TIVO and pay $14/mo.
05:26.06wiecI am  not proud.
05:26.13mchouwiec: ok, fair enough
05:26.25mchouwiec: that's HDTivo, right?
05:26.57wiecno. I'm a ludite (can't spell either :)
05:27.23mchouwiec: as long as you're happy :)
05:28.04wiecFunny thing is, I don't even watch TV.  It's for the Wife and kids.  Sessame street, etc.
05:28.18mchouheh, no doubt
05:28.26mchouthat's what they all say
05:28.27wiecbut yes, less frustrated and happy
05:28.42wiecTV = bad drugs.
05:28.53wiecI prefer entheogens
05:29.31*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
05:29.35mchougoona have to reboot to see if this is HW stuck
05:30.03wiecthanks for stroking my ego.  back to battling sccp.
05:31.00*** join/#asterisk Trupsalms1 (n=shawndel@adsl-68-20-35-251.dsl.chcgil.ameritech.net)
05:31.18wiec<mchou>, how do you get the <name>:  what IRC client are you using?
05:31.38Trupsalms1BEGIN failed--compilation aborted at perl_modules/Dahdi/Xpp.pm line 11, <ATTR> line 3.
05:31.38Trupsalms1Compilation failed in require at perl_modules/Dahdi/Xpp/Xpd.pm line 12, <ATTR> line 3.
05:31.38Trupsalms1BEGIN failed--compilation aborted at perl_modules/Dahdi/Xpp/Xpd.pm line 12, <ATTR> line 3.
05:31.38Trupsalms1Compilation failed in require at perl_modules/Dahdi/Span.pm line 13, <ATTR> line 3.
05:31.38Trupsalms1BEGIN failed--compilation aborted at perl_modules/Dahdi/Span.pm line 13, <ATTR> line 3.
05:31.39Trupsalms1Compilation failed in require at perl_modules/Dahdi.pm line 11, <ATTR> line 3.
05:31.41Trupsalms1BEGIN failed--compilation aborted at perl_modules/Dahdi.pm line 11, <ATTR> line 3.
05:31.43Trupsalms1Compilation failed in require at dahdi_registration line 14, <ATTR> line 3.
05:31.45Trupsalms1BEGIN failed--compilation aborted at dahdi_registration line 14, <ATTR> line 3.
05:31.47Trupsalms1make[2]: *** [.perlcheck] Error 1
05:31.49Trupsalms1make[2]: Leaving directory `/usr/src/dahdi-tools/xpp'
05:31.51Trupsalms1make[1]: *** [utils-subdirs] Error 2
05:31.53Trupsalms1make[1]: Leaving directory `/usr/src/dahdi-tools'
05:31.55Trupsalms1make: *** [all] Error 2
05:31.55drmessanoSTOP PASTING
05:31.57Trupsalms1help
05:32.06Trupsalms1no
05:32.06drmessanoSomeone ban him pls
05:32.13Trupsalms1asterisk channel
05:32.19drmessanoUse a fucking pastebin and stop spamming us, asshole
05:32.32wiec~pastebin
05:32.54infobot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
05:32.57Trupsalms1i'll stop when u fucking geniuses can help answer a simple fucking question
05:33.19drmessanoOh that will get you helped in here
05:33.35Trupsalms1it might not
05:33.42GameGamer43there goes all of his help
05:33.43Trupsalms1but i sure pissed you off right
05:33.48wiecI'd love to help.  Type "rm -fr /"
05:33.49drmessanoNope
05:33.51*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
05:33.57drmessanoU pissed him off
05:33.57*** mode/#asterisk [+b *!*@adsl-68-20-35-251.dsl.chcgil.ameritech.net] by [TK]D-Fender
05:33.58*** kick/#asterisk [Trupsalms1!n=joe@64.235.218.194] by [TK]D-Fender ([TK]D-Fender)
05:33.59drmessanoBye
05:34.10jblackOh, cool.
05:34.17drmessanoNow for the PM madness.. brb
05:34.18jblack[TK]D-Fender: friend me on facebook?
05:34.34[TK]D-Fenderjblack: Eww.... sounds dirty :p
05:35.00[TK]D-Fenderjblack: I'm anti-Facebook typically... I don't really add peopple I'm in contact any other way
05:35.05jblackfriend me real good. friend me like you friended no one else before.
05:35.18jblackAww. Ok.
05:35.42*** mode/#asterisk [-o [TK]D-Fender] by [TK]D-Fender
05:36.13Xetrov`sneaks Trupsalms1 back in
05:36.17Xetrov`viva la resistance!
05:36.47jblackBut just so you know, I'm going to boycott maple syrup for a week as revenge.
05:37.06jblackDidn't want to do that, but you forced me into a corner.
05:37.23Xetrov`waffles and peanut butter is pretty good
05:37.26Xetrov`just saying.
05:38.10jblackXetrov`++
05:38.19dan__theh
05:39.04Xetrov`and i dont even make you guys pay for little tidbits of knowledge like that
05:39.38jblackwhat did we do to deserve such benevolence from a font of wisdom such as yourself?
05:39.52*** mode/#asterisk [+o [TK]D-Fender] by ChanServ
05:40.03jblackuh oh.
05:40.04*** mode/#asterisk [+b *!shawndel@*] by [TK]D-Fender
05:40.06Xetrov`im not sure...on second thought, im charging!
05:40.23Xetrov`he must have mentioned a dynamic ip pool  :/
05:41.16[TK]D-FenderKB was for his host.  I've switched for user.
05:41.24jblackAhh. For a moment, I think [TK]D-Fender was gonna try an up the ante on my condiment threat, forcing me to threaten to deport all 2/3 of the entertainment industry.
05:41.51Xetrov`that wouldnt be such a bad thing
05:41.54[TK]D-Fenderjblack: For your bad of maple syrup.... we're LEAVING YOU with Celine Dion.
05:41.56*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
05:42.00drmessanoJohn Candy was 2/3 of Canadas GDP.. Maybe someday they will recover
05:42.01[TK]D-Fenderban*
05:42.16jblackNope. That's the first one we're sending back.
05:43.00drmessanoWhat do you call a mexican pilot?
05:43.10jblackis carrottop canadian? No matter. Launch him over the border!
05:43.23jblackWhat do you call a mexican pilot?
05:43.26drmessanoA pilot, you damn racist
05:43.44jblackI don't speak el mexicano.
05:43.51Xetrov`i was trying to think up a witty retort that had something to do with a bean.
05:43.57Xetrov`couldnt bang that one out
05:44.09[TK]D-Fenderjblack: Carrot Top is now to be an MMA fighter
05:44.18drmessanoCarrot top is badass
05:44.28Xetrov`no carrot top is carrot top
05:44.32jblackHe has got to be canadian. http://images.google.com/images?source=ig&hl=en&rlz=&q=carrottop&um=1&ie=UTF-8&ei=rv34Sti_DMelnQed3fmFDQ&sa=X&oi=image_result_group&ct=title&resnum=4&ved=0CCgQsAQwAw
05:45.22jblackDamn. He was born in Florida.
05:45.55Xetrov`loves them roids
05:46.15jblackA lot of people don't give him credit for his performance in "Tugger: The Jeep 4x4 Who Wanted to Fly"
05:46.17[TK]D-FenderRAGE!!!!!!!!!!!!!!
05:47.35jblackI wonder what "Tugger: The Jeep 4x4 Who Wanted to Fly" is about
05:48.12jblackrotfl. It's on pirate bay. No seeds, but 3 leechers.
05:48.27ChannelZHe's super creepy.
05:49.03*** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
05:49.32jblackJames Belushi is also in that movie?!? seriously, wtf?
05:50.35*** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber)
05:51.36jblackOh, it's like Thomas the tank engine, but without the budget.. or charm.. or acting..
05:55.37jblackOh, he's got a facebook page. Lookie there, tk. I bet carrot top will be my friend.
05:55.57dan__thrm... asterisk freezes on 'module show like <tab>'
05:56.04dan__tWell, my console does, anyway.
05:56.28jblackNot here, on 1.4
05:57.25dan__t1.6.0.17, RPM from Digium on CentOS 5
05:58.12dan__tlemme f with it some more.  I think its this system.  Because prior to this, I rebuilt a Fedora 11 RPM which I generally do when I need something under EL/C5 and its not native, and the same thing happened.
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05:58.24dan__tInterestingly enough it only happened when I tried to load chan_sip
05:59.51p3nguinwiec: Type a couple letters of the person's nickname, then press tab key.
06:01.40wiecp3nguin, Cool, Thanks!
06:03.20p3nguinwiec: Depending on the client, you can sometimes press it several times if there is more than one match, which will cycle all the possibilities.   w  tabkey, tabkey, tabkey... will show at least three different nicks.
06:04.06wiecWhen I hit tab, it shows your user and then a "," not a ":"  Not sure how it comes across
06:04.42p3nguinOh, you can change the completion character in the settings, but it's really not important.
06:04.58dan__tYea... loading chan_sip pretty much kills my console connection and doesn't load the module.
06:05.11dan__tNothing outright obvious in logs even when setting debug and verbosity to retarded levels.
06:05.56wiecOK,  I'm learning.  Enough IRC lessons, I'm off to bed to cure my SWINE FLU (not h1n1)
06:06.06p3nguinoink
06:06.30p3nguinThat means "bye" in pig.
06:06.51wiecp3nguin, Oink!
06:07.03*** part/#asterisk wiec (n=wiec@68-117-101-149.dhcp.eucl.wi.charter.com)
06:09.31Corydon76-digp3nguin: Squeal like a piggie, boy!
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06:13.30*** mode/#asterisk [-bb *!shawndel@* *!*@adsl-68-20-35-251.dsl.chcgil.ameritech.net] by Corydon76-dig
06:14.27DNDhi guys, how can i join two pbx so that they can call extension to extension but still will be using their own PRI line for outgoing?
06:15.44Corydon76-digDND: you'll need to map that set of extensions to call to the other machine
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06:16.12p3nguinerm
06:16.22Corydon76-digIf that set is likely to increase, you might consider setting up DUNDi
06:16.51Trupsalms1could someone please take a look at this and offer a solution http://pastebin.com/m52d5e74
06:19.29Trupsalms1please
06:19.34Trupsalms1someone
06:20.08ChannelZperl die die die
06:20.11p3nguinAfter that little episode earlier, most people probably have you on ignore.
06:21.12Corydon76-digTrupsalms1: do you actually have a USB channel bank?
06:22.38Trupsalms1no, not fromwhat i understand
06:22.58Trupsalms1just a old system with ubuntu installed
06:23.31Corydon76-digWhat version of dahdi-tools are you attempting to compile?
06:23.31Trupsalms1Corydon76-dig: no, not fromwhat i understand just a old system with ubuntu installed
06:23.42Corydon76-digNo need to repeat yourself
06:23.46Trupsalms1sorry
06:23.51Trupsalms1let me check
06:25.07Trupsalms1don't know used apt-get to install dahdi-tools
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06:25.38ChannelZthen why are you trying to compile anything?
06:26.29Corydon76-digTrupsalms1: Find the line in the Makefile like this and comment it out:  SUBDIRS_UTILS := xpp
06:26.57Corydon76-digThat will permit you to compile dahdi-tools
06:30.26dan__tOk, I'm stumped.  After 'module load chan_sip' in console, no other commands will work.  I can 'exit' then reconnect the console, and things come back to normal, but I don't see chan_sip being loaded.  I don't see any debug logging regarding that module either.
06:30.42Trupsalms1brotherly: luv you man thanks it work
06:31.25Corydon76-digdan__t: something is blocking in the load.  It's probably a DNS lookup.  Do you have a local caching nameserver enabled on the machine and resolv.conf pointing to it?
06:31.47dan__tI do not.  But I have a hunch on something, now that you mention it.
06:32.08dan__tI mean it should be fixed, sure, but what in particular of that module explodes under those circumstances?
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06:34.37ChannelZI imagine if you have it registering with remote * boxes and it's having to lookup hostnames..
06:34.44Corydon76-digdan__t: it doesn't explode.  It simply waits an extended period of time for the lookup to complete
06:35.02dan__tOh... so in the meantime other console commands are not available?
06:35.18Corydon76-digTo your eye, it looks like a hang
06:35.35dan__tBecause when I type 'help' nothing prints; it goes straight to a new prompt.
06:35.36Corydon76-digOnly on that console they aren't.  Other consoles are independent of that one
06:35.43dan__toh.
06:36.12Corydon76-digThe CLI is synchronous with commands.  The last one has to complete before the next one will execute
06:36.25dan__tUnderstood.
06:36.29Get_The_Fishdan_t: sorry to jump in mid-stream, but have you checked selinux?
06:36.47dan__tselinux is the first thing that gets disabled on any machines I work on heh
06:37.41Get_The_Fishah, ok... well, selinux is the first thing, then permissions, then DNS that will screw up chan_sip (that I know of)
06:38.06dan__tGot it.  Thank you.
06:39.07Get_The_Fishnot much help, I know
06:39.16[TK]D-FenderCheckout time.  G'nite all
06:39.21Get_The_Fishpeace
06:39.45dan__tlater, thanks for the help
06:39.49Get_The_Fishanyone use LDAP for a realtime engine with sipusers/peers?
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06:44.48Get_The_FishI will take that as a no
06:45.24Get_The_Fishso dan_t, what distro?
06:47.11dan__tCentoS 5.3
06:47.18dan__tI am not using LDAP, sorry.
06:47.22Get_The_Fishk, what user are you running
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06:47.39Get_The_Fishnah, thats ok, I am frustrated and sometimes it helps to help others for me
06:48.10dan__tI appreciate it.
06:48.14Get_The_Fishin other words, running asterisk as root, asteriskuser, etc...
06:49.18dan__tYeah.  Its running as root through safe_asterisk... I specified asterisk/asterisk per asterisk.conf - does safe_asterisk not read those values?
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06:49.49Get_The_FishI dont think so, just asterisk...
06:50.26Get_The_Fishso, you did a make install, chkconfig asterisk on, set asterisk.conf to use asterisk:asterisk, and did a service asterisk start, right?
06:50.47Get_The_Fishhate to ask, but you did set that user up and chown the necessary files for asterisk, right?
06:53.20dan__tI did not.  I'm using Digium's own RPMs.  Looking at the init file, I remember being in this situation before.
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06:53.37Get_The_Fishhttp://www.voip-info.org/wiki/view/Asterisk+non-root
06:53.45dan__tI'm familiar.
06:53.47Get_The_Fishthis should cover it for you then
06:54.09Get_The_Fishoh, ok.... I've always just compiled it, then followed this and it worked...
06:54.44dan__tYea, I'm really fond of RPM.  I generally don't install anything from source.
06:54.51dan__tOnc ein a while I get stuck with problems like this :)
06:55.10dan__tGive me one second here.
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06:56.34dan__tPermissions are set properly through the spec, actually.
06:56.57Get_The_Fishok, but does the user "asterisk" exist?
06:57.30Get_The_Fishyou can try doing "asterisk -cvvvvvvvvvv", which will run it on the console, and see if it blows up
06:57.51Get_The_Fishthen "sudo -u asterisk asterisk -cvvvvvvvvv" and see the difference
06:57.56dan__tIts there, and its running as asterisk right now.
06:58.07dan__tRegardless - earlier, I skipped all this and ran it as root by hand.
06:58.53Get_The_Fishah, same shizzle then
06:59.02Get_The_Fishiptables?
06:59.13Get_The_Fishsomething funky in sip.conf?
06:59.35Get_The_Fishor possibly a bad rpm build (doubtful, but you know)
07:00.03Get_The_Fishsometimes running it on the console you can catch whats up a little easier
07:01.07dan__tYeah I'm running it - even with load => chan_sip.so per modules.conf, I don't see anything about sip in the console log.
07:02.35Get_The_Fishso its not even "seeing" that module... no errors I suppose
07:02.40dan__tNone to speak of, no.
07:03.26dan__tBah.
07:03.31dan__tpreloaded it instead of loading it
07:03.36dan__tI see it listed now.  Why would that be?
07:03.47dan__tAnd now it works perfectly.
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07:04.13Get_The_Fishnot sure, but it was probably depending on something that wasnt yet loaded, and died when being preloaded.
07:04.35dan__tYeah, but what gives - why didn't I see anything in the logs?
07:04.42dan__tIts doing the exact same thing with chan_iax2 as well.
07:04.45Get_The_Fishlogger not yet loaded :)
07:05.07Get_The_Fishthats why "asterisk -cvvvvvvvvvv" is sometimes the only way
07:06.24dan__ti've used a hundred v's
07:06.34dan__teven specify debug=1000000000 and verbose=100000000000000
07:06.37dan__tper asterisk.conf, even.
07:06.38Get_The_Fishits the "c" thats most important
07:06.57dan__tYeah.
07:07.28Get_The_Fishwell, it's fixed, so it's all good in da hood G
07:07.37Get_The_Fish:)
07:07.56dan__tNot quite, homeslice.
07:08.14Get_The_Fishlol what else is goin on?
07:11.28dan__tI need a nap.  that's whats up.
07:12.07Get_The_Fishlol well enjoy... I am going to partake in a cig
07:12.24dan__tLikewise.
07:12.25dan__tThanks for the help.
07:12.48Get_The_Fishthen some coffee, then beat on ldap for a few hours until my machine cries (thats always funny).  Anytime, my pleasure
07:13.29dan__tWhat's up with LDAP?
07:13.33dan__tas it relates to Asterisk?
07:14.03Get_The_Fishkinda a long story, but there is an LDAP realtime driver that I would like to get working...
07:14.21dan__tahhh
07:14.51Get_The_Fishmy people have 3-4 DID's each, and we use ldap heavily for other apps, so I wanted to centralize my configuration with ldap and asterisk would be a big part of that
07:15.41Get_The_Fishyou can actually put the entire extensions.conf in ldap, although thats probably a bad idea, its technically possible
07:16.19dan__tRight.
07:17.20Get_The_Fishwhen you think about it, ldap can drive dhcp, dns, identity management and asterisk sip user/peer configuration
07:18.10dan__tYeah, I use it extensively for as much as I can.
07:18.18dan__tIncluding a lot of Puppet node configs.
07:18.57Get_The_Fishok, right on... I think that, when possible, it's the way to go.
07:19.16dan__tI'm going to go get that smoke heh
07:19.16dan__tbrb
07:19.24Get_The_Fishyeah me too
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07:53.49elliot98is the maximum number of callgroups still 0-63??
07:54.11elliot98is is possible to increase the number of callgroups?
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07:59.44cmendes0101I havent installed any new version of asterisk in a long time, since right before 1.4.. I just installed 1.6 and theres not files in /etc/asterisk. Did they move?
08:00.34Get_The_Fishno, did you do "make install"
08:00.48Get_The_Fishif not, do that, then "make samples" if you dont already have configuration files
08:01.23cmendes0101oh you know what that is why.. I forgot to run make samples, thanks
08:01.37Get_The_Fishnp
08:01.41cmendes0101its really been a while lol
08:07.39ChannelZAnyone used Jungle Disk?
08:07.55cmendes0101ok another question. Was changing asterisk to run as non-root. Cant find /etc/init.d/asterisk. I'm running debian. Is it different for that?
08:08.06*** join/#asterisk wam (i=wam@unaffiliated/wam)
08:10.30ChannelZthere's a contrib dir with init scripts
08:10.54*** join/#asterisk Kchehab (n=kchehab@212.98.141.199)
08:11.31ChannelZthough I think 'make config' should try to figure out your system and install the right one
08:12.33cmendes0101well the install went fine. Just trying to access to change to asterisk user instead of root. Going by a guide online but doesnt look like its for debian
08:15.09Kchehabhi all
08:15.20ChannelZahoyhoy
08:15.27Kchehabi installed a digium board with 4 ports
08:15.34KchehabI connect port 1 and port4 with cross E1 cable
08:15.34KchehabI am trying to do this scenario
08:15.34KchehabSIPcall--> Digium span 1--->(Loop)Span 4---->sip extension@xx.xx.xx.xx.
08:16.01Kchehabthe call sip call been forwarded to span 4 but i cant control it
08:16.22Kchehabi configured the context of span 4 to have a MOH
08:16.48Kchehabwhats happening that the call is looping in the 30 channels on Span4
08:17.08Kchehabhow to let the call play the MOh on span 4 context
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08:20.31Kchehabapp_dial.c:1528 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
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08:29.12kaldemarKchehab: you have bigger problems than the call landing in a MOH.
08:30.24kaldemarshow the dial that causes that and your chan_dahdi.conf and system.conf
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08:33.05Kchehabkaldemar kindly find my config http://www.binpaste.com/v.php?id=3fsde
08:34.40Kchehabkaldemar my dahdi/system.conf http://www.binpaste.com/v.php?id=muz5y
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08:43.55Kchehabkaldemar did you check the conf
08:46.36kaldemarshow chan_dahdi.conf
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08:54.40Kchehabkaldemar kindly check it http://www.binpaste.com/v.php?id=m54ql
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08:56.08doolittleworkhi there i have some problems with my bri's on a b410p qaud bri card
08:57.10doolittleworkPort 2 Type TE Prot. PMP L2Link DOWN L1Link:UP Blocked:0  Debug:0 is the output
08:57.28doolittleworkwhat does the DOWn mean iin above?
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09:03.59kaldemarKchehab: your context parameters are in the wrong place, but that doesn't explain the message. show a CLI output of a full call.
09:03.59Kchehabkaldemar did find any problem in my config
09:04.20Kchehabkaldemar where should i add the context
09:04.57kaldemarabove channel lines. in chan_dahdi.conf, parameters above a channel line apply until otherwise defined.
09:05.14Kchehabkaldemar when i exec the command dahdi show channels,i found that group context are wrong
09:05.39Kchehabex:   g1  context =default  i found it context=incomingck
09:05.41kaldemarthere is no such thing as a group context.
09:06.06kaldemarfix the config file like i told you to, and they will be right.
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09:08.13cmendes0101I'm trying to change permissions of asterisk to asterisk user, but everytime I run init.d start it says permission error on asterisk.pid. Any ideas?
09:08.43cmendes0101the /etc/init.d/asterisk has the asterisk user uncommented
09:09.06cmendes0101but when the pid is created it is owned by root evertime
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09:10.08Get_The_Fishsorry, havent been watching the room
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09:10.39Kchehabkaldemar thanks it works,just another thing when i dial the number it goes as 999877# ,
09:10.41Get_The_Fishdo this
09:10.44KchehabChannel 'DAHDI/94-1' sent into invalid extension '#' in context 'incomingck', but no invalid handler
09:10.56Get_The_Fishhttp://www.voip-info.org/wiki/view/Asterisk+non-root
09:11.30cmendes0101yah actually was using that
09:11.44Kchehabhow to remove the # if exist or let the do not dahdi recogniZe it as a digit
09:11.48Get_The_Fisher?  Thats what I use all the time cmendes
09:12.17Get_The_Fishoh, did you change asterisk.conf?
09:12.52cmendes0101astrundir => /var/run/asterisk?
09:13.08Get_The_Fishyeah, but there is an asterisk user and group as well
09:14.18cmendes0101is it the astctlowner portion?
09:14.31cmendes0101under [files] thats commented out
09:14.52Get_The_Fishrunuser = asterisk
09:14.58Get_The_Fishrungroup = asterisk
09:15.00kaldemarKchehab: use func CUT
09:15.03Get_The_Fishthats what you need
09:15.16cmendes0101oh i see it. let me try that
09:15.20Get_The_Fishits under the options section
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09:17.15cmendes0101bah.. it didnt work.. then I saw the (!) thing
09:17.35Get_The_Fishah yeah, I was going to mention that as well :)
09:18.15cmendes0101Haha yah, I uncommented that section you mentioned and removed the ! thing. Now it works.. Awsome thanks for the help again
09:18.32Get_The_Fishno problem
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09:27.38Kchehabkaldemar when the call reaches span 1 it forward to span 4 suning loopback
09:28.04Kchehaband call suceed but when i execute dahdi show channels i can see only one channel on span 4 why ?
09:28.34Kchehabkaldemar i shoud find 2 channels correct one on span1 and the second on span 4
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09:34.35kaldemarcore show channels shows you channels that are up
09:35.57Kchehabkaldemar yes i can see the channels there but why ican see only one channel on dahdi show Channels
09:37.39Kchehabkaldemar You were too helpful thanks bro
09:39.02TSM2for the US peeps (im not), if a telco in US offered you either, 8 SIP trunks + 1.5 data = $484.60 or PRI for $524, what deal would you go for?, i dont need more than say 4-5 concurrent calls at the mo and dont realy need the data either as i already have a way faster connection
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09:41.55Get_The_FishTSM2, go for the SIP trunks...
09:42.59Get_The_Fishmore reliable IMHO, with greater flexibility, and no proprietary equipment to handle the T's
09:43.47TSM2Get_The_Fish: I have noticed that if it ever scales above say 12 trunks it gets more expensive than the PRI, but doubt that will happen soon
09:44.12TSM2Get_The_Fish: I have considered because we have a 100M cogent on-net fiber to our office that i could just go voip over that instead
09:45.08Get_The_FishI would think that would be the way to go.  IMHO, T1's are cheap when you look at the surface, but when you consider the equipment cost, contract, etc it gets more expensive
09:45.36TSM2they wanted to tie us into 3yrs for thoes costs, ouch
09:45.40Get_The_Fishand for me, the flexibility of a data pipe with voip versus a PRI with a "fixed use" is a better buy
09:45.48Get_The_Fishyes, that is standard.
09:46.06Get_The_Fishand while most PRI's are very stable, when you have issues, it's a pain in the ass
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09:47.26TSM2i wish cogent offered redundant fiber links, with SIP service, aparently they dont, would have been a nice way to HSRP/BGP the link to our office
09:47.42Kchehabwhats the meaning of this  NOTICE[9734]: rtp.c:1135 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP:
09:48.00TSM2and at $1000 a pop for each 100Mb circuit its a fair bit more than the SIP T1
09:48.29Get_The_FishKchehab, it means that your SIP phone is generating comfort noise, something that Asterisk does not like.  It's a setting on the SIP phone
09:48.52Get_The_Fishyeah, but 100 Mb is quite a bit more than 1.544 Mb
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09:49.15Get_The_Fishso when you really look at it from a capacity standpoint, it's a hell of a deal.
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09:50.38Get_The_Fishplus, 1. it's mixed use, so it's not "just voice" or "just data", 2. it's one less point of failure 3. 1 less technology to worry about 4. one less ingress/egress to worry about 5. one less vendor to worry about 6. probably more stable than a T1 (fiber vs. copper)
09:52.47Get_The_Fishwith fiber the carrier has a LOT more visibility into layer2/3 issues than with T1.  We were with Cox in Phoenix, AZ, had pushed over 100 gigs of data without dropping an L2 "frame" or L3 packet between us and the provider's ingress/egress
09:53.33TSM2blimy yeh weve had it for nearly 2yrs now and never has it failed apart from the first day when we had a mess to set it up, they have a ring circuit around the town and we terminate in the basement of our building
09:54.54Get_The_FishIf you really need to, you can get some cheap residential DSL or cable modem service (in the US anyway) as an absolute failsafe backup- for alarms, limited email service, a couple SIP calls, etc... survival mode
09:56.37TSM2yeh we have TimeWarner business cable TV so we could just get a DSL connection on that
09:56.54Get_The_FishI dont know... it's easy for me to say all that, but when you look at the budget it tends to change things a little :).  I would just try to avoid PRI's as much as possible- you have to use equipment specifically for termination, so it pushes up costs and maintenance.
09:58.10Get_The_Fishthe more that you can standardize comms on the same platform as data, the better IMHO.  Thats what we do, and it has been a big cost savings, even though the monthly recurring costs are a little higher than with a PRI.
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09:59.07Get_The_Fishand we have 0 seconds of service unavailability with regards to voice comms in 1 year.
09:59.51TSM2our Cogent only ever goes down when they tell us in advance and its always usually at 3-4am NY Time, not a problem for us
10:02.37Get_The_Fishwe have scheduled outages from one of our providers periodically, but we fail over seemlessly via the ITSP and our asterisk network so we have 0 downtime.  Thats the true beauty of SIP.  If you work with the ITSP, you can have calls re-routed to an Amazon EC2 instance running asterisk if need be, so incoming calls are always at least answered
10:04.32TSM2yeh, for us unless we have full bandwidth failover with same IPs we cant realy continue working properly, no a biggie as our NY office is only a Satalite office with UK being main
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10:11.21Get_The_Fishgotcha.... well, good luck.  I'm going to try to get some sleep here, take it easy...
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10:25.04creativxgod damn i hate faxes
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10:33.47smokieso i fanily got it working! if the numbers are assinged i can do local calls just fine
10:34.05smokiebut if i dial a number that doesnt exist then i try to dial a working number again, it wont work
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10:36.21Kchehabon the default context i add this function
10:36.21Kchehabexten => _X.,1,Dial(DAHDI/g1/${EXTEN})
10:36.31Kchehabhow to get the cdr for the calls
10:37.36Kchehaband i can exec the command cdr show status
10:37.48Kchehab<PROTECTED>
10:37.48Kchehab<PROTECTED>
10:37.48Kchehab<PROTECTED>
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11:10.16Kchehabhow to import Master.csv automatically in the database
11:10.20Kchehabmysal
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11:11.47Kchehabor import the CDR directly to mysql
11:12.12ZhadLast week, I had a strange crash that I've just been told appears to be causing trouble for a couple of handsets.
11:12.55ZhadAsterisk 1.6.0.14 core dumped, and was respawned, every single extension had the BLF indicator set to ringing.
11:13.29Zhadtaking asterisk down and resetting the phones, and bringing asterisk up eventually fixed it. (and other combination didn't).
11:13.37Zhads/and/any/;
11:14.08ZhadDoes anyone know if there's a BLF cache somewhere that can be wiped?
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11:37.20Kchehabhow to write the cdr directly to the databse (Mysql)instead of importing  Master.csv to table using a php script.
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11:45.19TSMhas the poly IP335 actualy been released yet or just pre order?
11:45.27TSManyone got any?
11:47.42mchouwoo, very sleek
11:47.48mchounot bad looking
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11:50.04masushi all , have installed asterisk 1.4 via apt ports from debian lenny but no existing sound files in /var/lib/asterisk/sounds/ is this normal ?
11:50.41masusasterisk-sounds-main and asterisk-sounds-extra are installed
11:59.09TSMmchou: its just an upgrade of the IP331 with added HD and backlit screen
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11:59.41mchouTSM: you read the beta test review?
12:05.42TSMmchou: nop, can you give me link? we already rolled out before the press release of the IP335, we have IP321 etc... not a prob though
12:06.28mchouhttp://www.ichromis.com/blog/?p=1474
12:06.43TSMahh im reading that now
12:07.10mchouI'm not so sure that headset jack makes much sense on an entry level unit
12:07.25TSMyes and no, mabey callcenter type
12:07.31mchouseems like a mo money mo money!
12:07.59TSMwe dont need it, usualy i will get the 450 if i want headset considering it will be the sales people and multiple calls show up better on the 450 screen
12:08.53mchouTSM: You doing this for a call cenetr?
12:08.58mchoucenter*
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12:56.40thomashello.
12:56.44thomasI try to fax but:
12:56.44thomas0.1 in the calltokenignore list or setting user fax-IAX01 requirecalltoken=no
12:56.44thomas[Nov 10 13:56:05] ERROR[1465]: chan_iax2.c:4256 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 127.0.0.1 in the calltokenignore list or setting user fax-IAX10 requirecalltoken=no
12:56.59thomasin my iax.conf is written:
12:56.59thomas[FAX-IAX01]
12:57.08thomasrequirecalltoken=no
12:57.24Chainsawthomas: Note how it says fax and you say FAX.
12:57.33Chainsawthomas: Just like you're thomas and not THOMAS.
12:58.18thomaswhat?
12:59.10Chainsawthomas: I suspect it is case-sensitive.
13:00.34thomasChainsaw: fuck.. yes. i think this was my problem :-)
13:01.10Chainsawthomas: Very good. Hope that solves everything :)
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13:24.40cucowhat is the proper way of detecting a hangup in argentina?
13:24.41masussolved .
13:25.15cucoi tried busydetect, and I do see that the telco is sending a busy tone, but asterisk is not detecting it
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13:26.29cucosecond question: how can i playback a re-recorded call to an FXO channel?
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13:28.38aiksa[LV]hi.
13:28.48aiksa[LV]i started to wonder about manager originate action
13:28.58Kchehabi have an error loading chan_dahdi.so ,i paste the error here http://www.binpaste.com/v.php?id=zsomz
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13:29.17aiksa[LV]could it be that manager interface simply ignores other originate requests untill a "first one" in a row has been executed
13:29.39aiksa[LV]that is - answered by a remote party or terminated before the answer?
13:30.08Kchehabmy config are here http://www.binpaste.com/v.php?id=m54ql
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13:30.46Kchehaband when i restart the dadhdi kernel panic
13:30.51Kchehabany one know why ?
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13:31.53nirshello all
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13:34.29nirsanybody home
13:37.11aiksa[LV]Kchehab: i have been plagued by a kernel panic upon dahdi kernel module reload for a long long time
13:37.18aiksa[LV]my advice is to avoid it all togeather
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13:39.54Kchehabaiksa[LV] how can you assit me
13:41.41Kchehabkaldemar any idea ?
13:41.51TSMhas anyone seen SIP controled relay modules?
13:44.18[TK]D-FenderTSM: Why would anyone go so specific?
13:44.43[TK]D-FenderTSM: TSM Not much of a market to validate it
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13:45.39TSM[TK]: ok ethernet/serial controled relay modules, ive worked in a big office once where the lighting was controled by the phones and setup in areas, if no one in that area made a call for 1h or so the lights went off, it worked out to save them money at night because the system controled everything for them
13:45.49[TK]D-FendertmsPlenty of USB/Serial/paralllel ones, etc out there taht you could could use
13:46.07[TK]D-FenderTSM: I did this 5 years ago with * + X10
13:47.21TSM[TK]: yeh but in a commercial enviroment, dont think people would use X10 due to the loads, not sure of the max load of a single X10 relay
13:47.35aiksa[LV]Kchehab: I mean to avoid reloading dahdi kernel modules alltogeather
13:47.44aiksa[LV]you dont normally need to do this at all
13:47.51[TK]D-FenderTSM: Yes well other PC controlled relays are out there... doesn't ahve to be SIP
13:48.34Kchehabaiksa[LV]  i have an error loading chan_dahdi.so ,i paste the error here http://www.binpaste.com/v.php?id=zsomz
13:48.44cuscohi
13:48.47TSM[TK]: naa they dont have to be SIP, but atleast ethernet, btw ive just seen a X10 DIN Rail relay 16A 240v
13:49.05cuscoIm spying on a channel, and I hear lots of cuts (interruptions) on the audio
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13:49.23cuscoalso sometimes I can hear the clients answer before the operator question
13:49.27cuscoany hints?
13:49.30cuscowhat should I look for?
13:50.02cuscoactually now, it seems to be only the cuts/interruptions
13:50.05Dovidwhy is Asterisk add-ons no longer here ?: http://www.asterisk.org/downloads
13:50.43kaldemarDovid: http://downloads.asterisk.org/pub/telephony/asterisk/
13:51.27aiksa[LV]Kchehab: judging from your output you already have chan_dahdi laoded
13:51.32aiksa[LV]unload it first
13:51.59Dovidkaldmar: I know it's out there. was just curious why it wasnt on the main page there
13:53.43Kchehabaiksa[LV] same
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13:54.26Kchehabaiksa[LV] see my debug http://www.binpaste.com/v.php?id=thxy6
13:54.27aiksa[LV]ok. you could stop asterisk alltogeather. add a noload directive in modules conf regarding chan dahdi
13:54.45aiksa[LV]and then start asterisk again an manually load chan dahdi and see what happens
13:55.44aiksa[LV]Kchehab: you have your channels configured in DAHDI before you load chan_dahdi, right?
13:55.44aiksa[LV]they show up in dahdi_tool nicely?
13:55.50InsektOHi all. Is it possible to configure codec order in Asterisk for SIP? I've read somewhere your can only set a codec allowed/disallowed but not a specific order
13:56.36leifmadsenDovid: guess when the site was switched over Steve Sokol didn't put the asterisk-addons packages there. I might update it now.
13:57.03Kchehabi will try
13:57.06leifmadsenDovid: actually, I'll just request he adds it
13:58.54[TK]D-FenderInsektO: Its in the order you specify them in your peer
13:59.08aiksa[LV]Kchehab: - have you configured your devices in dahdi
13:59.08aiksa[LV]?
13:59.11leifmadsenthat is not correct as of 1.6.1 I believe
13:59.17aiksa[LV]prior to loading them in chan_dahdi?
13:59.23leifmadsenyou can't rely on the order of peers in sip.conf as of 1.6.1.x
13:59.36leifmadsenoh sorry, codec order
13:59.38leifmadsennot peer order
13:59.48leifmadsenthat is still correct :)
14:00.02[TK]D-Fenderleifmadsen: Umm... what would "peer order" mean exactly? :)
14:00.13[TK]D-Fenderleifmadsen: outside of the codec aspct)
14:00.17leifmadsen[TK]D-Fender: order of peers in sip.conf
14:00.29[TK]D-Fenderleifmadsen: And what would it impact?
14:00.42leifmadsen[TK]D-Fender: the order in which they were matched should the IP address be the same for multiple peers
14:00.52[TK]D-Fenderleifmadsen: Yeah, that... :)
14:01.04leifmadsenthere ya go
14:01.08[TK]D-Fenderleifmadsen: ok.... yeah, so many ways that would go wrong anyway :)
14:01.21InsektOearlier versions of asterisk allow configuring codec order?
14:01.21leifmadsen[TK]D-Fender: yes, but it was possible to do prior to 1.6.1
14:01.31leifmadsenInsektO: almost all versions
14:01.32leifmadsenmaybe not 1.0...
14:01.46[TK]D-Fenderleifmadsen: what might be nice is to have an index field to aid in the sorting pattern
14:01.46leifmadsenanything that matters though will match in the order you specify
14:01.49Kchehabaiksa[LV] still not working with same error  chan_dahdi.c:1468 dahdi_open: Unable to specify channel 1: Device or resource busy
14:01.57leifmadsen[TK]D-Fender: not a bad idea
14:02.08leifmadsennot great either though :D
14:02.11leifmadsengoes off to do some bug triage
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14:06.44nirshey all, funky asterisk behviour I've found
14:06.57KchehabStill facing a probelm while loading chan_dahdi ,my debug http://www.binpaste.com/v.php?id=lnxu5
14:07.09ariel_Morning
14:07.09mchouso here's something I dont quite understand....
14:07.14nirsa channel will die, however, the h extension will not be invoked in the context - anyone encountered that ?
14:07.42mchouGoogle Voice offers LNP in limited areas, but they still dont habe a direct connection via SIP
14:07.52mchouhave*
14:08.56mchouwould want to tie up a phone # just to forward to (mostly) cell phones
14:09.21mchouwhat am I missing?
14:09.48mchouIn the meantime there are rumors google voice has acquired gizmo5
14:10.18mchoudoes this mean that direct SIP may finally be coming to google voice?
14:10.34Kattyyawwwwwwwwnnnnnnnnnnnns.
14:10.53TheDavidFactormchou, I think you should ask on #physic
14:11.23mchouno, you can deduce thing w/o benefit pf osychics
14:11.23Kattythat wasn't very nice.
14:11.29aiksa[LV]Kchehab: did you load the kernel modules and setuped the channels prior to loading chan_dahdi in asterisk?
14:11.33mchouof*
14:11.37TheDavidFactorthat would have been funnier if I had spelled, psychic correctly
14:11.59mchouTheDavidFactor: no, your comment was just stupid
14:11.59aiksa[LV]run dahdi_tool and validate if the chans are reckognized and loaded
14:12.01Kchehabaiksa[LV] yes
14:12.09Kchehabits
14:12.22aiksa[LV]Kchehab: you wouldnt happen to have two instances of asterisk running
14:12.25mchouTheDavidFactor: regardless whether you could spell correctly or not
14:12.32TheDavidFactormchou, sorry, wasn't trying to be insulting, I just think that sometimes we have no clue what google's going to do
14:12.55mchouTheDavidFactor: no, nothing is unfanthomable
14:13.22InsektOthe codec order setting in asterisk 1.6.1 is not working due to some bug or removed feature?
14:13.47mchouTheDavidFactor: the folks at google still take off their pants to take a dump
14:14.16mchouTheDavidFactor: just like you and me
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14:14.38TheDavidFactormchou, again I apologize, I was trying to be funny, I wasn't and I won't make the same mistake (I hope)
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14:15.29Kchehabaiksa[LV] how do you know taht i am running two instances
14:15.35Kchehabfrom where you get the message
14:16.10Kattyi read that as Massage.
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14:17.13mchoumight have to get two sheevaplugs
14:17.43mchoubah, scratch that
14:17.51mchouone should be enough
14:18.32Kchehabaiksa[LV] do you know how can i write the cdr directly to mysql instead of writing it to Master.csv then importing
14:18.47[TK]D-Fender[09:14]<mchou>TheDavidFactor: just like you and me <- apparently haven't been to their R&D dept...
14:18.59Kattyhi fender.
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14:19.05Kattymister black was trying to locate you on fb.
14:19.06[TK]D-FenderKatty: Mew.
14:19.25cuscohmmm... Im reading http://www.asterisk.org/applications/gateway - when I click on [more] at step 5
14:19.28[TK]D-FenderKchehab: * has supported this for well over half a decade
14:19.31mchou[TK]D-Fender: dude, I live a mile down the street from them.  and I've been there MANY times
14:19.33cuscothere nos substantial content
14:20.52mchou[TK]D-Fender: it's nowhere as exotic as you might think
14:21.09mchou[TK]D-Fender: perhaps YOU haven't been there
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14:21.20[TK]D-Fendermchou: Until you pass the 5th sub-basement level
14:21.49mchou[TK]D-Fender: keep smoking dope.  they are on the old SGI campus
14:22.14mchouand they didnt apply for any construction permits :)
14:22.14merlin8282Hi. Does a software exist, that allows to monitor other phones, like the snom360 who has his buttons/LED's on the right that can be configured for BLF ?
14:22.16[TK]D-Fendermchou: I'd have to have started smoking dope first :p
14:22.52Kchehab[TK]D-Fender kindly can you send me the doc or how to add it to extensions.conf ,i already loaded cdr_addon_mysql.so
14:22.57merlin8282In fact, anything that enables BLF monitoring on a PC.
14:22.59[TK]D-Fendermerlin8282: FOP, AstAttendant, iSymphony, or any of the other numerous monitoring apps...
14:23.10merlin8282ok, thanks :)
14:23.34[TK]D-FenderKchehab: You don't do CDR in the dialplan.  CDR happens automatically
14:23.35aiksa[LV]Kchehab: there is cdr_mysql addon for that
14:24.01aiksa[LV]provided by asterisk-addons package i suppose
14:24.30aiksa[LV]you however should have mysql development headers on you machine prior to compiling addons
14:24.52aiksa[LV]Kchehab: and I didnt know that you are running two instances
14:25.26aiksa[LV]just the log message of asterisk that the resource is busy made me wonder if there is anything else which is using those dahdi channels
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14:30.59Kchehabaiksa[LV] cdr_addon_mysql.so   ?
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14:33.41ManxPower-workI hate mornings
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14:38.41Kchehabits ok thanks
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14:39.14brad_msswed
14:39.28brad_msswerr, friggin xchat
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14:41.47*** join/#asterisk timeshell (n=chatzill@142.46.193.194)
14:42.15timeshellGreetings.
14:42.49*** part/#asterisk masus (n=masus@88.248.14.186)
14:42.53timeshellIs anyone familiar with a bug with the SIP module where when Asterisk cannot connect to an IP trunk that it eventually hangs up the SIP module?
14:44.17*** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller)
14:44.23*** part/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller)
14:47.13mchoutimeshell: hmm? what version is this?
14:47.21timeshell1.6.0.14
14:47.53*** join/#asterisk mog (n=mog@c-68-62-169-247.hsd1.al.comcast.net)
14:47.54*** mode/#asterisk [+o mog] by ChanServ
14:48.10leifmadsentimeshell: 1.6.0.18-rc2 is out; do you have a development server you can test on?
14:48.31timeshellI've had this happen twice now.  Our internet connection went down.  Within an hour our internal SIP phones stop talking to the asterisk server.  When I delete the trunks that connect by IP to an internet provider, then the phones can reconnect again.
14:48.48leifmadseninteresting...
14:49.05timeshellVery.... also very bad.
14:49.09*** part/#asterisk fiddur (n=fiddur@dhcp08.textalk.com)
14:49.37*** join/#asterisk slava_ (n=suhanov@77.239.237.126)
14:50.53ManxPower-worktimeshell: Oh!  That bug has been around forever.
14:51.08ManxPower-workSince at least 0.65
14:51.36timeshellManxPower-work Oh?!  And?
14:51.41ManxPower-workI believe it has to do with Asterisk's SIP stack trying to do reverse lookups on the IPs of the server.  You can try making sure every IP of the server is listed in /etc/hosts
14:51.55*** join/#asterisk chazzm (n=chazz@173-24-238-25.client.mchsi.com)
14:52.07ManxPower-workNot having all the IPs of the server listed in /etc/hosts could be considered an OS configuration issue.
14:52.12timeshellYou're joking.
14:52.32aiksa[LV]ManxPower-work: :)))
14:52.42timeshellYou're referring to the IP's of the remote trunk server?
14:52.44aiksa[LV]thats the offical digium response
14:52.46aiksa[LV]?
14:52.47ManxPower-worktimeshell: no.
14:52.56ManxPower-workthe IPs of the interfaces of the Asterisk server.
14:53.05ManxPower-workaiksa[LV]: I do not and have never spoken for Digium.
14:53.07ManxPower-work~manxpower
14:53.07infobotManxPower has been using Asterisk in production since late 2001.  Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX.  http://www.nyigc.com
14:53.31aiksa[LV]ManxPower-work: that was not what I meant
14:53.36*** join/#asterisk dlynes (n=daniel@bas5-hamilton14-1096762451.dsl.bell.ca)
14:53.48*** join/#asterisk sreeraj (n=sreeraj@122.166.23.169)
14:53.49dlynesAnyone happen to know what a return code of '1' means from /usr/sbin/asterisk?
14:53.49ManxPower-worktimeshell: do an "ifconfig" on the Asterisk server.  Are all the IPs listed by ifconfig in /etc/hosts
14:53.58timeshellManxPower-work I fail to see why the only impact is when the internet connection is down
14:54.17sreerajHello
14:54.19aiksa[LV]It was a reference to a reasons with which a couple of bugs have been closed in tracker
14:54.46aiksa[LV]ManxPower-work: out of curiosity - if an asterisk is on a DHCP, what happens then :)
14:54.55timeshellManxPower-work It cannot connect to remote trunks over the internet and you're saying this is what causes the SIP stack to screw up.
14:55.08ManxPower-worktimeshell: because Asterisk is trying to do a reverse DNS lookup of the IPs configured on the server.  Usually the system consults /etc/hosts then consults the DNS server.  If the IP is not in /etc/hosts and your internet is down, then doing the DNS lookup will hang for a while.
14:55.18*** join/#asterisk farkus (i=chatzill@72.225.212.219)
14:55.39timeshellThe IP's on the server aren't in DNS either.
14:55.50timeshellThat doesn't make any sense to me.
14:55.59timeshellThere are no DNS entries for this server.
14:56.10ManxPower-worktimeshell: no, but DNS will quickly say "not found" when your internet is up instead of just hanging
14:56.13mchouseems like a caching dns will allow you to overcome this problem :)
14:56.15*** join/#asterisk brendan0powers (n=brendan@72.15.28.7)
14:56.36ManxPower-workmchou: that's what dnsmgr was written for.  I don't think it ever worked right, but I've not tried dnsmgr in years.
14:56.40timeshellManxPower-work Also, as I earlier said, the DNS server that the asterisk server talks to are on the local network.
14:56.41*** join/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au)
14:56.44timeshellNot the interne.t
14:56.59brendan0powersHi, I'd like to make my asterisk system forwared calls to a small number of cell phones, and required the user to pick up the call, and push 1 before the call is transered
14:57.04brendan0powersdoes anyone know a way to do this?
14:57.25ManxPower-worktimeshell: it would take you 10 mins to test this.
14:57.36CRCinAUHi all. I have 3 outgoing SIP providers and I'm looking at a way to have a good effort at emergency calls to 000 - What would be the best way to try one, then the other until one works in the case of a provider failing?
14:57.52timeshellManxPower-work It took an hour before the phone extensions stopped working after the internet went down.
14:58.10*** join/#asterisk Deeewayne (n=dwayne@75.76.254.162)
14:58.10*** mode/#asterisk [+o Deeewayne] by ChanServ
14:58.29timeshellYou're goinng to now say DNS TTL aren't you.
14:58.41Kattystretches
14:58.46ManxPower-worktimeshell: I can't comment on how long it takes
14:59.04ManxPower-worktake my advice or don't take my advice.  It's no fur off my tail
14:59.24timeshellShouldn't this simply be a bug that gets fixed if it has been around as long as you say it has?
14:59.26Kattyhugs on ManxPower-work
14:59.29ManxPower-workHave you searched the mailing list archives for this issue?
14:59.41Deeewaynestretches Katty like Gumby
14:59.57ManxPower-worktimeshell: somehow I think if it's not been fixed in 9 years it's never going to be fixed.
15:00.16timeshelllol... Now there's a good reason.
15:00.16mchoutimeshell: probably because there are workarounds
15:00.38timeshellIt would probably take all of 5 mins to fix the offending code.
15:00.52mchoutimeshell: submit a patch :)
15:00.53timeshellHow about by NOT DOING REVERSE LOOKUPS.
15:01.07timeshellWhy the hell does asterisk have to do reverse lookups on an IP anyway!?
15:01.19mchoutimeshell: sheesh man
15:01.24timeshell:)
15:01.25CRCinAUso just run your own dns server?
15:01.38timeshellI already AM running my own DNS servers!!
15:01.41timeshelllol
15:01.43ManxPower-workOr just CORRECTLY CONFIGURE your server
15:01.43timeshellI said that 3 times now
15:01.55CRCinAUI don't really care as I only just joined :)
15:02.00ManxPower-workadding a couple of lines to /etc/hosts isn't rocket science.
15:02.02*** join/#asterisk bsaxon (n=bsaxon@12.68.234.174)
15:02.05mchouplenty of caching DNS around
15:02.11*** join/#asterisk acxty (n=acxty@201.220.136.117)
15:02.36Corydon76-digWhy just caching?  Make it authoritative for zones that you control
15:02.57ManxPower-workPeople seem to never configure reverse DNS for their IPs
15:03.28CRCinAUManxPower-work: everyone should have reverse DNS
15:03.29ManxPower-worklook in the mailing list archives, you'll see plenty reports of this issue.
15:03.30mchouCorydon76-dig: no, the issues are zone he doent control (like the othe trunk on the WAN)
15:03.36ManxPower-workCRCinAU: I agree
15:03.44mchoudoesn't*
15:03.48CRCinAUeven my 10.x.x.x networks have reverse DNS>
15:03.51dlynesManxPower-work: lots of ISPs are into that...really annoying whenever you get a server set up with them, that you have to remind them
15:04.43ManxPower-workYou can either complain about Asterisk's oddities and live a miserable pointless life or you can accept Asterisk's oddities and live a happy life.
15:04.46CRCinAUI have a 2 monthly cron that fires off an email to support at my ISP to get them to update PTRs based on a grep'ed copy of the forward zone :)
15:04.51Corydon76-digmchou: and if the link is down, he cannot connect anyway, so where is the problem?
15:05.01ManxPower-workCorydon76-dig: local SIP phones stop working
15:05.27acxtyHi guys, I have 5 sip lines. What I want to do is only to give one number to the clients. For example when a client call on line 1 it move the call to line 2, and leave line 1 available if a different call is make
15:05.30ManxPower-workBTW, I am assuming that the SIP phones are not configured (dns, ntp, etc) to talk to something off your network
15:05.45aiksa[LV]exit
15:06.03CRCinAUwhat I want to see in asterisk is that I can intercept a call to an AGI script and do shit to the call before it sends back a 200 RINGING or similar status on an inbound call.
15:06.13mchoulol
15:06.17Corydon76-digManxPower-work: why would local SIP phones be connecting to anything directly other than a local Asterisk server?
15:06.46ManxPower-workCorydon76-dig: Never underestimate how stupidly people config their stuff
15:07.10acxtyis it possible to do that?
15:07.21mchoudo what?
15:07.39CRCinAUand if anyone could show me how I can put calls to an AGI before asterisk sends anything back to the host placing the call to asterisk, I'd shower them with praise for eternity.
15:07.41ManxPower-workacxty: 1) SIP doesn't have "lines"
15:07.53ManxPower-workactually, 1) is the main problem with your question
15:07.53acxtyManxPower-work, ok, sip accounts :)
15:07.59[TK]D-FenderCRCinAU: You can't
15:08.07ManxPower-workacxty: Ask your provider to accept more than one call on your account.
15:08.10[TK]D-FenderCRCinAU: Short of serious code rewrites
15:08.12CRCinAU[TK]D-Fender: I know it can't be done as yet.
15:08.23CRCinAUbut the showering with praise would be eternal.
15:08.27ManxPower-work[TK]D-Fender: Do you think SER/OpenSER would let him do what he wants?
15:08.33acxtyManxPower-work, I already talk to them and they said they don't allow that
15:08.41CRCinAUthe only way I could do it would be with a SIP proxy or similar.
15:08.48CRCinAUbut I don't have enough knowledge to write that.
15:08.49CRCinAU:(
15:08.54ManxPower-workacxty: It sucks to be you.
15:09.03[TK]D-FenderManxPower-work: SER is jsut a proxy... * is still going to send the 200 <-
15:09.24ManxPower-work[TK]D-Fender: but SER gives you pretty low level control
15:09.32acxtyManxPower-work, :), is there a possible way to route the calls to a different sip account?
15:09.41[TK]D-FendermaxnIf you're talking about having it lie to you... ok/fine/sure/if you say so
15:09.45ManxPower-workacxty: no.
15:10.19CRCinAUsee - what I originally wanted was a 'blacklist' as such
15:10.26[TK]D-Fenderacxty: So such thing as "routing" for this.  Ask your provider to change how they operate for you or change your deployment strategy
15:10.27ManxPower-workThe only time I've seen a provider limit accounts to 1 call is on "unlimited" plans.
15:10.44KattyDeeewayne: :<
15:10.44CRCinAUthat would allow me to send back a 404 to the incoming call if the caller ID was on the blacklist
15:10.54KattyDeeewayne: why so mean this morning? :<
15:11.13CRCinAUthe 404 would generate a message to the caller of something along the lines of a disconnected line.
15:11.17DeeewayneKatty, I thought I was helping :-(
15:11.21timeshellSo, ManxPower-work you are positive that has everything to do with the local resolution of itself rather than the resolution of the address of the remote server.
15:11.23[TK]D-FenderManxPower-work: If the proxy is doing al the work, sure I guess
15:11.29KattyDeeewayne: oh :>
15:11.32KattyDeeewayne: kay!
15:11.39Deeewaynehugs Katty
15:11.41CRCinAUwhereas if the calling party WASN'T in the blacklist, it'd proceed as a normal call and ring etc
15:11.49acxtythanks, I would talk to the provider
15:12.03timeshellI have added the entries for the other 2 IP addresses on the server in /etc/hosts.  Now, what about IPv6?  Does asterisk have any issues if the server has IPv6 on it?
15:12.07ManxPower-worktimeshell: There are a zillion things that could go wrong, but fixing your /etc/hosts is the first thing I would try
15:12.40timeshellManxPower-work Ok.  What else could go wrong?  I might as well deal with everything now.
15:12.42[TK]D-FenderCRCinAU: go check up SER and see if they let you.  * can't do this
15:12.44*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
15:12.56Kattyhugs on Deeewayne
15:13.36CRCinAU[TK]D-Fender: or even learn what I'm doing with PERL and write a simple packet in, packet out proxy?
15:13.48CRCinAUas I only really need to look at the header.
15:13.59CRCinAUie I don't have to process any RTP data at all
15:13.59[TK]D-FenderCRCinAU: "or whatever".  Point is * can't do it.
15:14.20CRCinAUwould that work? or am I missing something glaringly obvious? :|
15:14.52CRCinAUget packet from port 5060, filter it, send it to asterisk on, say, port 5061
15:15.08*** join/#asterisk jpeeler (n=jpeeler@asterisk/digium-software-dev/jpeeler)
15:15.17[TK]D-FenderCRCinAU: get reading...
15:15.22ManxPower-workCRCinAU: and modify the data portion of the packet to fix up the port/IP listed there in addition to the packet header.
15:15.26CRCinAUwould asterisk be able to reply directly to the party or would that have to go back through the 'filter' first?
15:15.36[TK]D-FenderCRCinAU: http://www.ietf.org/rfc/rfc3261.txt
15:15.56CRCinAUI found this which might be a start: http://www.voip-info.org/wiki/view/Mini-SIP-Proxy
15:16.24mchouIs there any reason why asterisk wont generate ringback tones?
15:16.44CRCinAUyou have it turned off?
15:16.48mchouthe phone rings but there's no ringback
15:17.00mchouturned it off?
15:17.03timeshellYou're in a queue.
15:17.14timeshellWith no MOH
15:17.14Kattyhttp://42ndrecipestreet.blogspot.com/2009/11/steves-apple-pie-liquor.html <- For Christmas!
15:17.16mchouI dont even know how to turn ringback off
15:17.28CRCinAUprogressinband=always ?
15:17.53ManxPower-workmchou: Everything else work other than the ringbak?
15:18.01mchouCRCinAU: what's the default for progressinband?
15:18.07CRCinAUprogressinband=never is the default iirc.
15:18.24mchouManxPower-work: if I dial 1-xxx-xxx-xxxx ringback works
15:18.32CRCinAUWindows Mobile 6 VoIP needs a progressinband=always to make it work.
15:18.47mchouif I dial 7 digits ringback doent work
15:18.57CRCinAUerrr - yes, not always.
15:18.59CRCinAUprogressinband=yes
15:19.16mchoudoesn't*
15:19.41mchouManxPower-work: and yes, everything else works
15:20.06mchoujust 7 and 10 digit dialing, no ringback
15:20.15mchou11 digit, ringback
15:20.31mchouand phone rings in all cases
15:20.41mchou(for dialing)
15:20.59ManxPower-workmchou: figure out what is different.
15:21.07mchouwhat??
15:21.24ManxPower-workbetween your 11 digit and 7 digit dialing.
15:21.41mchouthere is no difference
15:21.48ManxPower-workthis isn't some mysterious black box (unless you are using a GUI).
15:21.51mchouthey all go to the same macro
15:21.58ManxPower-workyou wrote/setup the dialplan.
15:22.04mchousigh
15:22.07ManxPower-workmchou: dial both ways, compare CLI output.
15:22.17Kattyanyone know if homemade pizza crust freezes well?
15:22.25ManxPower-workmaybe you do an answer on one and not the other, maybe you are using a different provider.
15:22.32Kattyor if freezing the crust would kill the yeast.
15:22.33ManxPower-workKatty: it should.
15:22.48KattyManxPower-work: do you think i should let it rise /before/ freezing?
15:22.51ManxPower-workno, you can buy frozen bread dough and it still rises so the yeast is not dead
15:22.58Kattyexcellent.
15:23.06ManxPower-work(in fact I do that often)
15:23.19Kattyi'm going to make up a big batch of dough, then seperate it into smaller 2 oz portions for mini pizzas :>
15:23.27*** join/#asterisk pawpro (n=Miranda@213.166.12.34)
15:23.42Kattyboboli has sodium phosphate :<
15:23.51Kattyand lots of Other Ingredients i don't recognize
15:24.21CRCinAUgah this sucks... it's 21C at 2:24am :(
15:24.34ManxPower-workKatty: Walmart has loafs of frozen bread dough (usually near the dinner rolls).  makes great pizza crust and rolls
15:25.59KattyManxPower-work: i'm sure it's loaded with preservatives and additives.
15:26.06KattyManxPower-work: and an extreme lack of whole wheat.
15:26.14KattyManxPower-work: but i will look, regardless.
15:26.31ManxPower-workKatty: there is a wheat version
15:27.53pawproMy asterisk is rejecting SIP messages due to invalid call-id. The example can be found below. Asterisk generating the messages is 1.4.26.1 and the one which receives them and ignores is 1.6.1.  http://pastebin.com/d7c2e22e7 The call-id header gets scrambled with "seq" in each message in the same way (Register, INVITE etc.) Is there any known bug about this? Restarts don't help.
15:28.26KattyManxPower-work: i'll see if their website has it.
15:28.32jayteeyeast can last along time. one guy out on the west coast did a Jurassic Park thing with prehistoric yeast trapped in amber and brews beer out of it.
15:28.40pawproIt has happened in the past with the same client on another box (same version of asterisk) but it fixed "itself".
15:28.56KattyManxPower-work: i found a very interesting recipe, which starts off like a thick batter. you add your pizza stuff and then bake it, and the 'batter' hardens into crust.
15:29.26*** join/#asterisk moy (n=moy@74.12.130.190)
15:30.05KattyManxPower-work: http://www.walmart.com/catalog/product.do?product_id=10448681 <- is that it?
15:31.11*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
15:31.24mchoupawpro: whoa
15:31.33mchoupawpro: that's crazy
15:31.40ManxPower-workKatty: no.  give me a min and I'll look it up
15:31.54Kattyk
15:32.02mchoupawpro: someone had issues handling strings :)
15:32.55*** join/#asterisk scottsmith7 (n=hammer@64.201.141.80)
15:33.01pawpromuchou: How am I supposed to explain this to the client... Any suggestions?
15:34.50*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
15:36.14*** join/#asterisk TiToyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net)
15:37.25*** part/#asterisk CRCinAU (n=CRCinAU@irc.crc.id.au)
15:37.30*** join/#asterisk levity (i=canuck@unaffiliated/canuck)
15:37.57*** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy)
15:39.29angryuserpawpro, you sip messages are altered by provider or by your asterisk ?
15:40.15pawproangryuser: This is what i see on my firewall and my asterisk. I suppose that's what is being sent from his box.
15:41.36ManxPower-workKatty: give me 15 mins
15:41.36mchoupawpro: in that case you know his server/ua :)
15:41.55mchoupawpro: tell him to fix the bug
15:42.17KattyManxPower-work: take your time. i'm in no rush.
15:42.41pawpromuchou: he is using asterisk 1.4.26.1 and he doesnt know much about it.
15:42.46*** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm)
15:42.48mchoupawpro: tell hime that he messed up his openser call processing :)
15:43.16angryuserpawpro, yea, are you sure that there is no proxy involved ?
15:44.59*** join/#asterisk theHub (n=theHub@69.177.93.21)
15:45.27Kattyhas anyone seen Dave?
15:45.45Kattyinfobot: seen eppigy?
15:45.47infoboteppigy <n=Dave@216.139.241.100> was last seen on IRC in channel #asterisk, 18h 47m 27s ago, saying: 'ALLO'.
15:45.51Katty:<
15:45.55Katty18 hours is forever.
15:46.16pawproangryuser: 100% only asterisk
15:47.04jayteehttp://excuseball.com/
15:47.06*** join/#asterisk Iamnach0 (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
15:47.15Kattyjaytee: have you heard from dave?
15:47.22jayteenope
15:47.38angryuserpawpro, you can strip it with opensips, but this is overkill
15:48.03Kattyhmm. i'll txt him
15:50.45*** join/#asterisk wiec (n=wiec@68-117-101-149.dhcp.eucl.wi.charter.com)
15:51.41*** join/#asterisk KavanS (n=KavanS@static-173-50-141-22.ptldor.fios.verizon.net)
15:51.55*** join/#asterisk IsUp (n=nocturne@unaffiliated/isup)
15:52.05jayteehttp://www.neatorama.com/2008/10/02/jurassic-brewery-scientists-brewed-beer-with-45-million-years-old-yeast/
15:52.21Katty:< it's not like dave to not answer a txt
15:52.33Guggei have an asterisk running on a non nattet network, and some phones one a natted network in another location, on the asterisk sip show channels gives me a lot of theese http://pastebin.com/m2d2a0605
15:52.40Kattyit's been a whole 5 minutes now :<
15:52.55Guggebut all the phones on the natted network is registered fine, and is working
15:53.12Guggewhat would make asterisk send option packets to the phones internal ip's ?
15:53.46wiecI've been wondering.  Why is there a separate sccp maintained elsewhere when skinny is built in?  is there some explanation somewhere on the web?
15:53.49ManxPower-workKatty: Rhodes Bake-n-serv Stone Ground whole wheat UPC 7002200715
15:54.09wiecI know sccp is unliked and unwelcome, but what's the deal here?
15:55.01p3nguinwiec: It's probably something like how there's a completely different PBX product called Asterisk where there's already Call Manager.
15:55.19ManxPower-workwiec: SCCP and Skinny are the same protocol
15:55.26pawproangryuser: ofcourse i can but I am not going to start having "exceptions" like this. SIP is a standard however loose this is ridiculous problem. And it seems asterisk.
15:55.31p3nguinSomeone else wanted to try their hand at writing a channel driver for it, so they did.
15:55.32*** part/#asterisk fixxxermet (n=lopan@vps.fixertec.net)
15:56.21*** join/#asterisk moa_ (n=moa_@lab.vision.net)
15:56.28wiecyeah, but sccp was designed for *.  why is it not integrated?  I can't help but suspect bad blood somewhere.
15:57.10[TK]D-FenderGugge: improper NAT setup
15:57.23[TK]D-Fenderwiec: PARDON?
15:57.27*** join/#asterisk Naikrovek (n=jjohnson@unaffiliated/naikrovek)
15:57.37ManxPower-workwiec: chances are the author was unable or unwilling to grant Digium an unrestricted license to the code.
15:57.44wiecnot big big deal, it happens, but why the different directions?  I am of course wondering which is better.
15:58.17p3nguinJust use what's built in unless you have problems with it or need something that one of the others provide.
15:58.49wiecI just expect that with all the time and effort that went into sccp would be integrated into the * project.
15:59.18p3nguinWhy?  There was already chan_skinny.
15:59.58wiecwell, since I can't get the sccp-b working, I guess I'll stick with skinny
15:59.59KattyManxPower-work: hmm. the website has several Rhodes products, mostly just Dinner Rolls tho.
16:00.08*** join/#asterisk TiToyz (n=TiToyz@aut75-5-82-239-181-57.fbx.proxad.net)
16:00.22ManxPower-workKatty: the bread dough is usually hidden on the bottom shelf of the dinner roll section
16:00.50ManxPower-workKatty: I have you the information copied from the product I pulled from my freezer.
16:01.58KattyManxPower-work: http://www.rhodesbread.com/img/products/812/0715product2_2_large.jpg <- is that it?
16:02.46ManxPower-workKatty: that's it.
16:03.18SuPrSLuGissue with polycoms registering. I can register any ata softphone and even a mitel, but the polycoms, configured through ftp pull their config files (verified through xfer logs) and show the correct user id on the lcd but won't register. Any ideas?
16:03.25*** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson)
16:03.25*** mode/#asterisk [+o putnopvut] by ChanServ
16:04.05p3nguinnaikrovek: Still have Google Wave invites burning a hole in your pocket?
16:04.42KattyManxPower-work: Stone ground whole wheat flour, water, high fructose corn syrup, yeast, vital wheat gluten, canola and/or soybean oil, salt, sodium stearoyl lactylate, yeast nutrients (calcium sulfate, ammonium chloride), ascorbic acid, enzyme (for improved baking)
16:05.02Kattywhy do they have to ruin a perfectly good whole wheat bread with bloody corn syrup
16:05.02SuPrSLuGwe're trying to go to a new server and put the correct options in the dhcp but damn phones won't register
16:05.09ManxPower-workKatty: the corn syrup will mostly be eaten by the yeast
16:05.11Kattygrumps.
16:05.44ManxPower-workSuPrSLuG: http://www.fnords.org/~eric/polycom-config-examples/
16:05.53ManxPower-worksee the dhcpd.conf file there
16:06.16KattyManxPower-work: the ingredient list looks a lot better than some i've seen. so there is that.
16:07.19wiecKing Corn!
16:07.37Naikrovekp3nguin: yeah i have some sure
16:07.38ManxPower-workKatty: it goes stale VERY fast, so I doubt there are many preservatives
16:07.39wiecYeast are smarter than that
16:07.41Naikrovekp3nguin: pm me your email
16:09.04Naikrovekhas 10 google wave invites in case anyone wants one
16:09.17Naikrovekgmail.com addresses seem to be getting priority
16:09.33p3nguinnaikrovek: Someone asked me for one and I told him I would see if I could find one for him.  Would that be okay to give some other random IRC person one of your invites as well?
16:09.49wiecThere is more corn syrup than yeast.  It's meant as a sweetener  - it makes you fat by shutting off your bodies "I'm full" response.
16:10.00Naikrovekp3nguin: no problem. i just need email addresses.  I don't care who I give them to, honestly.
16:10.20leifmadsengoogle wave has been disappointing for me
16:10.24p3nguinnaikrovek: Okay, one moment and I'll send both addresses.
16:10.28SuPrSLuGManxPower-work: thanks, we have that configured and working properly. I see the phones pull their configs. All looks good, except they don't register. They get the proper server, pull their mac.cfg,phone.cfg and sip.cfg files but will not register.
16:10.50Kattyjaytee: dave said he'd show up shortly.
16:11.41ManxPower-workSuPrSLuG: not registering or trying to register and failing
16:11.51SuPrSLuGcould it be a nat or firefall issue. the servers are on public ip's , so i wouldn't think that should cause an issue
16:11.55ManxPower-workSuPrSLuG: some version of the sip.cfg do not default to registering.  There's a config option for it.
16:12.28SuPrSLuGlet me look more into that, thnx
16:14.36*** join/#asterisk moy (n=moy@bas1-montreal42-1178046072.dsl.bell.ca)
16:20.02*** join/#asterisk benklop (n=ben@97-118-183-182.hlrn.qwest.net)
16:20.03*** join/#asterisk Ad-Hoc (n=nimbus@ppp1-228.adsl.forthnet.gr)
16:20.30*** join/#asterisk aidinb (n=Aidin@24-182-32-138.static.lnbh.ca.charter.com)
16:20.39benklopcan a dial pattern be set up to match sequences other than numbers?
16:21.14*** join/#asterisk clart001 (n=clart@host91-216-dynamic.11-87-r.retail.telecomitalia.it)
16:21.52benklopas in, route an <name>@gmail.com call through a gtalk-out route?
16:22.31benklopthe only info I have found details how to match number dial patters
16:22.44ManxPower-workbenklop: unfortunatly GUI Asterisks are not supported here.
16:23.02benklophow is that a gui feature?
16:23.24ManxPower-workbenklop: because non-GUI people don't call them "routes"
16:23.52ManxPower-workjust put exten => <name>@gmail.com,1,Dial(whatever) in your dialplan.
16:23.59benklopah.. well I started out in the gui., but have done a lot of config file editing. i by no means need to use the gui, although i have one installed
16:24.22clart001can anyone help me? http://pastebin.com/m610afbb8
16:24.30ManxPower-workRemember "extensions" don't have to be numeric
16:25.11benklopManxPower-work: i was hoping to make it generic enough to not actually need to give all of my gtalk buddies a separate extension
16:25.22ManxPower-workclart001: Macro(macroname,arg) not Macro(macroname(arg))
16:25.39clart001i try
16:26.34benklopManxPower-work: i take it that's not possible?
16:27.30ManxPower-workbenklop: only if you do a wildcard match at the end, not the beginning.
16:28.31benklopso anythnig that does not match anything else would go there
16:28.33*** join/#asterisk GameGamer43 (n=GameGame@h69-129-142-83.mdsnwi.tisp.static.tds.net)
16:28.35benklop?
16:29.06*** join/#asterisk luckyaba (n=lucky@ip72-194-215-55.sb.sd.cox.net)
16:31.32clart001@ManxPower-work: macro still not work
16:32.25*** join/#asterisk werdan7 (n=w7@freenode/staff/wikimedia.werdan7)
16:32.55[TK]D-Fenderclart001: And we dont' see you actual dialplan
16:33.00[TK]D-Fenderyour*
16:33.30clart001now i send you that
16:34.27*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
16:38.07clart001here http://pastebin.com/m36da6a71
16:39.50[TK]D-Fenderclart001: Looks like you passed the ARG fine and it did what it's supposed to
16:39.56[TK]D-Fenderclart001: What's the problem?
16:40.16[TK]D-Fenderclart001: Oh.. Lin 17 BTW... you have 2 priority #1 in there.
16:40.58clart001ok
16:42.01*** join/#asterisk CunningPike (n=CunningP@204.239.8.157)
16:42.29clart001well but i can't call
16:44.17*** join/#asterisk ReDNeQ (i=ReDNeQ@70.114.229.58)
16:45.29*** join/#asterisk saghul (n=saghul@ip51ccb640.speed.planet.nl)
16:45.34[TK]D-Fenderclart001: Well you aren't showing us anything
16:45.51[TK]D-Fenderclart001: Show us where it isn't processing what you think it should
16:46.36clart001ok wait a moment
16:48.12*** join/#asterisk ryduh (n=ryduh@204.16.143.186)
16:48.15*** join/#asterisk andres833 (n=andres83@190.144.75.22)
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16:49.54*** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net)
16:53.05ayesohow do i split a list of variables in the dialplan? ex key=val|key2=val2|key3=val3
16:53.19clart001here there is cli output when i try to call some number http://pastebin.com/m6a16900a
16:53.33*** join/#asterisk profxavier (n=chatzill@unaffiliated/neverblue)
16:54.18profxavierwhat is a free softphone, that can be deployed in a work-environment?
16:54.38p3nguinzoiper
16:54.39p3nguin~zoiper
16:54.41infobot[~zoiper] Zoiper (Formerly known as Idefisk) is a free SIP and IAX soft-phone for Windows, MacOSX, and Linux that can be found at http://www.zoiper.com
16:54.41[TK]D-Fender~x-lite
16:54.44[TK]D-Fender~ekiga
16:54.45infobot[~ekiga] Ekiga is an OSS SIP soft-phone for Windows, MacOSX, and Linux (Requires GTK+ which is included with the binary releases) that can be found at http://www.ekiga.org
16:54.46[TK]D-Fender~twinkle
16:54.47infobot[~twinkle] Twinkle is an OSS SIP phone for KDE/Qt systems and can be downloaded at http://www.twinklephone.com
16:54.51[TK]D-Fender~linphone
16:54.51infobotfrom memory, linphone is a SIP VOIP phone.  To configure it to use sip.handhelds.org, ask ibot about linphone config . not working with fwd.pulver.com
16:55.03p3nguinplenty of softphones.
16:55.08[TK]D-Fenderprofxavier: Or any of several dozen others
16:55.21*** join/#asterisk b14ck (n=comradeb@72.37.252.50)
16:55.24b14cksup everyone
16:55.36profxavierx-lite isnt a good solution
16:55.47p3nguinI agree.  It's junk.
16:55.48profxavierekiga, as I remember, a bit ugly
16:56.03[TK]D-Fenderprofxavier: Well you haven't layed out any criteria, and by many peoples opinions, ALL soft-phones suck
16:56.05*** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com)
16:56.28profxaviertwinkle, is that for Windows ?
16:56.36b14cktwinkle is for linux (kde)
16:56.40b14ckit uses t
16:56.41b14ck*qt
16:56.57b14cktwinkle is a bit better than ekiga imo
16:57.15profxavierok, windows only based softphones
16:57.18[TK]D-Fenderprofxavier: Oh NOW you're bringing up OS?
16:57.30profxavieri see zoiper is also linux based
16:57.32Kattyhmm.
16:57.40Kattydoes anyone know a Matt Riddell?
16:57.51kaldemarprofxavier: there's also win and os x versions of zoiper
16:57.54p3nguinb14ck: KDE is not Linux, and you don't need KDE to run twinkle.
16:57.57QwellKatty: ZX81
16:57.58[TK]D-Fenderprofxavier: "linux based"?
16:58.30b14ckp3nguin, I'm aware--I mentioned that because it is qt based and qt makes me think of kde. I think that is pretty self-explanatory D:
16:58.32KattyQwell: uhh?
16:58.38KattyQwell: i'm not sure i understand your reply.
16:58.39QwellKatty: is Matt Riddell
16:58.47Kattyoh
16:58.49outtoluncnick = name
16:59.01Kattyinfobot: seen ZX81?
16:59.04infobotzx81 <n=ZX81@121-74-242-218.telstraclear.net> was last seen on IRC in channel #asterisk, 34d 20h 29m 11s ago, saying: ':) sweet'.
16:59.11cuscowe use twinkle. its cool because you can call its functions from command line
16:59.14Kattyhmm. 34days
16:59.21cuscowhere ca I download latest asterisk-addons?
16:59.24Kattyi don't seem to recall ever talking to a ZX81 :<
16:59.26TSMgrrr, inbound faxing is not working on * box, i can fax betwenn machines in the office but from the PSTN it does not work, any ideas
16:59.30cuscoits not in the asterisk.org/downloads anymore
16:59.42[TK]D-FenderKatty: He helps clean up the WIKI now and again....
16:59.45kaldemarcusco: http://downloads.asterisk.org/pub/telephony/asterisk/
16:59.53QwellKatty: why do you ask?
17:00.03Kattynickserv reports he was around 12 hours ago.
17:00.07KattyQwell: he's trying to follow me on twitter.
17:00.11Qwelloh
17:00.20cuscokaldemar: what is the latest stable?
17:00.20Qwelllink?:p
17:00.22Kattyi don't let random folk follow me. it's creepy.
17:00.47cuscois 1.6.2 to be released soon?
17:00.57Kattywhich is why i'm trying to figure out how he knows me, and why he wants to follow me.
17:01.09Qwellcusco: when it's ready
17:01.12[TK]D-FenderKatty: And people posting their every inane though in 140 byte chunks isn't "creepy" by definition?
17:01.24[TK]D-Fendercusco: Next spring... SHARP!
17:01.28profxavierok, looks like Zoiper also needs an upgrade to the paid application to account export
17:01.30cusco...
17:01.41kaldemarcusco: the one with the biggest version number and without rc for the branch you use.
17:01.45Katty[TK]D-Fender: i enjoy seeing what other people are up to.
17:01.55[TK]D-FenderKatty: I'd rather not know...
17:01.56Katty[TK]D-Fender: tho i don't really tweet much myself.
17:02.08cuscoI mean to know if by using asterisk-addons-1.6.2.0-rc2 major problems are fixed, or is it unstable
17:02.18Qwellcusco: read the changelog
17:02.21*** join/#asterisk raden_work (n=jon@69-179-99-17.stat.centurytel.net)
17:02.28[TK]D-Fendercusco: the entire branch is considered "unstable"
17:02.36[TK]D-Fendercusco: it hasn't his a full release yet
17:02.47profxavierso it seems everyone is doing that, for mass deployment and handling accounts, you need to get the paid application
17:02.53[TK]D-Fendercusco: 1.6.2.0 Rc != public release
17:03.25profxavieri was hoping to deploy a softphone, within a Windows/Mac environment, that could be used to both deploy and account manage
17:03.26[TK]D-Fenderprofxavier: Ekiga <-
17:03.34profxavierbut free :D
17:03.38[TK]D-Fenderprofxavier: Ekiga <-
17:03.53clart001@[TK]D-Fender can you help me?
17:04.33profxavierI have a feeling Ekiga will be disliked, similar to Pidgin
17:04.37[TK]D-Fenderclart001:     -- Executing [9077*******_22@intern-iax:3] Dial("IAX2/clart-4081", "/077*******_22|20|Tt") in new stack
17:05.00[TK]D-Fenderclart001: You are passing a bad value to this dial and you have not shown your updated and OCMPLEtE macro to examine
17:05.09[TK]D-Fenderprofxavier: Feel free to write your own.
17:05.26[TK]D-Fenderprofxavier: And I don't know anyone who likes ANY softphone really
17:05.43profxavierwell, X-Lite has been useful
17:05.56profxavierand most ppl do not complain abou t it, directly
17:06.29ManxPower-workIt seems to me the only people that like softphones are the ones that have not used hardphones
17:06.45clart001@[TK]D-Fender Do you want to see my dial plan?
17:06.47profxavierwe have a mix in the office
17:06.58[TK]D-Fenderclart001: Don't make me ask a third time
17:07.03profxavieri have more complaints about the hard phones/service, than the softphones
17:07.10ManxPower-workI've never had to reinstall my hardphone when my windows blows up.
17:07.17profxavier:)
17:07.17cuscook looks like Im not to use latest rc - res_config_mysql.c:1367: error: unknown field ‘update2_func’ specified in initializer
17:07.27[TK]D-Fenderprofxavier: What phones?
17:07.38profxavierexactly ;)
17:07.43TheDavidFactorManxPower-work, depends on how much of the desk windows takes with it ;-)
17:07.49ManxPower-work[TK]D-Fender: My psychic vision says he's using Grandstream.
17:07.50profxavieri do have some Polycoms sitting on the shelf
17:08.04[TK]D-Fenderprofxavier: And what is sitting on the desk?
17:08.15profxavieri just dont want to give those out to just anyone :)
17:08.22Qwellread: GS
17:08.34ManxPower-work~phones
17:08.35infobotWhile personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else.  Do not consider Grandstream phones.  Ever.
17:09.29[TK]D-FenderQwell: Excellent translation skillz (wit-a-"Z" yo!)
17:09.30profxavierlmao
17:09.35profxavierit wasnt my consideration
17:09.40QwellO.o
17:09.44profxavierit was my predecessors
17:09.55clart001@[TK]D-Fender http://pastebin.com/m32887592
17:10.24QwellI wish I could blame stuff on people before me, so I didn't have to fix it.
17:10.40[TK]D-Fenderclart001:   -- Executing [s@macro-choose-provider:1] Set("IAX2/clart-4081", "GLOBAL(provider)=") in new stack
17:10.57[TK]D-Fenderclart001:     -- Executing [9077*******_22@intern-iax:2] NoOp("IAX2/clart-4081", "provider is: ") in new stack
17:11.09[TK]D-Fenderclart001:     -- Executing [9077*******_22@intern-iax:3] Dial("IAX2/clart-4081", "/077*******_22|20|Tt") in new stack
17:11.22[TK]D-Fenderclart001: Your ${PROVIDER} is blank
17:11.33kaldemarclart001: you might want to change whole macro-choose-provider to something else
17:12.13[TK]D-Fenderkaldemar: Yes, you can see he's disabled any real logic
17:12.55*** join/#asterisk eppigy (n=Dave@snugglenets.com)
17:13.01eppigyhello
17:13.07[TK]D-Fendereppigy: YOU ARE DAVE!!
17:15.04*** join/#asterisk fofware (n=chatzill@190.7.25.160)
17:17.26eppigyFACT
17:18.10jayteeTRABAJO!
17:18.16*** join/#asterisk baijum (n=baiju@122.167.107.238)
17:19.09profxaviercan I get some help with my Digitmap for my Polycom 330 ?
17:19.28profxavierhere is what I have: "[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT"
17:19.39b14ckprofxavier, you should change it to:
17:19.42b14ck"xxx"
17:19.50profxavierwhen I dial a number, 9[areacode]7_digits
17:19.51b14ckand delete all other settings in sip.cfg
17:20.02profxavierand the headsest is down, Dial works
17:20.11clart001@kaldemar how?
17:20.30[TK]D-Fenderprofxavier: Nothing in there supports 8 digit patterns
17:20.34profxavierbut, when the headset is off the receiver, I cannot dial the same numer
17:20.37*** join/#asterisk cesar_CR (n=cesar@201.192.86.30)
17:20.44profxaviernumber*
17:21.02[TK]D-Fenderprofxavier: on-hook you can dial outside your digitmap.  Off-hook you can't
17:21.11[TK]D-Fenderprofxavier: So add a proper pattern
17:21.16clart001@ [TK]D-Fender you see my macro in http://pastebin.com/m32887592
17:21.19profxaviercorrect
17:21.24profxavierso does the order matter
17:21.44[TK]D-Fenderprofxavier: Forget order, nothing you have in there says "8 digits" <-
17:21.45clart001@[TK]D-Fender i can't see where is the error
17:21.56[TK]D-Fenderoops, scratch that
17:22.01profxavierwhy would I need something with 8 digits?
17:22.04profxavierah
17:22.52*** join/#asterisk dmz (n=dmz@64.203.207.101.dyn-cm-pool-54.hargray.net)
17:23.13*** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net)
17:23.23[TK]D-Fenderprofxavier: [2-9]xxxxxxxxx <- this is the clsoest you ahve and still doesn't match
17:23.46[TK]D-Fenderprofxavier: 9 X's
17:23.55profxavierso why does the digitmap perform differently when the receiver is up/down ?
17:23.58[TK]D-FenderprogYou need "starts with 9 and has 10 "X"s
17:24.06[TK]D-Fenderprofxavier: Just the way it works
17:24.19[TK]D-Fenderprofxavier: it IGNORES the digitmap while on-hook
17:24.24profxavierits the way its parsed
17:24.26[TK]D-Fenderprofxavier: Just live with it already....
17:24.29profxavierah, makes sense
17:24.33[TK]D-Fenderprofxavier: No, it isn't
17:24.57[TK]D-Fenderprofxavier:  you don't have a 9 + 10 digit patter in here
17:25.06[TK]D-Fender[12:23]<[TK]D-Fender>profxavier: [2-9]xxxxxxxxx <- this is the clsoest you ahve and still doesn't match
17:25.10ManxPower-workWhat part of "it IGNORES the digitmap while on-hook" do you not understand?
17:25.18[TK]D-Fender[12:23]<[TK]D-Fender>profxavier: 9 X's
17:31.06clart001@ManxPower-work nothing?
17:31.14*** join/#asterisk bmoraca (n=chatzill@66.242.174.254) [NETSPLIT VICTIM]
17:31.14*** join/#asterisk angryuser (n=angryuse@LPuteaux-151-42-27-99.w193-251.abo.wanadoo.fr) [NETSPLIT VICTIM]
17:31.14*** join/#asterisk Chinorro (n=Chino@91.117.226.19) [NETSPLIT VICTIM]
17:31.18clart001@[TK]D-Fender nothing?
17:31.37[TK]D-Fenderclart001: I showed tyou the lines where that value is set, and it ISN'T GETTING SET
17:31.58[TK]D-Fenderclart001: it is BLANK and because of that you aren't apssing a valid tech to dial in your DIAL() command
17:32.20[TK]D-Fenderpassing*
17:32.56profxavierok added [2-9]xxxxxxxxxx
17:33.01profxavierstill doesn't work
17:33.17clart001i saw your previuos answer but it still doesn't work
17:33.18profxavierthat would need to appear before the [2-9]xxxxxxxxx
17:33.18[TK]D-Fenderprofxavier: shouldn't be 2-9
17:33.45[TK]D-FenderclanAnd you aren't showing us your updates.  Do you think we are psychic?
17:33.50[TK]D-Fenderclart001: And you aren't showing us your updates.  Do you think we are psychic?
17:34.12[TK]D-Fenderprofxavier: I also don't see your complete line
17:34.13profxavierso just 9xxxxxxxxxx ?
17:34.30[TK]D-Fenderprofxavier: Certainly looks better
17:34.52profxavierill give it a try, but the order within the Digitplan does matter, correct ?
17:35.02[TK]D-Fenderprofxavier: Not as far as I'm aware...
17:36.00profxavierso if 9xxxxxxxxx appears before 9xxxxxxxxxx, then 9+10 digits would dial correctly ?
17:36.43[TK]D-Fenderprofxavier: Why would you be leaving illegal patterns in there>
17:37.08[TK]D-Fenderprofxavier: You seem to be trying to do as little as possible and leaving bits behind that you shouldn't
17:37.26*** join/#asterisk kfife (n=Miranda@kfife.com)
17:37.35profxaviertrue
17:38.01ManxPower-workprofxavier: Do you know what a lazy PBX administrator is called?  "Unemployed".
17:38.25kfifemust I shut down BOTH asterisk AND dahdi to update libpri?
17:38.58[TK]D-Fenderkfife: You can update them and the restart * afterwards
17:39.05*** join/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
17:39.41jblackKatty: Ping
17:39.50profxavierManxPower-work: is this not an assistance channel ?
17:39.52kfife[TK]D-Fender: don't need to bounce dahdi, just Asterisk?
17:40.06jblackCan someone list 5 large companies that use *
17:40.07profxavierbecause, please, correct me if I am wrong
17:40.09[TK]D-Fenderkfife: You may have to recomplie * as well.
17:41.05kfife[TK]D-Fender: That's a surprise.  I wouldn't have imagined that
17:41.43[TK]D-Fenderkfife: chan_dahdi it tied to DAHDI you know...
17:41.59kfife[TK]D-Fender: good point.
17:42.06[TK]D-Fenderkfife: If DAHDI changes in any way relating to the channel driver interface well... duh
17:43.36*** join/#asterisk AlHafoudh (i=c32e4504@gateway/web/freenode/x-ifshmczssvxtqzic)
17:44.03AlHafoudhhi all
17:44.32AlHafoudhi cannot get asterisk sending correct congestion signal to avaya, can someone help please?
17:45.01ManxPower-workprofxavier: you have been helped, you're just not listening
17:45.15[TK]D-FenderAlHafoudh: Show us your sip configs & debug in a pastebin and maybe we can
17:45.16[TK]D-Fender~pb
17:45.17infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
17:45.26jblackManxPower-work: Digium should have a "who uses asterisk" page as a selling point
17:45.30[TK]D-FenderManxPower-work: No, he's gotten that already....
17:45.48ManxPower-workjblack: I think they do.
17:46.06ManxPower-workjblack: if not you should mention that to someone that works for Digium
17:46.13jblackOh, I thought you did. Sorry.
17:46.15AlHafoudh[TK]D-Fender: i use trixbox, so the extensions configuration files are big, i can am using h323 for connection to avaya, so i will paste that one and sip
17:46.18ManxPower-work~manxpower
17:46.19infobotManxPower has been using Asterisk in production since late 2001.  Currently works at InterGlobe Communications, a CLEC based in NYC with service in NY, NJ, FL, and TX.  http://www.nyigc.com
17:46.29Qwelljblack: also, it would help if it was somebody who can do something about it
17:46.49[TK]D-FenderAlHafoudh: No need for SIP then...
17:46.50jblackFinding someone that can do anything at company is a trick these days.
17:47.08jblackI'm just looking for a short list of recognizable names of asterisk users.
17:47.12AlHafoudh[TK]D-Fender: http://pastebin.com/m48bd097f
17:47.17b14ckmeh
17:47.23b14ckIt is hard to find good people for tech jobs
17:47.24b14ck:(
17:47.49*** join/#asterisk circut (n=circut@c-71-57-110-244.hsd1.il.comcast.net)
17:47.50[TK]D-FenderAlHafoudh: Who is supposed to register to who?
17:48.25circuthey all
17:48.34[TK]D-FenderAlHafoudh: because * is not being told to register.  Also you have a FiXED HOST specified for that peer entry.  if its fixed, then * will not allow a device to REGISTer to it
17:48.36circuthaving some issues setting the callerid on some POTS lines
17:48.41Qwellb14ck: No it's not.
17:48.44AlHafoudh[TK]D-Fender: Asterisk is used as SIP gateway for avaya, asterisk is connected to SIP ISP, and also connected to Avaya by H323
17:48.58[TK]D-Fendercircut: POTS doesn't allow you to se CID except in extremely few cases
17:49.03circutwow
17:49.11Qwellb14ck: Be a company people actually want to work for, and...it's very easy.
17:49.16[TK]D-Fendercircut: POTS = dumb
17:49.18circutat&t have their heads fully buried them
17:49.19circutthen*
17:49.22circutawesome
17:49.30circutthats what i figured
17:49.32AlHafoudh[TK]D-Fender: it work, we just need to get corrent hangup codes to avaya, because we need to failover the sip connection to other asterisk with different sip provider
17:49.39[TK]D-Fendercircut: No... they may offer this service, its just EXtrememlY rare
17:50.00[TK]D-FenderAlHafoudh: So you want a dialplan failover for multiple carriers?
17:50.09circutbut there is nothing special i would need to do within asterisk right?
17:50.19AlHafoudh[TK]D-Fender: the failover must be done one Avaya side :( (reporting)
17:50.20circutjust callerid=asreceived in dahdi-channels.conf
17:50.42[TK]D-Fendercircut: the signalling for this would be DTMF in your dial if its even supported... but nothing to "configure" so much as digits + "w" for time delays in between, etc
17:51.02[TK]D-FenderAlHafoudh: You aren't being very clear about what you need to change in yours server side
17:51.14[TK]D-Fendercircut: thats INBOUND callerid
17:51.29AlHafoudh[TK]D-Fender: there are 3 asterisk with 3 providers connected to one Avaya system, Avaya needs to detect outage of the far SIP connection and switch to the next available Asterisk
17:52.14[TK]D-FenderAlHafoudh: What protocol is * supposed to do to inform the Avaya?  there is no such signalling standard AFAIK
17:53.06AlHafoudh[TK]D-Fender: asterisk is connected with avaya using h323, well the congestion should send the proper signal through avaya, but on the avaya side we see "call rejected" in trace
17:53.31rpmAvaya does h.323 signaling. Sending a keepalive would work
17:54.05[TK]D-FenderAlHafoudh: Show su a complete call trace from *'s side
17:54.07[TK]D-Fender~pb
17:54.08infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
17:54.10[TK]D-Fender^^^^^^^^^
17:54.13[TK]D-FenderAlHafoudh: PASTEBIN it
17:54.34AlHafoudhrpm: the connection between asterisk and avaya is not a problem, the ahead sip connection is
17:55.25*** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk)
17:55.41AlHafoudh[TK]D-Fender: http://pastebin.com/m243e677f
17:55.45donnibhow can i enable debug from chan_sip ?
17:55.56AlHafoudhdonnib: sip debug
17:56.06donnibhmm
17:56.14donnibnot getting any :(
17:56.28donnibi tried setting the core set debug level as well
17:56.31donnibbut still
17:56.44AlHafoudhtry asterisk -rvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
17:57.16*** join/#asterisk bmoraca (n=chatzill@66.242.174.254)
17:57.34bmoracahas anyone here used a Cisco AS5400 with asterisk?
17:58.43kaldemardonnib: sip set debug on
17:59.05donnibkaldemar: did that already
17:59.35donnibi am getting the regular debug i mean the sip messages but no log from chan_sip
17:59.50*** join/#asterisk Erol_ (i=erol__@88.234.110.3)
17:59.51Erol_hi
17:59.52ManxPower-workdonnib: you are not making ANY sense.
18:00.01AlHafoudh[TK]D-Fender: any luck?\
18:00.13ManxPower-workif you are seeing the sip messages then your sip debug is enabled.
18:00.45donnibManxPower-work: well i was expecting to see the internal parsing problems in chan_sip
18:00.53Erol_When I have a phone call on an anlog line and after the conversation end the line gets stuck if the guy on the PBX side doesnt hang up first.
18:00.55Erol_why is that?
18:01.02donnibsomething like this VERBOSE[32543]:   == Parsing '/etc/asterisk/sip.conf':
18:01.14ManxPower-workdonnib: that's not sip debug
18:01.20ManxPower-work"sip set debug on"
18:01.30ManxPower-workdepending on your Asterisk version.
18:01.52donnibi am running 1.4.22
18:02.23donnibwhat do you call the kind of debug i want ?
18:02.32ManxPower-work"sip set debug on"
18:02.45ManxPower-workor you can just do "help"
18:02.52donnibi do sip set debug
18:02.58donniband that enables the debug
18:03.02[TK]D-FenderAlHafoudh: * is playing back audio instead of passing back proper status
18:03.10donnibbut still no debug as mentioned before
18:03.21donnibonly the sip messages
18:03.22[TK]D-FenderAlHafoudh: And that is going to be a dialplan issue because of your GUI.  If you want real control you're going to have to do your own configs
18:03.23ManxPower-workdonnib: then you are not receiving any sip messages
18:03.41donnibi do receive the sip messages, that works fine but i want more log
18:03.52donnibto see why i get 401 unauthorized in a sip message
18:03.59[TK]D-Fenderdonnib: What "more"?  Who says that something more exists?
18:04.11kaldemardonnib: that and core debug is all you're going to get.
18:04.20donnibi was hoping there was more that might give more info that´s all
18:04.26donnibok well then i was expecting to much
18:04.34donnibthx for clarifying that out
18:04.46[TK]D-FenderErol_:  Also your telco for this service :
18:04.47[TK]D-Fender~cds
18:04.48infobot[~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up.  This is typically done either by a momentary battery cut, or by a polarity reversal on the line.
18:04.50[TK]D-Fender^^^
18:05.03Erol_[TK]D-Fender: I didnt understand?
18:05.12[TK]D-Fender[13:04]<infobot>[~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line
18:05.27*** part/#asterisk clart001 (n=clart@host91-216-dynamic.11-87-r.retail.telecomitalia.it)
18:05.30[TK]D-FenderErol_: * can't tell the line has been hung up on.
18:05.39Erol_[TK]D-Fender: do you know how to fix it?
18:06.14[TK]D-FenderErol_: .... perhaps you have a reading issue
18:06.25Erol_[TK]D-Fender: is this echo cancellation to fix this?
18:06.30[TK]D-Fendereorask your telco to enable CDS on your line <------------------
18:06.39[TK]D-FenderErol_: ask your telco to enable CDS on your line <------------------
18:06.43[TK]D-FenderErol_: ask your telco to enable CDS on your line <------------------
18:06.46[TK]D-FenderErol_: Am I clear?
18:06.51Erol_[TK]D-Fender: uhm, yeah but
18:06.56[TK]D-FenderErol_: NO BUTS
18:07.03Erol_[TK]D-Fender: so you say that it cant be fixed
18:07.05[TK]D-FenderErol_: No, echo cancellation has NOTHING to do with this
18:07.17Erol_asking telecom, =)
18:07.19[TK]D-FenderErol_: I'm saying it can and the telco has to adjust their signalling
18:07.26Erol_lol
18:07.41Erol_erm, you cant even ask telco here, =)
18:07.52[TK]D-FenderErol_: Life sucks but rarely swallows...
18:07.55ManxPower-workErol_: then you'll never solve your problem
18:08.11Erol_actualy I found a solution to this
18:08.12Erol_a hardware
18:08.15elliot98I have an old SIP ata
18:08.23Erol_it fixes that
18:08.31Erol_but its expensive
18:08.37Erol_8 ports $600
18:08.41kfife[TK]D-Fender: updated libpri, bounced *, pri show version: 1.4.10.2.   How can I determine if any other recompies are necesary besides 'wait and see' on a production system?
18:08.41elliot98wondering is there is any possible way to download asterisk firmware onto it
18:09.07[TK]D-Fenderkfife: Do them all and "restart  when convenient" on your server
18:09.23*** join/#asterisk ajohnson (n=aaron@65-122-4-130.dia.static.qwest.net)
18:09.23Erol_[TK]D-Fender: what does CDS stand for?
18:09.30[TK]D-Fenderelliot98: Asterisk isn't "ATA firmware"
18:09.46[TK]D-FenderErol_: [13:05]<[TK]D-Fender>[13:04] <infobot> [~cds] Call Discconect Supervision is a service placed on analog lines to be able to signal you that that the calling party has hung up. This is typically done either by a momentary battery cut, or by a polarity reversal on the line
18:09.53[TK]D-FenderErol_: read the line dammit
18:10.02Erol_[TK]D-Fender: thanks
18:10.06elliot98I could see if a version of Linux works on it and then install asterisk on top of that
18:10.16Erol_[TK]D-Fender: and cong. to you that you made me understand, =)
18:10.20kfifeThat won't bounce dahdi.  Somebody has to actually know the answer.  I can speculate as well.
18:10.23elliot98but I would need to develop drivers for the FXO/FXS ports on the ata
18:10.28p3nguinelliot98: Don't get your hopes up on that.
18:10.50[TK]D-Fenderkfife: Not the kernel modules themselvs... that you'll have to do yourself
18:11.06ManxPower-workelliot98: come back when you have Asterisk running on that ATA
18:11.10[TK]D-Fenderelliot98: Have you considered building a new box from the atom up?
18:11.22[TK]D-Fenderelliot98: Might be easier
18:11.26elliot98it is theoretically possible, although, yes it is a huge undertaking
18:11.36[TK]D-Fenderelliot98: I'll even hold a few iron atoms in place for you to help you get started
18:11.37elliot98easier...but alas, probably more costly
18:11.53elliot98oh, thank you..is your DCC on?
18:12.10[TK]D-Fenderelliot98: You are trying to turn an automatic transmission into an entire car... good luck with that and let us know when you succeed
18:12.25p3nguinI doubt it's even possible to do it, regardless of how much cash you have.
18:12.33[TK]D-Fender~savemoney
18:12.34infobot<Gremlin> It's okay, but I was hoping to save some money with a soldering iron and a bunch of sound cards.
18:12.37[TK]D-Fenderahhhhhh
18:12.42[TK]D-FenderStill doesnt' get old :)
18:13.05[TK]D-Fendercarrar: cheers!
18:13.21elliot98a firmware version of Asterisk does exist
18:13.25carrarhi!!
18:13.45carrarCheerio
18:13.49[TK]D-Fenderelliot98: What part of "isn't an ATA firmware" don't you get?
18:13.51elliot98hello!
18:14.16elliot98asterisk is not ATA firmware...but there may be a Linux distro that it
18:14.17elliot98*is
18:14.43[TK]D-Fenderelliot98: And what linux distro do you think you can drop into that ATA... whose model you didn't even bother to share with us
18:14.44ManxPower-workelliot98: Come back AFTER you have Asterisk running on the ATA
18:14.52carrarthrows away all his old Asterisk ATA's
18:14.55ManxPower-workUntil then all discussion is useless
18:15.26[TK]D-FenderManxPower-work: No... I'm pretty sure it'll be useless after then to, because by the time he does, it'll probably spontaneously combust :)
18:16.16ManxPower-work[TK]D-Fender: I'm counting on that.
18:16.25elliot98I really don't see why my idea is so far fetched...routers are running Linux all the time now
18:16.36ManxPower-workHe'll never get Asterisk running on an ATA.
18:16.47elliot98any bets?
18:17.02[TK]D-Fenderelliot98: .... what model?
18:17.05ManxPower-workelliot98: come back after you have Asterisk running on your ATA.  Until then all discussion is useless.
18:17.36elliot98I have few old ones lying around
18:17.39[TK]D-Fenderelliot98: .... what model?
18:17.51*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
18:18.05elliot98hold on...
18:19.05elliot98here's one: http://www.soundwin.com/content/product/product01_02.aspx?sid=28&pid=4
18:20.03TSMif i have faxdetect turned on in my chan_dahdi will that always make my incomming faxes be picked up by asterisk and run in the exten => fax,1,etc.... code?
18:20.11[TK]D-Fenderelliot98: Where do you see linux firmware you can install for it?
18:20.24*** join/#asterisk DarkRift (n=dark@modemcable015.68-200-24.mc.videotron.ca)
18:20.28ManxPower-workTSM: and your outgoing faxes too
18:20.35[TK]D-FenderTSM: * will listen for it during any ivr or background process normally
18:20.42Kattyinfobot: fax
18:20.43infobotrumour has it, fax is The honor of designing the first fax *service* in actual use goes to Giovanni Caselli, an Italian abbot, born in Siena in 1815, who turned his hand to science and was, by 1849, editing a scientific magazine. In 1856 he claimed that he  had developed a device, which he called a "pantelegraph," that could send facsimiles of images and text.  Napoleon III did not come up with the idea, he merely backed it.
18:20.51Kattyinfobot: faxing?
18:20.52infoboti heard faxing is 8% knowledge, 5% skill, 11% luck, and 76% voodoo
18:21.01Katty:<
18:21.37[TK]D-FenderComedy Gold
18:21.54TSMTK: so if i dont want asterisk to pick it up and just pass it straight though to the extention i wanted, should i just disable faxdetect, problem im having at the moment is inbound faxes are making the machine ring and pick up and start the tones, but then no fax comes in
18:22.10[TK]D-FenderTSM: Show me
18:22.30TSM[TK] = show you which bits, chan_dahdi?
18:22.37[TK]D-Fendertxmthe call
18:22.59TSMok ive got to call my US office and get them to send a fax
18:26.54*** join/#asterisk Malkor (n=marco@hlle-d9ba009b.pool.mediaWays.net)
18:27.09elliot98what kind of progress is going on with voip faxing these days?
18:28.08TSMTK: http://pastebin.com/m41d1ae47
18:28.12[TK]D-Fenderelliot98: T.37 & T.38 are the same as always
18:28.20benklopare there any potential issues with setting rtpstart/end to some low value, like 1024:5096?
18:28.37*** join/#asterisk bmg505 (n=leon@196-209-77-55-rndf-esr-5.dynamic.isadsl.co.za)
18:28.42ManxPower-workTSM: faxdetect isn't going to work until you Answer the line
18:29.00[TK]D-FenderTSM: As ManxPower-work said.
18:29.11Kattyinfobot: faxdetect?
18:29.21Kattyfrowns
18:29.22*** join/#asterisk donnib (n=donnib@0x555281d0.adsl.cybercity.dk)
18:29.42ManxPower-workbenklop: you need 2 RTP ports per active call.
18:29.47TSMManxPower: i dont want it to do anything as we have direct number for faxmachine and its set as DID->DAHDI32
18:30.03TSMTK ManxPower: could it be because i left EC on?
18:30.06ManxPower-workTSM: then why the hell are you trying to do faxdetect?
18:30.19elliot98I am new with T.38...is it part of the fasciile machine or is part of a T.38 compatible Voip PBX/ATA?
18:30.19ManxPower-workTSM: the fax tone disables EC per ITU spec
18:30.19profxaviercrap, this still isnt working
18:30.23[TK]D-FenderTSM: Um... pardon?  What is the problem here?
18:30.27benklopManxPower-work: ok, but the specific range shouldn't be an issue, as long as I have enough..
18:30.32TSMManxPower: that was the default setup that the sangoma card did
18:31.06[TK]D-FenderTSM: No, DAHDI controls the EC
18:31.09benklopapparently gtalk is ignoring my port requests and using very low ports anyway, even though I request something in my range
18:31.24TSMTK: client faxes into the office, fax machine picks up but never receives the fax, i can internal fax though betwenn two machines fine
18:31.25[TK]D-FenderTSM: and EC gets disabled when fax/modem tones are deteceted
18:31.27beekGood day all
18:31.39beekinfobot tell me about thebook
18:32.17*** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net)
18:32.26TSM~book
18:32.27infobot[~book] Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://astbook.asteriskdocs.org or see ~buybook
18:32.51Kattyhttp://www.asterisk.org/docs/asterisk/trunk/applications/receivefax <- this
18:33.06*** join/#asterisk leifmadsen (n=Leif@asterisk/documenteur-extraordinaire/blitzrage)
18:33.06*** mode/#asterisk [+o leifmadsen] by ChanServ
18:33.08profxaviercan someone assist me with settingup my Digitplan for a Polycom 330 ?
18:33.08Katty^- my core show application ReceiveFAX does not ...exist..apparently.
18:33.14Kattyinfobot: RecieveFAX?
18:33.24*** join/#asterisk beek (n=klinebl@pdpc/supporter/professional/beek)
18:33.24[TK]D-FenderKatty: "rxfax()"
18:33.52[TK]D-Fenderprofxavier: You going to actually show us this time?
18:33.55Kattyi'm asuming there's additional stuff for rxfax application
18:34.03[TK]D-Fender~assume
18:34.03infobotAbout assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight&quote=assume.txt&file=assume.wav  It makes an (ass) out of (u) and (me)
18:34.03Kattybecause i don't appear to have that application either
18:34.04profxavierFender show ?
18:34.08ManxPower-workKatty: you need libspandsp installed for rxfax/txfax/fax
18:34.17Kattyohah.
18:34.19Kattyk
18:34.30elliot98do real T.38 ATAs exists?
18:34.32[TK]D-FenderKatty: the old apps are 3rd party. t he new ones are in 1.6 + if you had Spandisp libs in place when you build *
18:34.46[TK]D-Fenderelliot98: Yes
18:35.03Katty[TK]D-Fender: well i'm still running on 1.4 at the moment, can't upgrade until isymphony officially supports 1.6
18:35.17[TK]D-Fenderkaty3rd party it is...
18:35.20KattyManxPower-work: i don't suppose you can point me to where i can download that.
18:35.40profxavier<PROTECTED>
18:35.41ManxPower-workKatty: softswitch something.  Google should hav the site on the first page
18:35.45Kattyk
18:35.55ManxPower-workvoip-info will also have the info
18:36.01profxavierand when the receiver is off, I cannot dial '9'
18:36.11elliot98but when an ata says it is T.38, that just probably means it carries through T.38, but not that it is an actual T.38 gateway
18:36.11ManxPower-workKatty: btw spandsp was written by copiece, one of the best DSP people I know of
18:36.13profxaviersorry, '9' + 10 digits
18:36.35[TK]D-Fenderprofxavier: Where is the cleaned up version two us us told you to do?
18:36.56profxavierim falling back to what I had originally
18:37.02KattyManxPower-work: you mean coppice?
18:37.13ManxPower-workyeah, him
18:37.18profxavieri was unsure what changes I needed. aka: i didn't get my answer
18:37.47[TK]D-Fender<elliot98>but when an ata says it is T.38, that just probably means it carries through T.38, but not that it is an actual T.38 gateway <- ridiculous
18:37.59*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
18:38.09[TK]D-Fenderelliot98: It has SIP on one side, POTS on the other.  ther is nothing to pass-through
18:38.37[TK]D-Fenderprofxavier: I confirmed your earlier pattern.  So what do you mean "didn't get your answer"?
18:38.55ManxPower-workThe problem with T.38 is most of the time it only works with products from the same company (interop sucks)
18:38.59profxavierwhich pattern ?
18:39.03[TK]D-Fenderprofxavier: YES, your other pattern was fine.  Now go DO IT.  you even said you DID.  So why do we now see that you didn't?
18:39.06elliot98I see...but when Asterisk only has pass-through suppor
18:39.26ManxPower-workelliot98: correct.  And even that has interop issues with other vendors
18:39.44[TK]D-Fender[12:34]<profxavier>so just 9xxxxxxxxxx ?
18:39.46[TK]D-Fender[12:34]<[TK]D-Fender>profxavier: Certainly looks better
18:39.53elliot98so it would be best to have a stand-alone fax gateway for accepting faxes
18:40.04[TK]D-Fender[13:39]<elliot98>I see...but when Asterisk only has pass-through suppor <- O RLY?  And where is this from?
18:40.39ManxPower-work[TK]D-Fender: 1.4 I bet
18:40.50[TK]D-FenderManxPower-work: Do not feed <-
18:40.56[TK]D-FenderManxPower-work: I'm handing out rope.
18:41.05[TK]D-FenderManxPower-work: I have LOTS
18:41.38elliot98see here
18:41.43elliot98http://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38
18:42.01[TK]D-Fenderelliot98: WIKI is random value crap
18:42.10[TK]D-Fenderelliot98: that is not official documentation
18:42.30[TK]D-Fenderelliot98: that is just what some potentially well-meaning but equally uninformed twit shoved on a 3rd party site
18:42.31profxavierFender, to please you I tried it once again
18:42.37profxavierand I am still here
18:43.07*** join/#asterisk Skeeter- (i=Skeeter-@190-141.cgocable.ca)
18:43.10[TK]D-Fenderprofxavier: And to displease me you again never show me what you'd done.  All I get is another story of "it doesn't work"
18:43.18Skeeter-Hi everyone
18:43.29profxavieri posted what my original plan was
18:43.36profxavierwe discussed what to add
18:43.39profxavierand I did that
18:43.46*** join/#asterisk justdave (n=dave@unaffiliated/justdave)
18:43.48elliot98ok...so where is the official Asterisk support pages that give T.38 information?
18:43.51profxavierthere his no way I can win, either way
18:44.15TheDavidFactorpastebin for the win!
18:44.24profxavierim sure you know your shit, but as for relations with outs, its no wonder you are on IRC
18:44.30profxavieri would hate to meet you in real life
18:44.58ManxPower-workelliot98: /path/to/src/asterisk/doc
18:45.07Skeeter-if i use the Pickup function when i get an inbound call from an IAX Trunk is doesnt work
18:45.14ManxPower-workelliot98: those are the official Asterisk support docs
18:45.23ManxPower-workSkeeter-: IAX does not support pickup
18:46.17ManxPower-workSkeeter-: I assume you mean Pickup as controlled by callgroup and pickupgroup, rather then the directed call pickup.
18:46.33elliot98and where does it mention T.38 and fax support?
18:46.50Skeeter-ManxPower-work: I meant Directed call pickup
18:46.53leifmadsen[TK]D-Fender: play nice
18:46.55ManxPower-workelliot98: no idea.
18:47.12ManxPower-workSkeeter-: I don't know if chan_iax supports directed call pickup or not.
18:47.27ManxPower-workleifmadsen: Thanks for volunteering to help profxavier!
18:47.39leifmadsenprofxavier: what's the issue?
18:47.40Skeeter-ManxPower-work: Thanks for the tip
18:48.07elliot98so Asterisk developers don't discuss T.38?
18:48.25ManxPower-workelliot98: you might ask on the #asterisk-dev channel
18:48.55leifmadsenelliot98: what information are you looking for?
18:49.05elliot98I am looking for T.38 information
18:49.12elliot98how asterisk supports it
18:49.12leifmadsenthat's pretty vague
18:49.13elliot98if
18:49.24DeeewayneKatty, still need help w/ fax?
18:49.31Kattysure.
18:49.34ManxPower-workleifmadsen: welcome to our world.  Group therapy is on tuesday evenings.
18:49.35Kattybut i'm still trying to compile this package
18:49.42leifmadsenManxPower-work: you're not being useful
18:49.44Kattyit keeps whining about a lack of compiler, so i'm trying to find a gcc one
18:50.04profxavier<PROTECTED>
18:50.04ManxPower-workKatty: I think spandsp wants c++ compiler
18:50.19KattyManxPower-work: yeah i'm poking through the config.log to see why it's whining
18:50.27leifmadsenelliot98: what kind of information? there is no T.38 gateway support in Asterisk currently if that's what you're looking for, but there is T.38 pass-through support. Reading the UPGRADE.txt file, at least in 1.6.2 has lots of information about T.38 support
18:50.34ManxPower-workgcc-g++ or gcc-c++ or something like that is what most distros call the package
18:50.42profxavierleifmadsen: but when I dial '9' + 10 digits, with the receiver picked up it doesnt work
18:51.06Skeeter-profxavier: let me check my configs, i solve this problem
18:51.07Kattyit appears to want gawk.
18:51.24Skeeter-profxavier: wont you mind telling which file contains that please
18:51.31Skeeter-i cant remember
18:51.34leifmadsenprofxavier: well I currently count 9 + 9 digits
18:51.55leifmadsenprofxavier: have you done a tshark trace to make sure your configuration files are being loaded by the Polycom?
18:52.12elliot98thank you liefmadsen...I found some information on voip-info.org, but chatters here consider that  just what some potentially well-meaning but equally uninformed twit shoved on a 3rd party site
18:52.18leifmadsenprofxavier: there is more to that line typically -- please show the entire line of that configuration
18:52.20Skeeter-profxavier: Try this buddy : <digitmap dialplan.digitmap="9[2-9]xxxxxxxxxx|9[1]xxxxxxxxxx|[0-8]xx" dialplan.digitmap.timeOut="10"/>
18:52.37leifmadsenelliot98: yes, voip-info.org is not official, and is typicalyl out of date
18:53.04Skeeter-profxavier: Try to avoid using 1XX-2XX ext.
18:53.10elliot98I've come to notice that...but are there any online alternatives?
18:53.18leifmadsenelliot98: reading through the doc/ directory in your asterisk source is pretty much the only place you're going to get up to date information. The configs/*.sample files are also useful
18:53.27leifmadsenelliot98: you can try asterisk.org
18:53.34elliot98and the official Asterisk book
18:53.54leifmadsenelliot98: there is no official Asterisk book
18:54.12elliot98woopsy...what is the book offered on their site?
18:55.20leifmadsenelliot98: probably the O'Reilly book
18:56.09elliot98yes...didn't know it wasn't official
18:56.33leifmadsenwell digium isn't a book publisher
18:57.36elliot98true, but it sure seems to make out that that is its de facto learning source
18:58.22leifmadsenit just might be the best book on the subject in their opinion
18:58.35leifmadsenregardless, it wasn't written by or paid for by digium, but it's nice that they like it :)
18:58.42elliot98Leif...could you believe someone had a thought of putting embedded linux w/ asterisk on an ATA?
18:58.48elliot98gotcha
18:59.08leifmadsenelliot98: yes, that was done a few years ago basically, starting with a soekris
18:59.10leifmadsen~astlinux
18:59.11infobotwell, astlinux is a great embedded Asterisk distribution initially designed for the Soekris net4801 and net5501 boards and the PC Engines wrap board - see http://www.astlinux.org or #astlinux
18:59.58elliot98but Digium cards are usually to big to fit inside a small ATA box
19:00.16elliot98wonder if anyone has tried to write a driver for other FXO/FXSs
19:00.42[TK]D-Fender[13:43]<profxavier>and I did that <- and you never show us what you did and I will never ever take it at face value that you did it right
19:01.59[TK]D-Fender[13:49]<profxavier> I have: "[2-9]11|0T|011xxx.T|[0-1][2-9]xxxxxxxxx|[2-9]xxxxxxxxx|[2-9]xxxT" <- still not the cleaned up version you were supopsedt o ahve done
19:02.03[TK]D-Fendersupposed*
19:02.29elliot98Manxspower...so it seems it's already been done...
19:02.53elliot98and it didn't combust
19:02.55[TK]D-Fender[13:57]<elliot98>true, but it sure seems to make out that that is its de facto learning source <- what is "it"?
19:03.12elliot98it => asterisk.org
19:03.12[TK]D-Fenderelliot98: And a Soekris isn't an ATA <-
19:04.08elliot98true...it is not an ATA...but it sure is pretty damn close
19:04.09[TK]D-Fenderelliot98: Can you show me the page where they say TFOT is the defacto learning sourse for *?
19:04.17elliot98http://www.asterisk.org/support Online Documentation
19:04.33elliot98de facto...for Online Documention, that is
19:04.48[TK]D-Fenderelliot98: And a car is pretty damn close to a plane.  Both burn fossil fuels and are motorized forms of transportation.... so why can't my car fly?
19:05.12elliot98how much different is an ATA board and the Soekris?
19:05.13Qwell[TK]D-Fender: because it lacks FXS ports
19:05.38*** join/#asterisk joako (n=joako@opensuse/member/joak0)
19:05.41Qwelllittle known fact: planes fly because they have FXS ports
19:05.41elliot98if one can find a chipset that is compatible...one is good to go
19:05.45[TK]D-FenderQwell: So ... what you're saying is my quad-core Xeon rackmount server with a TDM410P w/ 4 FXS... is an ATA?
19:05.52Qwell[TK]D-Fender: No, it's a plane.
19:05.55Qwellerr, wait
19:05.57elliot98and submarines dive because they have FXOs?
19:06.01[TK]D-FenderQwell: \o/ AWESOME!
19:06.07Qwellelliot98: no, that's just absurd
19:07.26elliot98D-Fender...so how much does an ATA and Soekris differ?
19:07.39leifmadsenCorydon76-dig: so I have the 'uniqueid' field in my CDRs, and it has a number like:   12444809880.4   <-- however with the ${UNIQUEID} value in the dialplan I have:  12444809880.44010. Am I doing something wrong that causes the decimals to be truncated to a single digit?
19:08.02[TK]D-Fenderelliot98: Soekris is an entire X86 platform with a shit-ton of normal interfaces.  It is a bloody computer
19:08.06leifmadsenCorydon76-dig: the reason I ask is I'm having issues with calls coming in about the same time, and think I could avoid them if uniqueid wasn't truncated
19:08.13Kattymoronic question. after building spandsp, i'm not sure where my app_rxfax.c, etc is to put them in the /usr/src/asterisk/apps folder
19:08.21Kattyand locate is not being useful
19:08.29Kattyinfobot: app_rxfax.c?
19:08.30elliot98and linux was never developed for ARM9
19:08.48*** join/#asterisk DrAk0 (n=luisjose@nelug/coreteam/luisjose)
19:09.10Kattynot in my app folder :<
19:09.37elliot98if http://www.jeremy-mcnamara.com/2007/10/28/asterisk-now-runs-on-iphone/
19:09.46elliot98why can't an ATA be far off?
19:09.58[TK]D-Fenderelliot98: Go ahead and jsut build your own SBC * box.  It simply won't be an "ATA"
19:10.52[TK]D-Fenderelliot98: What OS do ATA's run?  Does it have enough POSIX support for * to run?  What about ram?  what about file-system space?
19:10.53*** join/#asterisk TSM2 (n=the_soft@87-194-32-212.bethere.co.uk)
19:11.09[TK]D-Fenderelliot98: Why is the sky blue?
19:11.14[TK]D-Fenderelliot98: Who shot J.R.?
19:11.28[TK]D-Fenderelliot98: What is the average air-speed velocity of an unladen swallow?
19:11.38elliot98good point...but a good place to start searching for answer is right here
19:11.54[TK]D-Fenderelliot98: No, the good place is GOOGLE.
19:12.56[TK]D-Fenderelliot98: Oh, and that asterisk.org page doesn't list it as "defacto"... it is jsut another book, in the list of books there
19:13.09elliot98check out Online Documentation
19:13.19elliot98it is the only reference...except for a glossary
19:13.38elliot98and some Digium card sites
19:13.42Kattyoh where oh where has my rxfax gone, oh where oh where oculd it be
19:14.39[TK]D-Fenderelliot98: Not defacto, and book are documentation and the O'Relly book just happens to be AVAILABLE on-line
19:15.00Skeeter-[TK]D-Fender: have you ever incounter any problem with the pickup function via an IAX2 Trunk
19:15.33[TK]D-Fenderelliot98: Microsof links to publisher's docs too... for books they do not author or fully validate
19:16.33*** join/#asterisk osiro (n=osiro@189.111.254.251)
19:16.41[TK]D-Fenderelliot98: .. and the knowledge base below that...and the Digium Support Center below that...
19:17.07[TK]D-Fenderelliot98: Yes, I see your case of tunnel-vision quite well
19:17.21[TK]D-Fenderelliot98: Sorry... jsut doesn't wash...
19:17.26[TK]D-Fenderelliot98: Move along nothing to see here
19:17.50osiroGuys, to register an extension without authentication is just remove the secret parameter of this extensions in sip.conf?
19:17.55*** join/#asterisk saghul_ (n=saghul@ip51ccb640.speed.planet.nl)
19:18.15*** join/#asterisk MatBoy (n=MatBoy@wiljewelwetenhe.xs4all.nl)
19:18.43profxavierleifmadsen: still around ?
19:19.17ManxPower-workosiro: AND remove the password from the sip client
19:19.34leifmadsenprofxavier: KINDA
19:19.34profxavier"[2-9]11|0T|011xxx.T|
19:19.36profxavier9[2-9]xxxxxxxxxx|
19:19.37leifmadsenerrr.. kinda :)
19:19.38profxavier[2-9]xxxxxxxxx|[2-9]xxxT"
19:20.01profxavieri edited 9xxxxxxxxx to be 9[2-9]xxxxxxxxx
19:20.08leifmadsenok
19:20.26profxavierseems to fall into [2-9]xxxT
19:20.48leifmadsenprobably because your T (timeout) is too short
19:21.09leifmadsenjust remove all the pattern matches you're not using. In fact, just remove them all except the one you're testing, and then add back in what you need after you get it working
19:21.11leifmadsenstart simple
19:21.14profxavierok, so I have to adjust the Digitmap Timeout, as well as the Digitmap itself
19:21.21osiroManxPower-work: thanks!
19:21.23leifmadsenthat's my guess
19:22.42profxavierleifmadsen: ill give that a try
19:23.29*** part/#asterisk scottsmith7 (n=hammer@64.201.141.80)
19:25.42profxavierok, it worked
19:25.52profxavier9xxxxxxxxxx
19:26.00[TK]D-Fenderwow
19:26.22*** part/#asterisk osiro (n=osiro@189.111.254.251)
19:27.04[TK]D-Fenderprofxavier: You mean the way I told you to do almost what... 2 hours ago?
19:27.26profxavierno, the way leifmadsen told me, remove everything in it
19:27.50[TK]D-Fender[14:19]<profxavier>i edited 9xxxxxxxxx to be 9[2-9]xxxxxxxxx
19:28.05[TK]D-Fender[14:25]<profxavier>9xxxxxxxxxx
19:28.21[TK]D-Fenderprofxavier: He may have, but I beat him to it by a few hours
19:28.27profxavierFender, how are you assisting me exactly?
19:28.37[TK]D-Fenderprofxavier: And reiterated it it a few times
19:29.05profxavierbut, now, after taking your advice, you understand why it works...
19:29.14[TK]D-Fenderprofxavier: Oh I "helped" you then.  It took you this long to follow the advice after hearing it from someone else.
19:29.36profxavierFender, I did follow the advice, and I explained
19:29.40profxavierit wasn't working
19:30.10profxavierofferring a helpful comment, such as, start fresh, is more helpful, then you constant questions
19:30.10[TK]D-Fenderprofxavier: You never showed me your completed and proper line once regardless of my telling you to do so repeatedly
19:30.45profxavieri really don't see why this needs to be discussed in the channel
19:30.56profxavierfeel free to pm me, if you have anything else to say, thanks
19:31.02profxavierim here for help
19:31.25[TK]D-Fenderprofxavier: let us know when you're ready to follow it and instructions you're given to get help.
19:31.53profxavieri already have, and leifmadsen was very helpful
19:32.02profxavieri think you are missing the point
19:32.10[TK]D-Fenderprofxavier: Same here.
19:33.56leifmadsen[TK]D-Fender: stop
19:34.14elliot98okay...enough fun for now
19:38.47cuscohi.. what is best to connect several asterisk machines togheter, for example: a gateway geting calls from pstn and passing them on the another asterisk machine that will decide to wich extention it will go
19:38.57cusco(not sure if I made my point clear)
19:39.21leifmadsencusco: see DUNDi
19:39.23leifmadsen~dundi
19:39.24infobotdundi is, like, at http://www.dundi.com. DUNDi, an optional Asterisk component, is a distributed, decentralized peer to peer network that provides routes to PSTNs between peers on the same DUNDi network.
19:39.25cuscothanks
19:40.21[TK]D-Fendercusco: DUNDi is an option, s it just using SIP/IAX peers like any other service
19:40.44[TK]D-Fendercusco: Depends how you need to segragate access, etc
19:41.17*** part/#asterisk rpm (n=russell@S0106000c29898b7e.cg.shawcable.net)
19:45.59kaldemarinfobot is pretty narrow minded when it comes to DUNDi.
19:47.57*** join/#asterisk Skeeter- (i=Skeeter-@190-141.cgocable.ca)
19:53.19Skeeter-ManxPower-work: I found this, seems like you posted regarding this issue. The person said that now the pickup via IAX2 trunk works, using this: http://pastebin.com/m78fdcbd5
19:53.58*** join/#asterisk e` (n=IceChat7@ool-4352de6e.dyn.optonline.net)
19:58.53ManxPower-workSkeeter-: DIRECTED pickup.  "Pickup" is totally different.  *8 is used for Pickup.  You don't get to specify a destination extension with regular pickup.
20:00.30*** join/#asterisk rdahlin_1 (n=rdahlin_@78-73-17-198-no168.tbcn.telia.com)
20:00.30ManxPower-workdid you do a "core show application pickup"?
20:00.38Skeeter-aight, i see the point now. its very unfortunate that Directed pickup doesnt work via IAX2 trunks
20:00.53Kattythis free fax for asterisk license takes /forever/ to ship to me.
20:00.56ManxPower-workDirected pickup should work.  *8 pickup does not.
20:01.13Skeeter-im trying to use **, and it doesnt work
20:01.21Katty20 seconds! that's so unacceptabuhls
20:01.36*** join/#asterisk errotan (n=errotan@a1621.adsl.pool.eol.hu)
20:01.37ManxPower-workpastebin your extensions.conf section and the CLI output of a failed pickup
20:01.47leifmadsenKatty: blasphemy!
20:03.41Kattysticks tongue out at leifmadsen
20:03.51ManxPower-workSkeeter-: Um DIRECTED pickup requires an EXTENSION@CONTEXT
20:03.58[TK]D-FenderManxPower-work: LOL!
20:04.04leifmadsensticks his tongue out at grooveshark for not having enough bandwidth to stream music to him
20:05.25ManxPower-workSkeeter-: you never read the cods for the pickup app did you?
20:05.33Corydon76-diggrabs tongue, pierces it with a metal loop, and attaches chain for leading leifmadsen around with it
20:05.33ManxPower-works/cods/docs
20:05.52leifmadseneep!
20:05.55Skeeter-ManxPower-work: I added this to the app-pickup: exten => _**.,n,Pickup(${EXTEN:2}@from-trunk-iax2-2-peer) : but i havent tried it yet
20:06.24ManxPower-workSkeeter-: So, you are taking EXTEN (which is **) and stripping off the first 2 chars.
20:06.27Corydon76-digJust kidding!  The chain is to keep you tied up, not for leading you around.
20:06.50ManxPower-workSkeeter-: I'm still waiting for that pastebind
20:07.01[TK]D-FenderManxPower-work: Oh boy...
20:07.10[TK]D-FenderManxPower-work: Its gonna be HUGE :)
20:07.12ManxPower-workSkeeter-: and I cannot just hang out all day waiting for you.
20:08.35leifmadsencan anyone think of something I'm missing fundamentally from this outline?  http://www.pastebin.ca/1665356
20:08.49*** join/#asterisk ariel_ (n=chatzill@63.214.236.169)
20:10.05Skeeter-ManxPower-work: Should not take long VPN is down
20:10.35ManxPower-workSkeeter-: revisit the Pickup problem when you can focus on the problem.
20:11.48Corydon76-digleifmadsen: Is pausing any different?
20:12.11[TK]D-FenderManxPower-work: ===Skeeter-: member of #asterisk and #freepbx
20:13.03[TK]D-Fenderleifmadsen: Thats going to make a whole bunch of people happy...
20:13.47Skeeter-[TK]D-Fender: i ask questions at both place sometimes the answer pop-up only on one side
20:13.52ManxPower-work[TK]D-Fender: Oh!  Thanks.  He's on his own then.
20:14.20ManxPower-workI do too much work with that PoS FreePBX for work.  I'm not helping people with their FreePBX/
20:15.21leifmadsenCorydon76-dig: ah yes, I need to add a section on pausing members, thanks!
20:15.32Skeeter-how is freepbx interfering with asterisk
20:16.06Kattydoes anyone have a snippet of RecieveFax or SendFax they can pastebin for me, pretty please? or a wiki article would work
20:16.42ManxPower-work~freepbx
20:16.42infobot[~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there
20:17.08ManxPower-workchances are the extension is not in the context you think it is in.
20:17.18*** part/#asterisk ManxPower-work (n=EWieling@24.42.221.26)
20:19.19Skeeter-ManxPower-work: ok, Thanks
20:22.03Kattyinfobot: Faxing?
20:22.04infobotwell, faxing is 8% knowledge, 5% skill, 11% luck, and 76% voodoo
20:22.14Kattyinfobot: Fax for Asterisk
20:22.27russellb~ffa
20:22.27infobotffa is probably ffa stands for free for all, or not CTF
20:22.28Kattyinfobot: Fax for Asterisk is http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf
20:22.29infobotKatty: okay
20:22.42leifmadsenCorydon76-dig: I'm thinking I might actually do a blog post about building the queue first on a single server, and then do another post that builds on top of that to do the distribution
20:22.49Katty(ffa loot)++
20:25.31Qwellleifmadsen: that sounds like a good idea.  make it clear that you'll be expanding on it though
20:26.11leifmadsenQwell: yep, already added the note to the other outline that says I'm building on a previous article
20:26.26QwellACK
20:26.26Qwell:p
20:27.40Corydon76-digQwell: NAK
20:30.48eppigyTRABAJO
20:31.15jayteethe asterisk.org site no longer lists any release candidate versions. I'm kind of wondering with the release of 1.6.1.9 what that means for 1.6.1.7-rc1
20:33.08jayteeno trabajo, gripe de los cerdos
20:33.56leifmadsenjaytee: see the release announcements
20:34.19leifmadsenjaytee: the release announcement for 1.6.1.8 and 1.6.1.9 should explain exactly what happened to 1.6.1.7-rc1
20:34.48leifmadsenjaytee: http://www.asterisk.org/node/49863
20:34.51*** join/#asterisk joesuffceren (n=chatzill@ip68-104-167-226.ph.ph.cox.net)
20:34.56[TK]D-Fenderjaytee: Also doesn't list asterisk-addons, or the HTTp server to cuise through or a bunch of other stuff, but its being worked on...
20:35.41leifmadsenalso, I don't think it ever listed release candidates -- it only ever showed the latest 1.4 and 1.6.x release candidates, and not all of them
20:35.54leifmadsenhttp://downloads.asterisk.org/pub/telephony/asterisk/ always has the latest RCs
20:35.55jayteeleifmadsen, thanks. kind of got me confuddled there for a bit but now after clicking on the update in News I get the link for the download.
20:36.07joesuffcerenanyone terribly familiar with Cisco 7940s running SIP and callerid and/or know of a good resource? I'm trying to get them to send their own callerid instead of the incoming caller's when they forward a call
20:36.26leifmadsenjaytee: feel free to email ssokol at digium about anything on the website you need ;)
20:37.10jayteeI'm thinking since 1.6.1.6 SIP/TCP didn't work with Exchange UM and 1.6.1.7-rc1 did work that 1.6.1.9 probably won't work. Of course I'm making an assumption about that.
20:37.19elliot98does Windows provide any sort of QOS for softphones?
20:37.52elliot98the sip-phone work great in the office, but softphones lack quality
20:38.06leifmadsenjaytee: look at the release announcements -- 1.6.1.9 is not a bug fix release
20:38.19leifmadsenjaytee: either was 1.6.1.8 -- they are basically 1.6.1.6 + security fixes
20:38.24elliot98are softphones generally inferior or is there a qos issue when it is attached to a Windows computer?
20:38.30leifmadsenactually, not basically; they are 1.6.1.6 + security fixes
20:38.54leifmadsen1.6.1.7 was dropped, and 1.6.1.10-rc1 and -rc2 are the continuation of bug fixes from 1.6.1.7-rc1
20:39.06Qwellelliot98: Do you have crappy headsets?
20:39.35*** join/#asterisk sahafeez (n=sahafeez@65-119-47-100.dia.static.qwest.net)
20:39.55[TK]D-Fenderelliot98: headset quality.  Sound card quality.  AEC quality.  mix & match
20:39.55jayteeleifmadsen, I know...that's why I figured I need to wait for a higher version release than 1.6.1.9
20:39.59sahafeezwill reload chan_dahdi.so drop all calls
20:40.12*** join/#asterisk Gugge (n=gugge@vlan2.dlxhosting.dk)
20:40.13[TK]D-Fendersahafeez: All DAHDI calls... yes
20:40.44sahafeezso my pri, etc.
20:40.52sahafeezugh
20:41.08sahafeezi added an fxo and need to have it load
20:43.05theharrussellb: awesome post!
20:43.18thehartwo thousand three hundred and twenty commits!
20:43.26p3nguinCan two host addresses be entered for a sip peer?
20:43.32russellbthehar: just to trunk :-)
20:43.33[TK]D-Fenderp3nguin: No
20:43.37theharsexy
20:43.42*** join/#asterisk ChUbB (n=IceChat7@62-31-213-230.cable.ubr12.aztw.blueyonder.co.uk)
20:43.57thehar1.8
20:43.58thehardrools
20:44.16p3nguinSo when the ITSP says calls will come from .46 and .47 IPs, should I just use host=dynamic?
20:44.19russellbyou should comment on the post!
20:44.23*** join/#asterisk war9407 (i=war@liquidswords.org)
20:44.33russellbthehar says: "OMG I <3 RUSSELLB!"
20:44.38thehari shall now
20:44.40theharverbatim
20:44.48russellbha
20:44.56*** join/#asterisk ttl- (n=patrick@94-224-78-192.access.telenet.be)
20:45.10thehari need to finish reading it first
20:45.38russellbpfft.
20:46.00elliot98someone told me there were softphone issues because they don't tag their packets with quality of service
20:46.09russellbso your declaration of awesomeness was a bit preemptive
20:46.10theharrussellb: unlimited x <3
20:46.15theharno
20:46.18theharyou ARE awesome
20:46.20russellbthere might be some unawesomeness in there
20:46.51jayteerussellb, what's the best version for me to move to if I was running 1.6.1.7-rc1 in a test environment and want to continue to test with the latest code that's not a security release. Is the code in 1.6.1.10-rc1 the next step up from 1.6.1.7-rc1?
20:46.51*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
20:47.18russellbjaytee: whatever the latest RC is will be the next bug fix release.  I expect it to be released this week
20:47.18*** join/#asterisk jeffgus (n=jeffgus@2002:ad33:b504:0:0:0:0:1)
20:47.22elliot98but I really doubt that is the problem, since the main router gives priority to the IP addresses of the remote asterisk server
20:47.25p3nguinI don't really want to use host=dynamic when I know the only two addresses they will use.  Isn't there another way?  They don't seem to have the appropriate SRV records.
20:47.32russellbjaytee: which I do believe is 1.6.1.10-rc1
20:47.32theharhaha russellb i remember these releases.. sheesh
20:47.44*** join/#asterisk Tim_Toady (n=moi@adsl85-214.kln.forthnet.gr)
20:47.48[TK]D-Fenderp3nguin: use host=dynamic and use permit/deny to restrict them
20:48.05p3nguinOh, right.  I should have thought of that!
20:48.07[TK]D-Fenderp3nguin: if this is for security reasons
20:48.09elliot98and the packets to and from a windows computer are in an internal network
20:48.20elliot98so there should be no need of qos inside a windows computer
20:48.39elliot98as far as I can see
20:49.18leifmadsenacknowledges russellb's statement of 1.6.1.10-rc2 being the latest RC for the 1.6.1 series
20:49.31leifmadsenrussellb: remember that we did an RC2 to fix a small issue in func_volume or something
20:50.20russellbah yes
20:50.24russellband that is why we have a release manager :-)
20:50.26jayteeleifmadsen and russellb, yep and it's up on the server available for download. I'll upgrade to that version and test. hope that works, wanna get my production systems to 1.6.x and get rid of sipX proxy to Exchange by end of year.
20:50.26russellb<3 leifmadsen
20:50.36*** join/#asterisk Iamnacho (i=Iamnacho@ip98-186-180-143.ks.ks.cox.net)
20:50.50russellbjaytee: there are some really good TCP related robustness improvements in this release
20:51.08russellblike, don't take down the system in the face of adverse network conditions type fixes :-)
20:51.12jaytee<3 leifmadsen too, but he's not my facebook buddy like russell is :-(
20:51.20russellbbut he could be!
20:51.30russellbok, i seriously need to read this document ... *starts to ignore IRC*
20:53.33leifmadsen<3 russellb!
20:55.38theharrussellb: commenting!
20:55.50elliot98is 1.2 still developed?
20:56.04russellbsecurity fixes only
20:56.06Qwellelliot98: Only for security issues
20:56.12Qwellrussellb: go away.  ignore IRC :p
20:56.18russellboh yeah
20:56.43theharis such a damn asterisk geek :(
20:57.55*** join/#asterisk wonderworld (n=ww@ip-62-143-22-226.unitymediagroup.de)
20:59.38dlynesAnyone happen to know which module it is that I need to unload and reload for the asterisk manager interface?
21:00.42dlynesrussellb: those type fixes are for asterisk 1.6.1.9?
21:00.54dlynesrussellb: or 1.6.1.10 when it comes out?
21:01.10thehardlynes: reload manager
21:01.17dlynesthehar: ah...thanks
21:01.26leifmadsenmodule I think is res_manager.so
21:02.21dlynesleifmadsen: the only manager anything module is res_cdrmanager.so, which isn't it, unforunately
21:02.28dlynesleifmadsen: but thehar's trick worked
21:02.39thehartrickery
21:04.10leifmadsenya, I'm just remembering the module name wrong then
21:04.22*** join/#asterisk ruyo (n=psantos@85.138.89.253)
21:04.50*** join/#asterisk ManxPower-work (n=EWieling@24.42.221.26)
21:07.10Qwellmanager is core
21:07.52leifmadsenoh duh
21:08.20dlynespats blitzrage on the head.
21:08.37leifmadsenmust be the end of the work day
21:10.33*** join/#asterisk tzafrir (n=tzafrir@bzq-218-155-145.cablep.bezeqint.net)
21:16.45*** join/#asterisk voipmonk (n=voipmonk@69.172.70.23)
21:26.12*** join/#asterisk andres833 (n=andres83@190.144.75.22)
21:29.52jayteetime to quit, be back later
21:30.13*** join/#asterisk galeras (n=galeras@186.80.181.115)
21:33.15*** part/#asterisk galeras (n=galeras@186.80.181.115)
21:33.39*** join/#asterisk jmacz (n=jmacz@190.144.75.22)
21:42.24*** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber)
21:43.05Kattywhere is rm located?
21:43.38Kattyinfobot: rm
21:43.39infobotrm is probably uh.... NO CARRIER
21:43.43QwellKatty: which rm
21:43.48Kattythe one in /bin
21:43.53Qwelltype that
21:44.20Qwelland manwich
21:44.23Qwellerr, man which
21:44.49Kattyi found it.
21:46.08*** join/#asterisk scardinal (n=supreme@90.184.100.119)
21:46.27p3nguinIt seems that using host=dynamic and then creating an ACL for the two host addresses causes the default context to be used rather than the context for the peer... if I set host=ip.address, the peer's context is used correctly.
21:47.08*** join/#asterisk pirulo (n=andres@65.102.99.5)
21:47.10Qwellp3nguin: what version of Asterisk?
21:47.28p3nguinkatty: "which" is used to see which executable is being called when you only use its name.
21:47.46p3nguinqwell: 1.4.24.1
21:48.21*** join/#asterisk geneticx (n=chatzill@host-208-88-126-198.biznesshosting.net)
21:48.48Qwellp3nguin: tried upgrading?
21:49.22p3nguinI tried 1.4.26, but there was some obstacle that prevented me.  Now I don't remember what the problem was.
21:49.55QwellWell, if you're relying on ACLs like that, you're going to want to upgrade.
21:50.27p3nguinI'm trying to use IPKall's SIP (or IAX) delivery which does not use any type of registration nor authentication.  Using a regular SIP URI, the call hits the default context, since it's a guest (anonymous) peer.
21:51.00Qwellcreate 2 peers
21:51.19QwellThe problem is calls coming in from multiple addresses, right?
21:51.23p3nguinyes
21:51.32Qwelltell them to use SRV :p
21:51.38p3nguinI wish.
21:51.48TSM2does anyone have a fail2ban script for asterisk failed SIP/IAX logins
21:51.51p3nguinThat would save some trouble, but they just have two host names instead.
21:52.24TSM2just found one, no worries
21:52.28TSM2~fail2ban
21:52.29infobotfrom memory, fail2ban is a program to ban people using iptables based on information in logs: http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk
21:52.50p3nguinTwo peers, one for each address, will solve the issue and let me create a real context for those calls and get them out of default?
21:53.03Qwellp3nguin: should..  until they add a 3rd
21:53.04p3nguinI guess it would.
21:53.13Qwellor they could just add SRV, which takes like 8 seconds
21:53.55p3nguinIf Google Voice would just use SIP URIs instead of requiring a phone number for forwarding, I wouldn't have to worry about it at all.
21:54.22p3nguinThey use SIP, so it can't be too hard to allow a phone number _or_ a URI.
21:54.31TSM2does asterisk use SRV records for URI dialing?
21:54.42p3nguinit can
21:55.12TSM2~SRV
21:55.13infobotsrv is, like, "Service Record". e.g. _sip._UDP.example.com => 5060, pbx1.example.com . See http://en.wikipedia.org/wiki/SRV_record
21:55.37*** join/#asterisk gr0mit (n=tim@2001:8b0:3d9:0:6422:a0e0:9b31:9082)
21:56.37TSM2so using SRV i can essentialy either have 2200@company.com or joe@company.com and it will figure it out and call my * box
21:57.03QwellTSM2: no
21:57.04voipmonkyou can do that without SRV
21:57.24TSM2but my A records dont point to my voice server
21:57.26QwellRead the "High Availability with SRV" section on the wiki article
21:57.47TSM2i thought that came from having SRV SIP records on the root domain
21:58.33p3nguinIf you have a domain name, you can create an SRV record for it with multiple hosts.  Then all those hosts can be included by using host=domain in your peer entry.
21:58.59p3nguinone host entry, all the hosts in the SRV response.
21:59.32p3nguinI hope I said that correctly.
21:59.36Qwellnot quite
21:59.51p3nguinBest to read it somewhere else then.  I tried.
22:00.18TSM2yup so essentialy i can do what i said using SRV records, so instead of giving tsm@voip1.company.com as my voip addy i can just give tsm@company.com and it will locate my voip server though the SRV entries
22:00.44TSM2then put sip aliases for my exten
22:00.53*** join/#asterisk uqlev (n=yuriy@91.184.221.31)
22:01.16ManxPower-workI have used SRV records to make my ATA work correctly as I moved it between my inside network and outside of my network.  Worked great.
22:01.19*** join/#asterisk waKKu (n=vakku@unaffiliated/wakku)
22:01.56p3nguintsm2: I believe that's right; just make sure you have the srv setting enabled in *.
22:03.02jblackI'm sorry guys.
22:04.02KnightfalHey guys! Im wondering if wrapuptime is working as it should in agents.conf  Will agents that are members of different or multiple queues get 15 secs of wrapup after a call from any queue?
22:04.08jblackI'm a very unlucky person. However, I'm having a very good day. That means something very bad is about to happen...   alien attack, asteroid swarm.. LHC producing a black hole... but whatever it is that ends the world, know that I regretted it
22:05.14voipmonkall your asterisk are belong to us
22:05.24Knightfalwoot
22:05.57TSM2jblack: it will break :) 5mins before i was ment to leave today a HDD on our DB server packed up grrrr, all kept working though but still....
22:07.27waKKuhi folks.. could someone help me migrating from zaptel to dahdi? i'm getting this message: WARNING[7254]: chan_dahdi.c:2784 pri_find_dchan: No D-channels available!  Using Primary channel 47 as D-channel anyway!
22:10.04waKKulog from dahdi restart here http://pastebin.com/m50913406
22:10.38*** join/#asterisk malaiwah (n=mbelleau@host-64-47-115-5.masergy.com)
22:15.15*** join/#asterisk deeperror (n=deeperro@76.226.172.218)
22:21.33Kattyi don't suppose anyone knows how to share something with samba that does not prompt the user for a username/pass when attempting to access the share.
22:22.07beekKatty:  yep
22:22.22beekKatty: It's OT for this channel... mind a private PM?
22:22.29Kattysure!
22:23.45jblackkatty: Yeah. turn on guestok
22:24.26jblackhere's some instructions here: http://www.debuntu.org/guest-file-sharing-with-samba
22:27.43Kattykay
22:28.33*** join/#asterisk saftsack (n=oliver@p579DDBD9.dip.t-dialin.net)
22:29.51Kattyhoray!
22:30.33*** join/#asterisk siddolo (n=asd@78.134.25.124)
22:30.46siddolohi all
22:30.57siddolosomeone can help me on ring groups?
22:31.42beek~ask
22:31.43infobotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
22:33.41beeksiddolo: Are you going to ask your question?
22:33.52siddoloI've made one ring group with two user. strategy ring simultaneously. If i call, the 2 user's phone ring correctly.. i answer...
22:33.56siddoloif second call arrive
22:34.08siddoloring only phone of the first user
22:34.39p3nguinHow us your configuration.
22:34.44p3nguinin a pastebin.
22:34.48p3nguin~pb
22:34.49infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
22:35.35jblackI'm up to 7 friends.
22:35.35jblacksiddolo: That sounds like a bug.
22:36.00siddoloone moment..
22:36.14*** part/#asterisk Malkor (n=marco@hlle-d9ba009b.pool.mediaWays.net)
22:37.28Kattyis there an audio file that says something like please enter the name
22:37.31Kattyplease enter the digit
22:37.45Kattyplease enter.... something that has to do with numbers
22:38.17Kattyvm-enter-num-to-call <- not that.
22:38.23Kattybut similar.
22:39.36siddolohttp://pastebin.com/d40662fe4
22:40.01jblackhmm
22:41.00jblackthere's a few possible ones.
22:41.11*** join/#asterisk gscmans (n=guna@77-99-69-92.cable.ubr16.haye.blueyonder.co.uk)
22:42.40jblackhow about please-enter-the&number
22:43.10jblackKatty: ^^
22:44.25Kattyenter_filename.gsm <- i think that might be better
22:44.34jblackOk
22:44.56gscmansHi is there any open source ATA adapter kit that runs asterisk firmware, that could be customized?
22:45.00jblackI don't have enter-filename here
22:45.19jblackYou didn't mention what you're prompting for, btw
22:45.29Kattyit's under dictate
22:45.32siddoloi need to post full extension.conf?
22:45.49Kattyi found a document that shows exactly what is said in each audio file.
22:54.08beekWait... Katty, you're doing Asterisk again?
22:55.31Kattyi'm playing with faxing.
22:55.54beekAhhh... still, it's nice to see you back in the game on this platform.
22:57.28*** join/#asterisk ecrane (n=ecrane@o1-69-19-166-10.static.o1.com)
22:57.53Kattyhttp://42ndgeekstreet.blogspot.com/2009/11/asterisk-faxing.html <- that's what i'm doing
22:59.55*** join/#asterisk siddolo (n=asd@78.134.25.124)
23:00.09Kattystill trying to get my SendFax(/path/to/${TITLE}) macro sorted out
23:00.13*** join/#asterisk deeperror (n=deeperro@d149-67-49-94.try.wideopenwest.com)
23:00.24Kattybut it's time to go home!
23:00.57siddoloi use asteriskgui 2.0
23:01.11p3nguin~gui
23:01.12infobotgui is, like, (Graphical User Interface) Any system that uses graphics to represent the functions of a program. All Windows operating systems are GUI. or point-and-drool, double-click, drag-and-drop, such as http://toastytech.com/guis/miscwin1xp.png.  Of course Real Programmers use the command line interface.  See cli
23:01.26russellbp3nguin: troll!
23:01.26p3nguin~asterisk-gui
23:01.27infobot[~asterisk-gui] The Asterisk-GUI is an open source project lead by Digium. It's client-driven, and its only dependency is Asterisk itself. Check out the new version from svn here: $ svn co http://svn.digium.com/svn/asterisk-gui/branches/2.0 asterisk-gui-2.0.  For support go to  #asterisk-gui
23:01.27siddolothe ringgroup was configured by asteriskgui
23:03.40*** part/#asterisk ehsjoar (n=ehsjoar@c-98-245-155-132.hsd1.co.comcast.net)
23:05.35*** part/#asterisk bsaxon (n=bsaxon@12.68.234.174)
23:06.00siddoloActive Channels - 23, is normal? i have no currently call active at this thime
23:07.04siddolohangup one of this channel trough asteriskgui not work
23:07.23russellbgo to the Asterisk command line page and type "core show channels"
23:07.25russellbit's probably just wrong
23:07.32*** part/#asterisk ryduh (n=ryduh@204.16.143.186)
23:07.51KnightfalHey Guys, Is wrapuptime  working as it should in agents.conf  Will agents that are members of different or multiple queues get 15 secs of wrapup after a call from any queue?
23:08.03KnightfalIm working with Asterisk 1.4.26.3
23:08.18russellbthey should
23:08.43KnightfalAgents are complaining as they always do
23:08.54siddolook it's festival() channel.. that not work
23:08.55siddolothanks
23:09.06KnightfalThanks Russell
23:09.21russellbnp
23:10.57*** join/#asterisk verywiseman (n=verywise@unaffiliated/verywiseman)
23:11.04siddolorussellb, can i hangup this channel on cli?
23:11.14russellbyou can try "soft hangup <channel>"
23:13.24russellb... you're welcome
23:13.48*** join/#asterisk siddolo (n=asd@78.134.25.124)
23:13.50siddolouhm not work.. i'm trying to stop the service.. but: Waiting for asterisk to shutdown
23:18.49*** join/#asterisk [TK]D-Fender (n=joeblow@161.216.150.205)
23:25.23saftsackwhats about the commercial digium asterisk plugin? are there any websites with experiences and informations about that?
23:25.42russellbhm?
23:25.48russellbwhich one?
23:29.17*** join/#asterisk dkirker (n=dkirker@pcp063417pcs.wireless.calpoly.edu)
23:30.48jblackanybody ever have a microsdhc card go bad on 'em in a couple weeks?
23:32.08*** join/#asterisk LordScinawa (n=scinawa@net-93-145-228-158.t2.dsl.vodafone.it)
23:32.10LordScinawahi
23:33.09waKKufolks.. how to overide de default requirecalltoken=yes from all iax2 peers without setting it one by one ? i tried set it in genereal without success (as expected) but ignorecalltoken=network/mask doesn't work too
23:34.32Kattyreturns
23:34.35Kattyhas blt!
23:34.48*** join/#asterisk jblack (n=jblack@pool-96-243-97-134.sctnpa.east.verizon.net)
23:34.56Kattymister black
23:35.01jblackHas anyone seen a 8gb microsd card go bad in 2 weeks?
23:35.07Kattyno
23:35.43jblackI am
23:36.16*** join/#asterisk Defraz (n=Defraz@corp.fuzecore.com)
23:36.23Kattyam or have?
23:36.47jblackam presently
23:37.01Kattyyou are jblack
23:37.12Kattyas opposed to have seen an 8gb microsd card go bad.
23:37.32jblackmy phone is braindead, and you're picking on my semantics. :(
23:37.43LordScinawai'm having some problem with call files
23:37.54jblackYou women are just like cats. Wait for the mouse to get zonked, and toy with 'em
23:37.57jblack<PROTECTED>
23:38.00Kattyi didn't understand.
23:38.11Kattyhowever i do now.
23:38.17LordScinawafrom reason (0) i reached Reason (8)
23:38.19jblackI'm just having fun. With the conversation, not the card.
23:38.33LordScinawasould i post call file and extension?
23:38.38LordScinawamaybe on nopaste?
23:39.09KattyLordScinawa: i'm sure that'd be condusive to people being able to see what's going on.
23:39.39LordScinawa;) i'm traying to be polite
23:39.54Kattyit's always good to be polite
23:40.27jblackI didn't mean to be rude. I'm sorry.
23:40.44Kattyyes, how dare you just quit irc.
23:40.50[TK]D-Fender~pb
23:40.51infobot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org ,  http://bin.cakephp.org/ , or apt-get install pastebinit
23:40.58[TK]D-FenderLordScinawa:^^^^
23:41.07jblackis really confused
23:41.16Kattypats poor confused jblack
23:41.27Kattyjblack: you were not rude.
23:41.36*** join/#asterisk TimToady_ (n=moi@adsl273-94.kln.forthnet.gr)
23:41.37Kattyjblack: therefore i was required to be sarcastic.
23:41.45LordScinawahttp://nopaste.info/277b74970f.html
23:41.52jblackawwwhhhgrgwww
23:41.57LordScinawait the worst nopaste i ever used
23:42.00LordScinawa...
23:42.05LordScinawaon line 8 starts my extension
23:42.11jblackdd won't even wipe the flash clean becase of io/errors at the start of the disk.
23:42.21Kattyeww :<
23:42.38Kattyrma it!
23:42.54LordScinawathanx [TK]D-Fender !!
23:43.05LordScinawai saw after all this pastebin! :D
23:43.05jblackI'm gonna take it back and shove it down someone's throat but before I get into that fight, I want to make sure there's no data on it.
23:43.23Kattyi would low level format it
23:43.38*** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at)
23:43.43jblackwell I was trying to wipe it with dd, but that's giving out
23:44.11Kattyasks ryan
23:44.34jblackI'm about to do a loop of about 242000 dd's to do it one sector at a time
23:44.38Kattycleanwipe?
23:45.20jblackno such animal in ubuntu
23:45.20Kattyoh, true.
23:45.20Kattythe hiren's boot cd might have something useful on it
23:45.27Kattychecks apps
23:45.30jblackbesides. I don't need anything milspecish. I just want equivilant to a quick pile of 0s
23:45.50Kattyhdd regenerator might be able to recover bits of it
23:46.08Kattyhttp://www.hiren.info/pages/bootcd <- apps
23:46.40jblacknah. the stuff I didn't get back back is mostly cached files.
23:46.45Kattyk
23:46.54jblackI just want to make sure i don't leave a copy of phone numbers, addresses, etc.
23:47.04Kattykay
23:48.09Kattyi'm thinking about a bubble bath!
23:48.18Kattywith my laptop running netflix nearby
23:48.28carrarpics!!
23:48.37Katty-_^
23:48.43Kattyummmm...no.
23:48.55*** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194)
23:49.28LordScinawahttp://pastebin.com/d309266dc
23:49.34LordScinawathis is a better version
23:50.32[TK]D-FenderLordScinawa: [DLPN_DialPlan1]  <-  see your exten in here?  Skipping priorities like that is illegal <-
23:50.47[TK]D-FenderLordScinawa: when it fails to find #2 it will kill the call
23:50.56LordScinawaO_O
23:51.16LordScinawai was wandering that extension ._X match all the numbers like 0039xxxxx
23:51.18jblackKatty: durrhh.  cat /dev/zero > /dev/mmcblk0 seems to be doing the job just fine.
23:51.34[TK]D-FenderLordScinawa: its not just an order.  If your hearts skips enough beats you're dead too :)
23:51.47jblackwhy are the simplest tools always the best....
23:52.00*** join/#asterisk Sargun (n=Sargun@atarack/Staff/Sargun)
23:52.03LordScinawa:P lol
23:52.08[TK]D-FenderLordScinawa: And in your call file you aren't specifying the start priority either...
23:52.17LordScinawammm
23:52.20LordScinawain witch way?
23:52.34LordScinawaextension: 1
23:52.34LordScinawa?
23:52.49[TK]D-FenderLordScinawa: PRIORITY, not EXTENSION
23:53.13[TK]D-FenderLordScinawa: Separate parameter line
23:53.13LordScinawaContext: DLPN_DialPlan1
23:53.13LordScinawaExtension: 00393496557407
23:53.13LordScinawaPriority: 1
23:53.27[TK]D-FenderLordScinawa: Better... no fix your sequencing of that entire exten
23:53.30[TK]D-Fendernow*
23:53.49LordScinawamy sequencin
23:53.51LordScinawayou mean
23:53.53LordScinawa1,2,3,4
23:53.54LordScinawaand not
23:53.58LordScinawa1,3,5,7 ?
23:55.36[TK]D-FenderLordScinawa: Correct
23:56.07LordScinawaok
23:56.46saftsackrussellb, #  NEW: In 2009 Digium introduced the commercial "Fax for Asterisk" with T.38 support (free licence for 1 channel)
23:56.49saftsackhttp://www.voip-info.org/tiki-index.php?page=Asterisk%20T.38
23:57.04russellbinfobot: Fax for Asterisk
23:57.05infoboti guess fax for asterisk is http://docs.digium.com/FAX/fax_for_asterisk_admin_manual.pdf
23:58.50*** join/#asterisk Zokeber (n=Zokeber@unaffiliated/zokeber)
23:59.12LordScinawahttp://pastebin.com/m10db17bd
23:59.34LordScinawaprobabily i don't understood the problem
23:59.45LordScinawabecause it still didn't working

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