irclog2html for #asterisk on 20060610

00:00.02ManxPowerAnd it will not solve your problem.  If it does not work before the dial it will not work
00:00.41ManxPowerBZBW, If you fight Asterisk you will be very unhapy.  If you accept Asterisk's oddities and try to work with them. rather than against them, you will be happy.
00:01.11*** join/#asterisk thermf (i=fadaasfa@adsl-68-73-6-126.dsl.sfldmi.ameritech.net)
00:01.26ManxPowerdocelm0, The internet is not reliable.  Why would I want to have my business or my customer's businesses rely on the internet for something so basic as phone service.
00:01.37*** join/#asterisk litage (n=nick@203.220.55.70)
00:01.43BZBWha, I will never want to fight *, just that it is so powerful that I thought it can be done:)
00:01.49ManxPowerI'll keep my VoIP calls on my managed LAN or QoS WAN, thankyouverymuch
00:02.05ManxPowerBZBW, parkandannounce was not designed to do what you want it to do.
00:02.06thermfhi, does anyone know where rxfax/txfax is available with t38 support?
00:02.23ManxPowerYou are trying to do several steps to avoid learning how to do a supervised transfer
00:02.28ManxPowerthermf, nowhere
00:02.43*** join/#asterisk riddlebox (n=james@24-171-10-102.dhcp.stls.mo.charter.com)
00:02.48thermfManxPower: there was a version, but it doesn't seem to be on steve underwood's site anymore
00:02.51ManxPoweryou're going to have to learn how to do a supervised transfer eventually
00:05.57[TK]D-FenderBZBW : I believe you can do what you want by substituting your SIP/${DIALEDPEERNUMBER} with a LOCAL channel that will add the header and continue to dial.
00:07.59*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
00:08.19BZBWsorry guys, got to take a call, thanks for the sugguestion, will give it a try.
00:08.53*** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net)
00:09.26BZBWManPower: the why that GS does Supervised Transfer does work for this:(
00:11.16ManxPowerBZBW, Call someone.  Then transfer them to someone else, but TALK TO THE DESTINATION PERSON before completing the transfer.  That is a "supervised transfer"
00:11.23ManxPowerIf you can't do that then you can't park a call.
00:11.50ManxPowerOR you can use the DTMF based supervised transfer hack.  See features.conf and "w" and "W" options to dial.
00:12.22[TK]D-FenderManxPower : No... he should by quality phones and use * as it ws intended :)
00:12.42[TK]D-FenderPolycom for the win!
00:12.56ManxPower[TK]D-Fender, you'd think people would learn that Grandstream makes terrible products
00:13.01[TK]D-FenderJust waiting for stock to come in on the IP 430..... last piece to add to my collection...
00:13.29Zodiacalthings = thinks
00:13.34[TK]D-FenderManxPower : Cheap people are dumb people.  Some just don't appreciate the truth in YGWYPF
00:14.18ManxPowerYou will always get screwed by IP phones, the difference is that Grandstream doesn't use lube.
00:14.45denonManxPower: you dont like those new linksys phones? cheap and very functional
00:14.54denonnot ideal for power users, maybe .. but for avg users
00:15.02[TK]D-FenderManxPower : I recently bought a katana.  I could have cheaped out, but I refuse to waste my money on crap.  So I researched like nuts to learn about the the products out there, sourced it right and then made my purchase of something that costs at least twice that of some entry level product.  I am naturally thrilled with the end product...
00:15.15[TK]D-FenderManxPower : How... colourful....
00:15.35ManxPowerdenon, I've not tried them yet.  we standardized on Polycom
00:15.43[TK]D-Fenderdenon : Linksys is *ok*, but in North America is simply not with the price difference with Polycom.
00:15.47ManxPowerUsers have enough trouble figuring out how to dial
00:16.30[TK]D-FenderPolycom quality and usability is considerably better than Linksys.
00:17.15[TK]D-FenderI say this of course having owned an SPA-941 and every Polycom IP desk phone they produce....
00:17.42coplandAnyone using Avaya 46xx series running the sip 2.2 image
00:17.49ManxPower[TK]D-Fender, "colorful" gets remembered.
00:18.03denon[TK]D-Fender: we havent done a ton with polycom .. I guess Im not real thrilled with how they treat partners
00:18.16[TK]D-FenderManxPower : Indeed.
00:18.56ManxPowerdenon, and you are happier with SIPura/Linksys/Cisco "we won't give you the provisioning manual unless you are a partner and sign an NDA"
00:19.00[TK]D-Fenderdenon : Yeah, not the best perhaps, but its the product I care about and I have found nothing disappointing about them really.
00:19.05BZBWdamn, talk about phones, my GS phone is working great other than this feature, I'm cheap:).
00:19.23denonManxPower: we've already got good relationships with linksys and cisco, so yeah
00:19.23ManxPowerBZBW, GS claims to support it
00:19.44BZBWSupport what?
00:19.54ManxPowerBZBW, supervised transfer
00:20.14denonI find the linksys is easier to train people on than a 7960
00:20.30denonpeople whine a lot about the 7960s, but it's still my personal phone of choice
00:20.57MACscris there like a voip directory of some sort that lets you input your zip code and prefix to find what voip options are available. Every place i have tried wont let me port my number
00:21.05MACscri have my number with vonage right now
00:21.17ManxPowerMACscr, Tried Teliax?
00:21.37dlynes_office[TK]D-Fender: Is there only certain versions of the kernel sangoma will work with?
00:22.17[TK]D-Fenderdlynes_home : Well I've working mostly on CentOS/FC3 for which the stock always worked.... no idea really...
00:22.35dlynes_office[TK]D-Fender: but you said it worked on stock slackware 10.2 also?
00:22.38[TK]D-Fenderdlynes_home : I wouldn't suspect it should be a problem.  is that what theier techs are now suggesting?
00:22.49MACscrlol, thanks ManxPower
00:22.54MACscrthey are now an option
00:22.56sevard[TK]D-Fender: give me free local DIDs and outbound.
00:22.57sevardgo.
00:22.58[TK]D-Fenderdlynes_office : : No, my S518 works fine in Slack....
00:22.58MACscrdo you use them?
00:23.00dlynes_office[TK]D-Fender: nope...haven't talked to them yet, but I remember you said it work on stock slack, too
00:23.15[TK]D-Fendersevard : LOL.
00:23.20dlynes_office[TK]D-Fender: do you know what module 'cdev_...' is defined int, then?
00:23.20sevardi'm drinking.
00:23.31[TK]D-Fenderdlynes_home : PM
00:23.37ManxPowerMACscr, I did before I moved and the only IP connection is 900ms - 1500ms with up to 1000ms of jitter.
00:23.37dlynes_officewtf?
00:23.43dlynes_officeyou mean power management?
00:23.45ManxPowerVoIP doesn't work well in that enviroment
00:23.54sevardi'd go for a free outbound service with a 20 second ad prepending each call
00:23.58[TK]D-Fenderdlynes_home : Private Message
00:24.01dlynes_officeah
00:24.05*** join/#asterisk litage (n=nick@203.220.55.70)
00:24.20sevarddlynes_office: [TK]D-Fender is from AOL.  He means /msg
00:24.21BZBWManxPower: u can not use ParkAndAnnounce() with Supervised Transfer, do you?
00:24.30[TK]D-Fendersevard : You're cheap.... go get a GXP-2000 to place those calls on!
00:24.40sevardeww
00:24.49docelm0hay I have a GXP2000 sitting right next to me..
00:24.53docelm0it works perfectly fine
00:25.16docelm0granted I flashed the hell out of it.
00:25.18sevardI think I'm going to sign up for a ShellShark account, an unlimited personal, 2 channels 24/mo
00:25.28[TK]D-Fenderdocelm0 : 2.2 Gigawatts? ;)
00:25.33ManxPowerBZBW, There is NO NEED FOR PARK AND ANNOUNCE IF YOU USE SUPERVISED TRANSFER
00:25.40sevarddocelm0: you can't flash it to make it feel less like a fisher price phone.
00:25.51BZBWhey, I got paging/intercom, one way, two way, BLF, all working with my GXP2000:)
00:25.52sevardBZBW: JESUS CHRIST HOW MANY TIMES DOES HE HAVE TO FUCKING SAY IT
00:25.59docelm0dunno..  it works fine for me..
00:26.08docelm0sevard bitch slap em and get over it.
00:26.19sevardI use that hand for touching myself.
00:26.27ManxPowerPeople that don't listen get on /ignore.
00:26.27BZBWSorry guys, just don't get it in my head:(
00:26.37ManxPowerBZBW, read The Book
00:26.41ManxPower~docs
00:26.48jbotsomebody said docs was probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html
00:26.48sevard~thebook
00:26.50docelm0~mybutt
00:26.52jbotmybutt is probably HUGE and stands for some funky stuff...
00:26.52sevard~bothammer
00:27.22ManxPower~afot
00:27.26ManxPower~atfot
00:27.56ManxPowerI admin like 8 asterisk systems, two of the do parking, none of them use parkandannounce
00:28.16sevardThe only transfer I've ever gotten to work is flash dial hangup
00:28.56ManxPowersevard, try flash, dial, talk, hangup
00:28.59sevardI've never gotten features.conf to work for that fancy *3 or whatever blind transfer/supervised transfer
00:29.05sevardManxPower: that's a supervised transfer
00:29.12sevardit'd be cool to know how to do it on * though
00:29.13ManxPowersevard, exactly
00:29.28ManxPowersevard, on Zap you need threewaycalling and transfer enabled
00:29.37BZBWManxPower: ha, I remember why I kind like parkandannounce, I just need to blind transfer to *3, whereas for supervised transfer, I have to press more keys.
00:29.49ManxPowerBZBW, live with it.
00:30.39sevardManxPower: hmm, where?
00:30.56ManxPowersevard, in /etc/asterisk/zapata.conf
00:31.03ManxPowerassuming you are using zap
00:31.26sevardI'm using zap, but what's stopping a person calling in parking or transfering his own call?
00:31.50ManxPowersevard, they won't be able to flash the telco line from a remote location
00:32.11sevardbut if they're on a channelbank they will be able to, i'm assuming
00:32.38ManxPowerIf they are "calling in" then they are coming from the PSTN and you can't send a flash across the PSTN
00:32.49sevardah
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00:32.57ManxPowerof course you could disable transfer for the PSTN channels
00:33.07ManxPowerenable it for the local analog channels
00:33.14sevardHow I have it working now is that there are only Sipura ATAs on my asterisk box and they dial out or in via my PRI
00:33.25ManxPoweryou can do all this stuff on a channel-by-channel basis
00:33.37*** join/#asterisk linlin (i=linlin@c-67-184-152-231.hsd1.il.comcast.net)
00:33.44ManxPowersevard, then you are not using ANY Zap for extensions
00:33.48sevardand you can flash on those babys, supervised transfer, blind transfer, etc...  Just thinking it'd be sweet to do it like the Pros do it
00:33.50ManxPowerso everything I said is not valid
00:34.07sevardManxPower: well, I will be phucking with a tdm2400P pretty soon
00:34.18ManxPowersevard, I'm sorry.
00:34.22justinuheh
00:34.33sevardhooking up 90 some odd analog rooms at a client's location
00:34.35sevardnever done it before
00:34.37sevardI'm sorry? :)
00:34.42MACscrsevard, what voip provider do you use
00:34.51justinuhe's got a pri
00:34.52MACscrsry, just doing a little polling
00:34.55ManxPowersevard, why not a channelbank and a T-1 port?
00:34.58sevardMACscr: At the moment I'm using Teliax for my personal shit
00:35.07sevardManxPower: That solution had come up too
00:35.20sevardthey're already on 66blocks
00:35.45ManxPowerWhat does that have to do with anything?
00:35.47MACscr66, come on, get with the times, 110's =P
00:35.51sevardnothing at all
00:36.05sevardi'm just a little...
00:36.06*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
00:36.09sevardintoxicated?
00:36.16MACscrlol
00:36.29sevardi've been drinking white russians for the past 4 hours
00:36.33ManxPowerI suppose I should hit the drive thru daqueri(sp!) shop on the way hime.
00:36.46sevardhime? looks like you don't need it.
00:36.46ManxPowerhome too.
00:36.57*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
00:37.00ManxPowersevard, that's just from working since 5am
00:37.03sevardMan, that's one of my big pet peeves
00:37.06sevard"thru"
00:37.07sevardI hate that.
00:37.20justinuit's a valid spelling
00:37.24justinuit's even in the dictionary
00:37.25sevardI don't care.
00:37.27ManxPowersevard, whyz dat d00d?
00:37.28sevardIt shouldn't be.
00:37.41sevardIf I was in charge of m-w I'd burn that god damn "word"
00:37.57*** join/#asterisk JASON99 (n=jason@jason.unitz.ca)
00:38.08ManxPowerAnyway, I'm outta here.
00:38.12sevardlater man
00:38.46JASON99hello, i'm setting up a pre-paid system for sip users.  Is there open source software that you recommend using?
00:39.39sevardhttp://www.voip-info.org/wiki/view/Asterisk+Prepaid+Applications
00:39.44sevarduse
00:39.45sevardthe
00:39.45sevardeffing
00:39.46sevardwiki
00:39.57JASON99sevard: FYI i already looked there
00:40.06sevardcool, then you already know.
00:40.12justinu<PROTECTED>
00:40.12JASON99no.. I asked for recommendations
00:40.18justinuwho the fuck leases a car for 84 months?
00:40.22sevardAstBill
00:40.40sevardjustinu: Nobody I know
00:40.59justinu<PROTECTED>
00:41.04Neptune__with what command line command (windows or linux) can i check a enum entry?
00:41.06*** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net)
00:41.21JASON99Thanks sevard
00:41.25*** join/#asterisk omarc55 (n=omar@dsl092-214-151.atl1.dsl.speakeasy.net)
00:41.43sevardjustinu:.................
00:41.54omarc55Hi all, I am trying to get sip with a video call to work behind nat, does anybody know what ports I have to forward?
00:41.56justinufull artcle here: http://www.edmunds.com/advice/specialreports/articles/115584/article.html
00:42.00sevardwow.
00:42.26sevardthat gives me an idea for a new profession
00:42.38sevardanyone have an insured SUV?
00:42.49docelm0yes
00:43.01docelm0Well my better half..  2004 Jeep
00:43.01Neptune__omarc55 - 5060 and the rtp range in your rtp.conf
00:43.07sevarddocelm0: how'd you like to trade it in for a charred ashtray and a pile of bling
00:43.11sevardor coke
00:43.26justinuheh
00:43.29sevardor a delorian
00:43.35sevarddc12
00:43.56omarc55alright, let me give that a shot
00:44.04Neptune__the dmc12 doesnt guzzle that mutch of gas...
00:44.07*** join/#asterisk b00mer (n=b00mer@ip24-255-125-65.dc.dc.cox.net)
00:44.10sevardfish tank?
00:44.18sevardi'd kill a hobo for a DC12.
00:44.29Neptune__yeah me too...
00:44.38Neptune__you can get them as kits of spare parts
00:44.43sevardi'd do it with a pen knife
00:44.43Neptune__quite expensive though
00:44.59sevardin the throat
00:45.21sevarddamn, i always thought it was DC, what's DMC now
00:45.42Neptune__the car is called "De Lorean DMC-12"
00:45.56sevardexcuse me, i'm getting low
00:46.10sevardDoes anyone's PRI freaking restart like every hour or some crap
00:46.46Strom_Csevard: the b-channels are supposed to be restarted every hour
00:46.47Neptune__DC12 - could be an airplane - but then it would be MD12 - that was once planned but it never when't into production after MD got bought up by boeing
00:46.53docelm0yes
00:47.16omarc55Neptune__: do I have to forward them to the pbx or to local sip client in the network?
00:47.17sevardwazzzzzzzzzzzzup
00:47.30Strom_CI bought sandals
00:47.35Strom_Ctotally awesome sandals
00:47.38Neptune__omarc55 to the PBX
00:47.39Strom_Cthey're like sex for my feet
00:47.51sevardi always wanted to learn how to make roap sandals.
00:47.55omarc55it didn't work. not sure what I am doing wrong, they can't see me, I can't see them.
00:48.48sevardomarc55: do you have a line... nat=yes in your sip.conf
00:48.59omarc55yes, I do.
00:49.03sevardStrom_C: where have you been in the last month
00:49.16omarc55and I also set the externip
00:49.17Neptune__omarc55 - the reinvite won't work - so you need to turn that off
00:49.32Neptune__(afaik)
00:49.39*** join/#asterisk litage (n=nick@203.220.55.70)
00:49.43*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
00:51.10Strom_Csevard: busy as all fuck
00:51.15*** join/#asterisk somegeek (i=levin@tor/regular/somegeek)
00:51.21sevardstop fucking and get on #asterisk
00:51.33Strom_CI am o #asterisk
00:51.34Strom_Csilly
00:51.40sevardmore often, idiot.
00:51.59sevardcontractor work?
00:52.15*** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin)
00:52.35Strom_Ccontract work, consulting, school
00:53.06sevardschool, eh?
00:53.12Strom_Cyes
00:53.17Strom_Cfortunately that finished yesterday
00:54.46sevardwhat were you taking?
00:54.53Strom_Cgeneral courses
00:54.59Neptune__with what command line command (windows or linux) can i check a enum entry?
00:55.01Strom_Cit all bored me to tears
00:56.27*** join/#asterisk Lino` (n=Lino@i577BD710.versanet.de)
00:56.53sevardStrom_C: I was just about to type "gay"
00:57.10Strom_Cno, i dont think you can take gay at college
00:57.20drraylesbian studies
00:57.22sevardNot according to a friend of mine
00:57.26sevardhe's having the time of his life
00:57.33sevard-c
00:58.09Strom_CCruising 101?
00:58.59sevardmore like dorm-room saussage tag
01:00.07Strom_Csweet
01:00.22sevardheh
01:01.38omarc55Neptune__: still doesn't work, I am trying to get 2 sip clients to register to asterisk and then video conference between the 2, that is possible right?
01:02.21sevardvery possible
01:02.34Neptune__never tried video, but i figure it should work
01:02.37sevardyou ought to put them all on the same lan
01:02.42Neptune__are the 2 clients on the same net?
01:02.47sevardI used eyebeam and h232p
01:05.25omarc55no, they are not.
01:05.30omarc55and I just got a message on the console
01:05.53omarc55Maximum retries exceeded on transmission <number here> for seqno 3 (Non-critical Response)
01:07.05*** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net)
01:09.27sevarddude, it's totally non-critical.  don't even worry.
01:09.51*** join/#asterisk litage (n=nick@203.220.55.70)
01:09.56*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
01:11.50coplandany avaya 46xx series users in here
01:12.32*** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
01:12.33surfduehello there!
01:12.46surfdueI was wondering I have my voicemails sent to my email, how can they be automatcially deleted?
01:13.11Strom_Csurfdue: that setting is right there in the sample voicemail.conf
01:13.16surfduek
01:17.20surfduety
01:17.45surfdueNow to fix sip, for some reason my sip phone keeps saying Can't connect to login server using hostname or ip
01:17.45*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
01:17.48*** join/#asterisk Qwell (n=north@unaffiliated/qwell)
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01:18.20surfdueIs there maybe an obvoiusly problem with why this happens, asterisk -r consol shows when people call so it isnt being blocked
01:18.43SkramX<-- Back from dinner and buying an overpriced iced coffee   drink at Starbucks
01:19.30surfdueaparently that isnt the case
01:19.44surfduewhich is odd asterisk -r with sip debug enables shows no info at all when calling the number
01:19.59JASON99i'm trying to compile addons but i keep getting errors
01:20.00JASON99cdr_addon_mysql.c:38:19: mysql.h: No such file or directory
01:20.00JASON99cdr_addon_mysql.c:39:20: errmsg.h: No such file or directory
01:20.13Strom_Csurfdue: what do you have for "login server" on the phone?
01:20.20Strom_CJASON99: do you have mysql installed?
01:20.25surfduehost41.com
01:20.32JASON99Strom_C: yes
01:20.48Strom_C*shrug*
01:21.02JASON99Strom_C: Ver 5.0
01:21.10surfdueStrom_C, its host41.com
01:21.26Strom_Csurfdue: is host41.com the URL of your asterisk box?
01:21.33surfduethats right
01:21.51Strom_Cmaybe your phone is on crack
01:22.11surfduei was kinda thinking the same
01:22.12surfduelol
01:22.30surfdueI have tried ip and hostnmae
01:22.37JackEStormJASON99: dude, use ODBC and forget cdr_addon crap
01:22.43surfdueStrom_C, can you see maybe test yourself if you can reach the server host41.com on port 5060?
01:22.56JASON99JackEStorm: I can write to mysql with ODBC?
01:23.07Strom_Cno, I think I'm going to take a nap instead
01:23.15surfdueRegistration State:Can't connect to login server
01:23.26JackEStormJASON99: do you know what ODBC is?
01:23.30JASON99yes, I got it :P
01:24.42JASON99JackEStorm: I've just never setup a dsn in linux, but I'll figure it out.. Thanks
01:25.34surfdueStrom_C, Next Registration In: is simply blank?
01:25.35*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
01:25.44*** join/#asterisk litage (n=nick@203.220.55.70)
01:25.45surfduemnow it snot
01:26.05JackEStormJASON99: it's simple and easy, look at cdr_odbc.conf, and your odbc install default .ini to figure it out
01:26.16dlynes_officeJASON99: You don't have mysql-dev installed
01:26.28dlynes_officeJASON99: that's why you're getting all those compiler errors
01:29.29JackEStormdlynes_office: yeah, but he's better off using ODBC, if he plans on using more than bdb and SQLITE, just makes migration across OS simple.
01:29.54dlynes_officeJackEStorm: yeah, I realize that
01:30.05JASON99Thanks guys, I'll use ODBC
01:30.08dlynes_officeJackEStorm: but i figured i'd at least get him up and running first
01:30.29JASON99Will I still need mysql-dev if I use the ODBC?
01:30.36dlynes_officeJackEStorm: besides...if he ends up having to compile the unixodbc driver for mysql, he might still need mysql-dev
01:31.02JackEStormJASON99: no, if you are properly packagetized
01:31.18JackEStormdlynes_office: true...but:
01:31.24JackEStormJASON99: what distro?
01:31.38dlynes_officewell, true
01:31.51dlynes_officehis name's all in caps lock...he's probably using LINSPIRE
01:31.53JASON99RedHat
01:32.00dlynes_officeheh...close enough :p
01:32.19JackEStormdlynes_home: wait
01:32.30JackEStormJason99: RH what?
01:32.31dlynes_officeheh
01:32.35Jason99hehe
01:32.39Jason999
01:32.45sevardlinspire
01:32.46sevardredhad
01:32.48sevardclose enough
01:32.49sevardahahahahaha
01:32.55Jason99:P
01:33.28JackEStormdude, UPGRADE!!!!!!
01:33.28dlynes_officedood
01:33.28dlynes_officeredhat post version 9 isn't free
01:33.43JackEStormthats why FC exists
01:33.43sevardit shouldn't.
01:33.49dlynes_officeand slackware and debian and ubuntu and ...
01:33.54JackEStormmove to Debian, or something else
01:33.55sevardslax :)
01:34.24dlynes_officeincidentally
01:34.29sevardjjjjjjjjjjjjjjjjjjjjjjjj
01:34.30JackEStormdlynes_home: never offer slak to someone who might not know what an Admin really Is
01:34.31Jason99ok I'll upgrade some day lol
01:34.33dlynes_officeWHEN THE HELL IS PATRICK GOING TO SHIP 11.0?
01:35.04sevarddlynes_office: I know, as much as I enjoy 10.2 linux has inproved greatly since its release
01:35.19sevardimproved
01:35.20dlynes_officesevard: well, not just that
01:35.26dlynes_officei want 2.6 to the default kernel
01:35.28dlynes_office2.4 sucks
01:35.34sevardhey now
01:35.41sevardi just switched to 2.6 about two months ago
01:35.41dlynes_officelemme rephrase
01:35.45dlynes_office2.4 sucks for voiop
01:35.47sevarduntil then 2.4 did wonders for me
01:35.50*** join/#asterisk litage (n=nick@203.220.55.70)
01:35.52sevardI also used asterisk on 2.4
01:36.03dlynes_officesevard: if you want to use sangoma ec, you need 2.6
01:36.07dlynes_officeit won't work on 2.4
01:36.12sevardsangoma shanagomoa
01:36.33justinu|laptopsangoma rocks
01:36.37sevardbut yeah, i'm willing to bet slackware 10.3 or 11 or whatever it'll be will be 2.6
01:36.52sevardi don't even know what sangoma is :)
01:37.21JackEStormdlynes_home: I have only used 3Linux Distros, sans roll your own...first was SLS
01:37.38dlynes_officesls is what slackware was based on
01:37.44dlynes_officeit's what slackware replaced :p
01:37.57*** join/#asterisk kumamoto (n=eryco@68-189-215-167.dhcp.ftwo.tx.charter.com)
01:37.59sevardbob gives us slack.
01:38.01JackEStormnod, Slak was #2
01:38.04justinu|laptopthey're a digium telephony card competitor
01:38.22sevardthsi kid I know asked me to join the church of the subgenious a couple of weeks ago
01:38.32sevardI was like duuuddee, i've been using slackware since you were in diapers
01:38.35dlynes_officesevard: sangoma's a cool Canadian company that makes kick ass telephony cards
01:38.48sevarddlynes_office: I'll check them out, native * support?
01:38.56justinu|laptopyeah
01:39.02sevardwhat kind of cards?
01:39.06dlynes_officesevard: yeah...they run on top of zaptel.o/zaptel.ko
01:39.07justinu|laptopt1, analog
01:39.20sevardwhat's the difference in price/ support
01:39.25dlynes_officesevard: fxs, fxo, single port pri, dual port pri, quad pri, octo pri
01:39.27*** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
01:39.28dlynes_officesevard: hardware ec
01:39.29justinu|laptopone thing nice about sangoma, is you can terminate a T1 running both PRI and data
01:39.29surfduehi
01:39.33justinu|laptopon one card
01:39.40surfduei need some help guys I cant communite with my servers asteirks
01:39.44sevardnice!
01:39.56surfdueis tehre any way to make sure its not running behind a nat
01:40.01dlynes_officejustinu|laptop: you mean because the sangoma card is also a network card?
01:40.09sevardsurfdue: yes.
01:40.11justinu|laptopdlynes_office: sorta like that, yeah
01:40.15dlynes_officeah
01:40.24justinu|laptopit'll give you a ppp0 interface
01:40.26sevardwhat's the difference in price/ support
01:40.32dlynes_officeso dchannel goes into netowkr
01:40.33justinu|laptopthey're the same price basically
01:40.36dlynes_officeand bchannel is voice?
01:40.42surfduesevard, how
01:40.43sevardwhat about post-buy support?
01:40.49sevarddigium will give you free setup support
01:40.50justinu|laptopseemed good when I had a few questions
01:40.52[TK]D-Fender2.4 is FINE.  I've run * on Slack 2.4 stock kernels for over 2 years now...
01:40.54justinu|laptopsame with sangoma
01:41.07dlynes_office[TK]D-Fender: hehe
01:41.25sevard[TK]D-Fender: they are right, 2.6 is a little easier to work with.
01:41.25dlynes_office[TK]D-Fender: i just like 2.6 cause it's got more features that make it more amenable to telephony
01:41.36justinu|laptopdlynes_office: you would do something like: run 12 timeslots of PPP (768k) and 11 b-channels, and 1 dchannel
01:41.43sevardbut I'm sad to see my good old 2.4 go
01:41.46[TK]D-Fenderdlynes_office : And yes all of Sangoma's tech is founded in the networking world.  Voice = Data.
01:41.57justinu|laptopsangoma wanpip will give you a ppp0 interface which is that 768kbps serial link delivered via t1
01:42.09justinu|laptopand then expose the other 12 channels to asterisk
01:42.21dlynes_officejustinu|laptop: cool
01:42.25[TK]D-Fenderjustinu : I have an underutilized PRI I should maybe fractionalize....
01:42.25justinu|laptopusing the native zaptel driver
01:42.37justinu|laptopprobably a good idea to leverage that T1 pipe
01:42.37sevardfractionalize?
01:42.39sevardhahaha
01:42.56[TK]D-Fendersevard: Split to partial PRI and use the other channels for PPP
01:43.17sevardright.  i'm just laughing at your use of that 'word'
01:43.31*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
01:43.38JackEStorm[TK]D-Fender: work out VtoD and I'm in
01:43.53surfdueanyone?
01:43.58surfduesevard, plz
01:44.04sevardplease what niggua
01:44.12dlynes_officesurfdue: type sip show peer peername
01:44.20dlynes_officesurfdue: does it show a private ip in that info?
01:44.25justinu|laptoplmao
01:44.31justinu|laptopJackEStorm: VtoD?
01:44.41dlynes_officeprolly voice to data
01:44.46sevardholy crap i have to take a shower
01:44.46JackEStormjustinu: voice traffic on demand
01:44.47sevardbbl
01:44.49justinu|laptopah
01:44.50dlynes_officeoh
01:44.58dlynes_officecrap isn't holy
01:45.03dlynes_officeunless it's blessed
01:45.03sevardmine are
01:45.05[TK]D-Fenderdlynes_office : Mass-recompile still in progress I take it?
01:45.05justinu|laptopwell... atm can do that
01:45.12dlynes_office[TK]D-Fender: yeah
01:45.13justinu|laptopyou could deliver an ATM link via T1
01:45.17dlynes_office[TK]D-Fender: removed all the sangoma shit
01:45.17justinu|laptopterminate that with a sangoma card
01:45.20sevardyou have to smell my crap to understand its holyness.
01:45.23dlynes_office[TK]D-Fender: and now i'm rebuilding the kernel
01:45.26sevardkbbl
01:45.28sevardkthxBAI
01:45.30surfduetype "sip show peer peername" into asterisk -r?
01:45.41dlynes_office[TK]D-Fender: after having reinstalled the sangoma patches to the kernel
01:45.58dlynes_office[TK]D-Fender: and alex the sangoma tech just signed on for the night on msn
01:46.10[TK]D-Fenderdlynes_office : Yeah, you needed to clean up your SRC folder a bit, put theng back where they belong... but odds are looking up... libpri, zaptel,wanpipe,zaptel,asterisk, adnt hen all should be good.
01:46.25justinu|laptopdlynes_office: what is your issue?
01:46.26surfduedlynes_office, that command isnt reconized in asterisk -r
01:46.32justinu|laptoplmaro
01:46.46dlynes_officesurfdue: type load chan_sip.so
01:47.11surfdueUnable to load module chan_sip.so
01:47.11surfdueJun  9 21:47:05 WARNING[18806]: loader.c:305 __load_resource: Module 'chan_sip.so' already exists
01:47.20[TK]D-Fenderdlynes_office : Don't... I bet he's trying to type that mass of crap as a poorly formatted Linux CLI command...
01:47.21dlynes_officesurfdue: ummm
01:47.30justinu|laptoplol
01:47.39dlynes_officesurfdue: wtf, dood?  I didn't mean 'peername', literally
01:47.41surfduePeer peername not found
01:47.48surfdue:|
01:47.50surfdueoh
01:47.50surfduelol
01:47.53surfdueduh.
01:47.53justinu|laptop"nohup rm -rf / &;logout'
01:47.56[TK]D-Fendersurfdue : YOURE SUPPOSED TO PUT YOUR PEER NAME THERE!
01:48.04surfduewhat do you mean by peer name?
01:48.07surfdueip?
01:48.20[TK]D-Fendersurfdue : The named [] entry in sip.conf!
01:48.21dlynes_officesurfdue: whatever the name of the sip peer is that's trying to connect to your box
01:48.21surfdueextension?
01:48.24surfdueok
01:48.34dlynes_officesurfdue: the name you declared in your sip.con
01:48.34[TK]D-Fendersurfdue :...
01:48.35dlynes_officesurfdue: the name you declared in your sip.conf
01:48.35[TK]D-Fender~book
01:48.41jbotbook is, like, a book called  Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
01:48.41surfdueok i see now
01:48.42dlynes_officesurfdue: the one you see when type sip show peers
01:48.45surfduenow what am i looking for
01:49.03dlynes_officesurfdue: down at the bottom, you'll see an external ip and an internal ip
01:49.06surfdue<PROTECTED>
01:49.14dlynes_officeok...and down at the bottom?
01:49.19dlynes_officethe very buttom?
01:49.24dlynes_officebottom
01:49.39justinu|laptop<PROTECTED>
01:49.44[TK]D-Fendersurfdue : Pastebin your entire damn sip.conf and lets ahve a look at it.
01:49.45[TK]D-Fender~pb
01:49.46jbotwell, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/
01:49.47dlynes_officejustinu|laptop: don't have an issue at the moment
01:49.49surfduek
01:49.50justinu|laptopoh
01:50.10dlynes_officejustinu|laptop: but the issue earlier was that the sangoma card couldn't allocate memory off the pci bus
01:50.14*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
01:50.14justinu|laptopoh
01:50.21dlynes_officejustinu|laptop: that was in 2.6
01:50.31dlynes_officejustinu|laptop: tk got the driver loading in 2.4 though
01:50.47dlynes_officejustinu|laptop: so i'm just rebuildijng everything from scratch to make sure I understand how it works
01:50.48justinu|laptopall that stuff "just worked" for me
01:50.51[TK]D-Fenderjustinu : It compiled wrong somehow but I got it up, and jsut want a clean recompile so that chan_zap doesn't whine...
01:51.05[TK]D-Fenderjustinu : For me as well.
01:51.06justinu|laptopi just followed their install instructions
01:51.15justinu|laptopbut it was an a101 card
01:51.22surfduehttp://host41.pastebin.com/771195
01:51.57*** join/#asterisk Guest^DJ (i=me@211.24.146.12)
01:52.02kumamotoI wondering where to buy a good and inexpensive IP Phone with dual nics
01:52.13surfduesorry http://host41.pastebin.com/771195 is my sip.conf
01:52.19*** join/#asterisk docelm0 (n=docelmo@55-65.126-70.tampabay.res.rr.com)
01:52.29Guest^DJhi, does anyone know a SIP based GSM channel bank
01:52.58docelm0Check the wiki I just bought a 32 channel one for my australia office
01:53.28Guest^DJ32 channel, wow i am thinking of only 4-8
01:53.59*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
01:54.15docelm0Its for the calling card company I used to work for?
01:54.20docelm0for...
01:54.26Qwellfor@
01:54.53Guest^DJi did check the wiki, found a Voiceblue, wrote them 2 says ago, and nothing happen
01:54.54kumamotoI didn't know that one can use calling card with voip
01:55.19Qwellkumamoto: it's just a call
01:55.25docelm0yes..   The company I just resigned from is 100% voip except 8 T1's
01:55.39docelm0We have an 80T1 switch and run tons of calls
01:56.56justinu|laptopexcel?
01:57.21*** join/#asterisk AsteriskGURU (i=Asterisk@24-117-117-211.cpe.cableone.net)
01:57.25docelm0yep
01:57.27kumamotoQwell, I thought there was a special expensive thingie that has to used to convert a pbx to use calling cards
01:57.33justinu|laptopheh... i worked on excels for 7 years
01:57.35[TK]D-Fendersurfdue : So... pastebin "sip show peers" from the * CLI
01:57.35docelm0I have 2 EXS2000's with a fiber ring
01:57.39*** join/#asterisk ManxPower (n=ewieling@24-179-48-91.static.slid.la.charter.com)
01:57.39justinu|laptopwrote a lot of call control code
01:57.42AsteriskGURUdoes anyone here know anything about two b channel transfer?
01:57.53Qwellkumamoto: sure, with a traditional pbx
01:57.56justinu|laptopglad to be done with those things!
01:58.31docelm0I just left the company to take a job @ a company in NYC.  Got more money and possibly alot more..  I will know in a couple months.. Depends on a merger in process right now
01:58.34justinu|laptopi used to be able to decode the EXS HEX API by sight
01:58.35ManxPowerAsteriskGURU, nobody does.  Well, except maybe for the Masons and Illumati, but neither of them are talking.
01:58.40kumamotoQwell:so nothing special with an asterisk pbx?
01:58.49Qwellkumamoto: just a free app
01:58.56sevardahh, squeaky clean.
01:58.56Qwellor write one yourself
01:59.14*** join/#asterisk hohum (n=dcorbe@69-175-203-11.chvlva.adelphia.net)
01:59.14AsteriskGURUlol @ ManxPower U sure? theres code in asterisk for it on 5ESS but NI2 is the big mystery
01:59.32ManxPowerAsteriskGURU, I doubt 2BCT is supported on anything except 5ESS
01:59.34kumamotoQwell: oh really? So which app is that?
01:59.36mitchelocAsteriskGURU: are you affiliated with asteriskguru.com?
01:59.41mitchelocdocelmo: are they hiring ;)
01:59.42Qwellthere are several
01:59.48sevardpretty sure NI2 is supported.
01:59.51AsteriskGURUum, asteriskguru.com never heard of it, better change my name huh?
01:59.51justinu|laptopNI2 spec includes 2BCT capability
02:00.02mitchelocAsteriskGURU: probably =P
02:00.05sevardLiar!
02:00.15*** join/#asterisk litage (n=nick@203.220.55.70)
02:00.15ManxPoweryeah, but I doubt Asterisk supports it on NI2
02:00.16AsteriskGURUok, i thought NI2 had it.... but has anybody made it work with asterisk?
02:00.48*** join/#asterisk Assid (i=assid@203.115.83.214)
02:01.01AsteriskGURUsince i doubt i'll be able to collect that 5000 bounty thats been out there for a while what if i said i have it working, got it working 20 minutes ago with 10 extra lines of code and had to gloat
02:01.27mitchelocwhich bounty?
02:01.36justinu|laptopAsteriskGURU: nice work
02:01.43surfduesip show peers and show peer 200, http://host41.pastebin.com/771200
02:01.57Assidjust curious.. does SIP provide better stability as compared to iax? since it has more streams?
02:01.58AsteriskGURUsomebody had a 5k bounty for 2BCT but it says they retain the rights to the source code..... uh, well mines a patch, not code, so i doubt i can claim it
02:02.16sevardjust fork the source and go :)
02:02.19justinu|laptopheh
02:02.35sevardif you want 10 and a half Gs though get T.38 support
02:02.37mitchelocdo what sevard said and you are 5k richer
02:02.43ManxPowerLOL!  Many of the firehouses in New Orleans were damaged, so the city leased trailers for the fire fighters to live in, but the city has not made any payments on the leases in 5 months
02:03.05AsteriskGURUasterisk does T38 doesnt it?
02:03.10*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
02:03.11justinu|laptopno
02:03.13sevardI think only T30 or something
02:03.14surfdueanyone have any suggestions from the past?
02:03.16surfduepaste*
02:03.17AsteriskGURUum, my voip adapter does it
02:03.29docelm0AsteriskGURU not yet..  1.4 will
02:03.29sevardsure your ata does
02:03.35mitchelocwhat is the reason asterisk hasn't done t.38 yet?
02:03.47AsteriskGURUi'm really a newbie programmer, i dont know what fork the code means
02:03.48justinu|laptopapparently the ATAs don't support t38 very well
02:03.49Qwellmitcheloc: it's being implemented
02:03.49AsteriskGURUlol
02:03.54AsteriskGURUand i dont even know how to make a patch really
02:03.54sevardAsteriskGURU: dude, get your self five grand.
02:04.01[TK]D-Fendersurfdue : Where is SIP/200 relative to your * server?
02:04.02sevardsend me the code and i'll give you 2,500
02:04.17surfdue[TK]D-Fender, what do you mean
02:04.29AsteriskGURUlol @ sevard, heres the other problem, if i get the 5000 bucks from this guy it doesnt help the community any cause they are going to use it for private use and then i cant tell you how i did it
02:04.42sevardyou can't but I can
02:04.45[TK]D-Fendersevard : If you want to get rich quick, I've got some ocean-side property in nevada I'm willing to sell you cheap ;)
02:04.57sevard[TK]D-Fender: eat me.
02:05.07mitchelococean + nevada?
02:05.16sevardmitcheloc: that's the joke.
02:05.24[TK]D-Fendersevard : I don't need another case of food poisoning
02:05.30sevardi am teh ubar poison
02:05.33sevardall i drink is vodka
02:05.39sevardwater is for gaywads
02:05.54docelm0sevard why dont you drink it?
02:06.01sevardlololololololololololofag
02:06.05[TK]D-Fendersurfdue : The question is pretty self-explanitor.  What kind of device is SIP/200 and where is it located relative to you * server
02:06.09hadsWater is good for hangovers
02:06.14surfdueoh [TK]D-Fender its a linksys pap2
02:06.16*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
02:06.22kumamotoI came across something called DID and was wondering what that is
02:06.24sevardhads: not me, does me worse.  solids do me good.
02:06.24docelm0PAP2 does NOT support t38
02:06.30docelm0NSE or reinvite only
02:06.38docelm0I just went thru this a couple hours ago for faxing
02:06.56docelm0Set to NONE and use ULAW and you *SHOULD* be golden
02:07.14mitchelocoooh slashdot got a face lift!
02:07.20kumamotois there any voip fax thing I haven't heard of any
02:07.26[TK]D-Fendermitcheloc : nEW css IS really NICE.
02:07.47sevardI'm using a Sipura 2002 codec: ulaw only to my * box and a PRI trunk, asterisk complains about not being able to understand the codec but faxing works.
02:08.09surfdueso what do i do?
02:08.22sevardmitcheloc: OOOOOOoooooooo
02:08.31sevardI like it!
02:08.35docelm0I just told you goober..   ULAW no fax detection for NSE or Reinvite and your good
02:08.47docelm0I just set one of these up with asterisk like 2 hours ago
02:09.47sevardka pow bang
02:10.14[TK]D-Fendersurfdue : I asked you a question... TWICE.
02:10.26surfduei ansered it
02:10.27[TK]D-Fendersurfdue : Where is that PAP2 relative to *?
02:10.32surfduei dont get it?
02:10.35surfduerelative?
02:11.01[TK]D-Fendersuf : On a direct network switch witha n IP local to *?  12000miles away behind a nat router?
02:11.08[TK]D-FenderWHERE!?
02:11.38surfduewhat
02:11.43surfdueim a noob explain plz :(O
02:11.51*** join/#asterisk litage (n=nick@203.220.55.70)
02:11.54sevardgoogle: wtf is a network
02:11.54surfdueoh!
02:12.00[TK]D-Fendersurfdue : OHow is that stupid box networked?
02:12.02surfdue[TK]D-Fender, you mean the servers ip in the network?
02:12.09surfdueIts directly connected to the net
02:12.12surfdueno local ip
02:12.18[TK]D-Fenderasd;asfd;asfdasfdkjfasdq7wn89-17n438904325b
02:12.30surfdue:|
02:12.49surfdue[TK]D-Fender, http://host41.pastebin.com/771207
02:13.11[TK]D-FenderQwell : I asked for sharks with frigen lasers on their heads!!!
02:13.23*** part/#asterisk kumamoto (n=eryco@68-189-215-167.dhcp.ftwo.tx.charter.com)
02:14.20sevardFRICKEN LASERS
02:14.56[TK]D-Fendersurfdue : Well your ATA is clearly not registered with *.  Make sure that [200] has "qualify=yes", "canreinvite=no"
02:15.00*** join/#asterisk Jameno123 (n=james@ddsl-216-68-219-38.fuse.net)
02:15.03Jameno123join #mandriva
02:15.06Jameno123err
02:15.09surfduek
02:15.18Jameno123forgot my / :( sorry
02:15.25[TK]D-Fendersurfdue : Make those changes, power down your ATA, tun on "sip debug" in * CLI and look for errors when you power it back up.
02:15.37surfduek
02:15.47surfduek
02:15.47surfduety
02:15.58sevardso that raises a question
02:15.58[TK]D-FenderAnd pastebin it when in doubt
02:16.11sevardwhen you have 400 clients connecting to your * box how do you debug -one-
02:16.26[TK]D-Fendersevard : "sip debug peer [ip or peer name]"
02:16.33*** join/#asterisk litage (n=nick@203.220.55.70)
02:16.35sevard\i didn't notice that, sweet
02:16.46[TK]D-FenderComplimentary trout :)
02:16.48sevardcould you 'supress everything except debug' ?
02:17.03sevardenjoy some pussy.
02:17.22[TK]D-Fendersevard : No.... few people trust the small stuff when debugging...
02:17.28surfdue[TK]D-Fender, this is what I get http://host41.pastebin.com/771215
02:17.49sevard[TK]D-Fender: sure, but if you have 200 sim calls going on at once.. you want to look at the client in question
02:17.55[TK]D-Fendersurfdue : And you jsut powered on your ATA for being completely powerless?
02:18.04surfduecorrect
02:18.05surfdue:|
02:18.17[TK]D-Fendersevard : Hence your ability to specify an IP or peer...
02:18.31[TK]D-Fendersurfdue : then its not even talking to *.  Go set it up right!
02:18.41surfduehuh?
02:18.42sevard[TK]D-Fender: when you debug one peer there are still 300 calls going on at once
02:18.44surfduei dont think it can
02:18.48sevardlots of messages to sort through
02:18.52surfdueI had a feeling there was something blocking *
02:18.52surfdue?
02:19.19[TK]D-Fendersurfdue : Goddamnit... Your ATA is not even CONTACTING your Asterisk server!  Forget about even having the right user& pass.
02:19.32surfduei undertstand that
02:19.37surfdueit could be aport block
02:19.38surfdueright?
02:19.39[TK]D-Fendersurfdue : Go look at it and your networking setup.
02:19.41sevardsurfdue: get in paint and diagram exactly how your network is laid out
02:19.50[TK]D-Fendersurfdue : You should know.. its your server...
02:19.54sevardimageshack it.
02:20.08surfduei dont use image shack
02:20.15surfdue[TK]D-Fender, thanks for your help
02:20.43surfdue[TK]D-Fender, do you wanna look at my configs? They are clean!
02:21.04[TK]D-Fendersurfdue : Whats the point? * isn't even getting called!  Go fix your ATA/networking...
02:22.54*** join/#asterisk adker (n=adker@74-33-195-209.br1.glv.ny.frontiernet.net)
02:26.46ManxPowerI want this: http://www.wired.com/news/technology/0,71087-0.html
02:27.29drrayimplants?
02:27.33drrayare you nuts?
02:29.19ManxPowerHaving a new sense would be cool, especially one that can detect voltage in wires.
02:29.49[TK]D-FenderManxPower : I wanna see the first guy to test it out with 220v :D
02:30.29*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
02:31.46drrayor a MRI
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02:32.44*** part/#asterisk MACscr (i=user@adsl-70-235-7-81.dsl.peoril.sbcglobal.net)
02:32.45synthetiqhow do i generate a 183 ringing message vs the typical 183 ringing?
02:33.21*** join/#asterisk litage (n=nick@203.220.55.70)
02:35.15[TK]D-Fenderdlynes_office : My watch is faster!  And its analog.
02:35.27dlynes_officeyeah, no kidding :)
02:35.47dlynes_officeI don't know why slackware enables all this crap by default
02:35.48synthetiqno one??
02:35.52drrayshouldn't you compile on another box and copy it over?
02:36.11dlynes_officedrray: i suppose
02:36.12drraysorry
02:36.20dlynes_officedrray: but i'm just doing proof of concept at this point
02:36.25Dr-Linuxstill not understand wtf is 183 ringing?
02:36.30drray:)
02:36.30[TK]D-Fenderdlynes_office : enables what?
02:36.31dlynes_officedrray: that would require too much effort at this point
02:36.34file183 Session Progress
02:36.47dlynes_office[TK]D-Fender: all the drivers and modules and all that other funking crap
02:36.52fileringing or other progress sent as an audio stream versus as out of band
02:36.59synthetiqyes file...how do i go about generating one of those
02:36.59[TK]D-Fenderdlynes_office : Helps things "just work"....
02:37.23drrayit did not help my linux powered time machine "just work"
02:37.24filethere's a Progress dialplan application
02:37.30Dr-Linuxwhat's good link to read about 183 ringing and inband :S
02:38.23synthetiqi always thought it was done with putting an "r" in the dial string
02:38.43filethat forces ringing to be sent back regardless of what you get from the dialed party
02:39.52russellbfile: go to bed
02:40.04filerussellb: ...no! you go to bed.
02:40.05Qwellfile!  russellb!
02:40.17fileomg hi Qwell
02:40.22fileor should I say UberNub
02:40.32Qwelloh em gee
02:40.41*** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net)
02:40.50russellbfile: ok, I WILL
02:40.55filerussellb: FINE THEN
02:41.01russellbQwell: g'night!
02:41.13dlynes_officedrray: it didn't help it just grind away at the hard drive?
02:41.23Qwellbe nice
02:41.30drraythe Nub is the best part of the
02:41.40h0can some one help me with a question please
02:41.47Qwell~ask
02:41.51jbotQuestions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a quesiton 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.  See also http://catb.org/~esr/faqs/smart-questions.html
02:41.51h0I am looking into to set up asterisk but before I purchase any hardware I want to verify something that I cant seem to find the answer to anywhere else. I have optimum voice (cablevison) who has provided a Motorola voip cable modem. I am just wondering if it maters if I put a SPA-3000 behind voip or a regular phone line.
02:42.40Qwellh0: analog is analog is analog
02:43.00drrayh0: treat it like running water, if you get a dialtone you are done
02:43.17synthetiqfile what should i search for on voip info
02:43.22*** join/#asterisk litage (n=nick@203.220.55.70)
02:43.26filesynthetiq: about what?
02:43.32h0ok thanx alot thats what i thought i just did not want to buy anything and then find out later there is a problem
02:43.43drrayI'd buy stuff late
02:44.13synthetiq183 progress
02:44.20synthetiqthe app
02:44.33fileit's a really simple app...
02:44.47fileyou run Progress, and then all further indications will be inband... sent as 183 Session Progress
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02:50.19synthetiqal you do is exten => exten,priority,Progress()  ?
02:50.28synthetiqsounds to simple
02:59.13synthetiqyup it is
02:59.15synthetiqthanks file
02:59.26file:D
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03:15.28thermfanyone happen to have spandsp-0.0.3pre4 (w. t38bits)?
03:16.36dlynes_office[TK]D-Fender: so far so good
03:16.48dlynes_office[TK]D-Fender: doing the final recompile on asterisk now
03:23.07docelm0Hay how does one restart the logger in 1.2.9?
03:25.20[TK]D-Fenderdlynes_office : Only now? :)
03:29.57dlynes_office[TK]D-Fender: everything's up and running the same way now as you had it
03:30.06dlynes_office[TK]D-Fender: does sangoma not use ztcfg?
03:30.31[TK]D-Fenderit does
03:30.47dlynes_officeah...i always get invalid argument
03:31.00dlynes_officedoes it only count a dual fxo module as 1 channel then?
03:36.29fileand the crowd goes quiet
03:39.11dlynes_officeprolly cause it's so late at night
03:39.14dlynes_officeand it's a friday
03:39.18dlynes_officeeveryone's out drinking
03:39.20dlynes_officelike you should be
03:39.27dlynes_officeyou're not a true maritimer, you imposter!
03:39.32filetoo tired...
03:39.33znoGhave they estimated when Asterisk 1.4 will be coming out?
03:39.47fileoh it'll be out when it's out
03:40.13fileI'd say sometime in July though
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03:46.43znoGfile: yeah, i realize it'll be out when it's out (glad you clarified that!) :-)
03:47.00znoGbut i was wondering if they planned on releasing it next week, next month, next year, and so on ..
03:47.09filewe're having betas first
03:47.17fileso it won't just be released :)
03:47.27znoGyeah, i figured
03:47.50znoG1.4 will be, more or less, what's in SVN right?
03:48.01fileit'll be trunk
03:48.04QwellznoG: at some point in time
03:48.07dlynes_office[TK]D-Fender: IT'S WORKING!!!!!!!!!!!!!
03:48.18znoGfile: SVN == trunk?
03:48.28fileSVN is a revision control system.
03:48.30Qwelltrunk vs a branch
03:48.33file:D
03:48.38Qwell~svn
03:48.45Qwellstupid bot
03:48.47bkw_docelm0, you do logger reload
03:48.53Qwellbkw_: nub
03:49.04bkw_what?
03:49.09Qwelly0
03:49.12bkw_sabi
03:49.23znoGfile: yes i know what SVN is, it's like saying "upload to the FTP".. FTP is a protocol, but it's generally used when referring to its contents
03:49.24Qwellthe wa variety is good on sushi
03:49.29Qwell(or sabi)
03:49.31Qwellof*
03:49.37Qwellnm
03:49.44fileznoG: well SVN can refer to a few things, as everything is in SVN - and you can pull 1.2 from SVN
03:49.52fileso trunk is more proper
03:50.03znoGfile: valid point, trunk would be where all the development is going on ... right?
03:50.05filewhere all the bleeding edge... development... stuff
03:50.10znoGok cool
03:50.16znoGthat answers it
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03:56.21synthetiqwhat would cause a 503 error
03:56.36bkw_isn't that service unavali?
03:56.39synthetiqyes
03:58.06docelm05 minutes until new pricing hits..  :)
03:58.19thermfof what?
04:01.10docelm0US48/CANADA/Toll Free
04:02.11thermfwhat carrier?
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04:05.15mdiehlHi all.
04:06.22mdiehlDo I expect to be able to change the callerid number of an incoming call without hosing the callerid name?
04:07.12mdiehlI'm trying to strip off a leading "+" if it is present.  When I do, the name doesn't display on my cid unit.....
04:07.13grabowskimdiehl: You could copy the original callerid to another variable before you change it
04:07.22grabowskimdiehl: oh
04:07.34mdiehlI've been fighting this for two days now.
04:08.04mdiehlMy provider adds a + to the cid number.  Some of my other voip clients do not.....
04:08.14mdiehlSo I can't simply strip off the first char.
04:08.45grabowskimdiehl: You should be able to, how I am not sure
04:09.07mdiehlexten => s, 3, gotoif(${REGEX("^\+",${CALLERID(number)})},4,5)
04:09.07mdiehlexten => s, 4, set(CALLERID(number)=${CALLERID(number):1})
04:09.33mdiehlWhen this code runs, I get the phone number displayed twice on my phone, no name.
04:12.15grabowskidoes CALLERID(number) work? I thought it was CALLERID(num)
04:13.00mdiehlIt seems to.
04:13.46grabowskimdiehl: http://www.voip-info.org/wiki/view/Asterisk+variables
04:14.02grabowskimdiehl: read the part about ${CALLERIDNUM}
04:15.08mdiehlI thought that was deprecated.
04:15.35grabowskimeshuga: yes but read the note about it
04:17.12grabowskimdiehl: sorry that was ment for you ^
04:17.14mdiehlSays, I should be using CALLERID(num).....
04:17.17mdiehlI got it.
04:17.20mdiehlThanx.
04:17.33mdiehlStrange.  Wonder why (number) is working.......? <grin>
04:18.13grabowskimdiehl: I am also wondering about the "Note that this is not necessarily numeric as the name would indicate"
04:18.34grabowskimdiehl: Are you sure that CALLERID(num) only contains the number?
04:18.35mdiehlThat's in the case where the caller isn't from the pstn.
04:18.39mdiehlYes.
04:18.44grabowskimdiehl: with the + that you would like to remove of course
04:18.51mdiehlRight.
04:19.12mdiehlFor example "john doe <sip@home>"
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04:23.43docelm0Plainvoip
04:23.52docelm0I just sent an email out on the -biz list
04:24.38grabowskidocelm0: Yea I noticed its "till the end of the month"
04:25.01grabowskimdiehl: sorry, im out of ideas and can't find anything on google
04:25.21senglandAnyone know of a way to connect active channel to a confrence room using channelspy with the AMI?
04:25.32grabowskimdiehl: is that regex gotoif syntax correct?
04:26.08mdiehlNot sure.  Seems to be working, but I've not tested it very hard.
04:26.51grabowskimdiehl: show function regex shows that its a seperate cmd from the dial plan..
04:27.51grabowskimdiehl: I have not needed to use regex in any of my dialplans yet so I'm sorry I have no idea
04:28.45mdiehlWhat do you mean by a separate cmd from the dialplan?
04:28.52grabowskimdiehl: show function REGEX
04:29.28grabowskimdiehl: I mean can you use it within a gotoif or does it need to be a seperate line / priority before the gotoif.
04:29.57mdiehlIt's a function, so I'm assuming I can use it in an expression......
04:30.10grabowskimdiehl: I guess
04:31.22compu73rg33kwhat's the coolest thing you guys have done with VOIP ?
04:31.33compu73rg33kin *
04:32.10grabowskicompu73rg33k: Not me but someone http://uc.org/read/Zasterisk
04:33.16mdiehlI've started my own phone company..... just a few friends, but hey....
04:33.25compu73rg33kyeah that's cool
04:33.33compu73rg33knice grabowski. I know someone who made http://asterwake.com heh
04:33.44compu73rg33kbah bad link hold up
04:34.05mdiehlI went from giving Quest $40/month to spending $15/month.  That's WAY cool!
04:34.58compu73rg33koh damn he gave up the site :( basically it was a wake up call program in asterisk it was cool
04:35.08compu73rg33kyeah nice mdiehl
04:35.40mdiehlThinking about doing that myself.  Using call files created by a perl script run via cron.....
04:35.51compu73rg33khaha I think tha'ts exactly waht he did
04:35.56docelm0I started a real phone company pushing millions of minutes
04:35.57docelm0:)
04:36.06compu73rg33khttp://biimsoft.com/pages/projects/code/asterwake/ for the code mdiehl if you want
04:36.16mdiehlThanx.
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04:43.28asterisk-dudany good canadian voip carriers anyone knows of?
04:44.17dlynes_officeasterisk-dud: you looking for retail or wholesale?
04:44.29asterisk-dudretail
04:44.55dlynes_officeasterisk-dud: you could try looking at www.calltermination.com; they have a huge list of terminators both wholesale and retail
04:45.01asterisk-dudactually, what's the diff, i work for a fairly larg company
04:45.11dlynes_officeasterisk-dud: define large?
04:45.24Qwellin minutes per month
04:45.25dlynes_officeasterisk-dud: like how many minutes of long distance per month?
04:45.26asterisk-dud75 extentions
04:45.50asterisk-dudmts lines are costing well over a grand
04:46.07dlynes_officehow many minutes per month though?
04:46.33asterisk-dud4000
04:46.36dlynes_officeand also where in Canada are you located?
04:46.37asterisk-dudballpark
04:46.41asterisk-dudmanitoba
04:47.06dlynes_officeasterisk-dud: well, you're probably out of luck for "good" voip carriers in Canada, then simply because of where you are
04:47.26dlynes_officeyou're not going to get super great ping times, but you'll get ok ping times
04:47.34[TK]D-Fenderyup, all the good ones are mostly eastern Canada
04:47.42dlynes_office[TK]D-Fender: and western canada :)
04:47.44[TK]D-FenderTO/MTL
04:47.49dlynes_officeVancouver
04:47.49asterisk-dudwell i have vonage and it sucks balls
04:47.57asterisk-dudis there anyway to improve it/
04:47.58dlynes_officevonage sucks a lot of stuff
04:48.05dlynes_officeincluding investors' money :p
04:48.06[TK]D-Fenderdlynes_office : I've had trouble finding ones to incoming DID....
04:48.23scrubadubi have an icable smta icsg101c, couldnt find much using it with asterisk. I'm wondering if it is possible to use it like a sipura device, allowing PSTN inbound calls routed to an asterisk server, and outbound calls from an asterisk server?      http://www.icablesystem.com/english/product/sub_ics_g101.html
04:48.31dlynes_office[TK]D-Fender: well, after I finish this godforsaken billing system, we can sell you vancouver dids
04:48.43Jason99I'm just wondering if there is a way to call up a context or something after the call hangs up?
04:48.44scrubadubbefore i've only used it as a terminating PSTN/voip box, so i'm unsure if i can use it with asterisk
04:49.28asterisk-dudwhat is a pri line?
04:49.56[TK]D-Fenderscrubadub : No, thats for plugging PHONES into, not LINES.
04:50.17[TK]D-FenderJason99 : look at the "h" exten.
04:50.29dlynes_office~wiki pri
04:50.37dlynes_officeerm
04:50.39scrubadubright ok what's the major difference between my smta and the sipura box, like what should i be googling for
04:50.41Jason99[TK]D-Fender: Thanks
04:50.46dlynes_office~wiki primary rate interface
04:51.12[TK]D-Fenderscrubadub : Sipura sells various products, some for plugging PHONES into, and ones fro pluuging LINES into.
04:51.17dlynes_office[TK]D-Fender: btw
04:51.22sengland[TK]D-Fender you tried voxbone for incoming in Canada?
04:51.24dlynes_office[TK]D-Fender: everything's up and running just ducky now
04:51.52dlynes_office[TK]D-Fender: that sangoma tech that i got promoted to is much better at troubleshooting than the other one
04:51.58[TK]D-Fenderscrubadub : You you looking to use that to plug a PHONE into to talk to *?  If so, fine.  If you want to take it and try to take an analog phone LINE into *, you're out of luck.
04:52.10[TK]D-Fenderdlynes_office : 100% now?
04:52.18dlynes_office[TK]D-Fender: yeah, on 2.4, anyways
04:52.19scrubadubyeah i'm looking for something sitting between my asterisk server and PSTN/Voip i guess "plugging lines into". are there keywords i should google for that would bring up other vendors / products
04:52.20Jason99[TK]D-Fender: Will that take effect after the call was connected for a while and then hung up?
04:52.36dlynes_office[TK]D-Fender: now i need to throw a monkey wrench into it and make it work on 2.6 :)
04:53.04[TK]D-Fenderdlynes_office : I've run A200's on CentOS 2.6 without issue... I don;'t think its a kernel issue so much as a lack of headers to match your custome kernels...
04:53.21Jason99nm.. I answered myself
04:53.31[TK]D-FenderJason99 : Yes, get reading...
04:53.38dlynes_office[TK]D-Fender: iirc, make linux, make zaptel, make libpri, Config install, make zaptel, make libpri, make asterisk, right?
04:53.52scrubadubalthough [TK]D-Fender it does have both a fxo and fxs port which was why i was wondering
04:53.55Jason99[TK]D-Fender: Thanks thats exactly what I was looking for
04:54.20dlynes_office[TK]D-Fender: couldn't be a lack of headers...I installed the kernel from source, not from kernel-dev/kernel-bin
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04:55.36[TK]D-Fenderscrubadub : Appears I read it a bit wrong... but it may only be a failover outbound port... unsure if it will take calls IN...
04:56.02[TK]D-Fenderdlynes_home : Maybe its a question of locations, or maybe a kernel dependency of some kind...
04:56.59scrubadubok, basically i'm looking for hardware for my first asterisk setup, one PSTN line, probably just use skypeout for voip connection, do you have any models you recommend. Cant find a whole lot besides http://www.voip-info.org/wiki/view/Asterisk+setup+home
04:57.29scrubadubtrying to decide if i should go pci card or external box
04:57.53[TK]D-Fenderscrubadub : Linksys SPA-3102 <-
04:57.56scrubadubbudget is around 150
04:58.00scrubadubok thanks i'll check that out
04:58.04[TK]D-FenderThats $90
04:58.22[TK]D-Fenderoffers 1 FXO, 1 FXS, each operates independantly
04:58.50dlynes_office[TK]D-Fender: i'm guessing maybe a botched install of sangoma or something
04:59.06dlynes_office[TK]D-Fender: when i retry it, i'll just recreate my kernel source directory again
04:59.20dlynes_office[TK]D-Fender: and then reinstall sangoma according to how I did it on 2.4
04:59.45[TK]D-Fenderdlynes_office : IIAB.....
05:00.09dlynes_office~wiki iiab
05:00.24dlynes_officeiiab?
05:00.27[TK]D-FenderIf It Ain't Broke....
05:00.31dlynes_officeah
05:00.34[TK]D-FenderGLUTTON...
05:00.34dlynes_officeyeah...pretty much
05:01.21dlynes_officeanyways...on that note, i'm heading home for the night
05:01.40[TK]D-FenderOk, take care.
05:01.45[TK]D-FenderI'm off as well...
05:01.47[TK]D-Fendercheckout time...
05:01.58dlynes_officethanks again for the help earlier
05:02.21dlynes_officegot another a101 on the way, too
05:02.26*** join/#asterisk litage (n=nick@203.220.55.70)
05:03.36[TK]D-Fendercool, look forward to it bombing out on you too ;)
05:04.02[TK]D-Fender(j/k)
05:04.17dlynes_officelol
05:04.34dlynes_officeyou're just full of sugar and spice and everything nice, aren't you?
05:04.54[TK]D-FenderOk, and you should try it out ont hat C#, and while you're at it, lets just bypass G711 and go right to transcoding to G729, k?
05:05.09dlynes_office?
05:05.13[TK]D-FenderC3
05:05.21dlynes_officeheh
05:05.43dlynes_officeI do transcoding on those machines
05:05.47[TK]D-FenderChoke that poor bastard till it smurfs out on you...
05:05.49dlynes_officebut not in g729
05:06.00dlynes_officeI've tried in ilbc though :P
05:06.10dlynes_officeand gsm
05:06.29[TK]D-Fendergsm isn't so bad. ILBC is.
05:06.34dlynes_officeanyways..i'm outta here now, though
05:08.19[TK]D-Fender*yawn* ditto
05:10.41Jason99Ok, using the contexts I know when the call is dialed and when the call is hung up, but is there a way to know when the call is answered after it's dialed?
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05:39.49rushowrhey quick question, should be fairly easy...Anyone know for sure what version the Filter() function is available in?
05:41.20Qwellrushowr: trunk?
05:41.29Qwellbed
05:41.31rushowrso it's not available in 1.2.8
05:41.35Qwelldunno
05:41.36rushowrbed?
05:41.48Qwellbed, as in...the place I sleep
05:41.51rushowrah gotcha
05:41.52rushowrsorry
05:41.53rushowrthanks mate
05:42.11Qwelland, umm..
05:42.18QwellWhy are you still using 1.2.8?
05:42.25Qwellsee channel topic
05:42.27rushowrclient's machine
05:42.32rushowrno IAX
05:42.45rushowrsip and zap only, I've recommended to them though
05:42.50rushowrthanks again, will hit them again
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05:58.15stephane_jour
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06:16.58dlynes_homestephane_: Est-ce qu'un triggere?
06:17.50stephane_triggere ?
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06:36.21MACscrsince I have a dynamic ip address at my home office, should i not even think about running an asterisk server here?
06:38.00Gabriel25MACscr you can fix that
06:38.12MACscrjust run an updater?
06:38.16Gabriel25dyndns
06:38.34MACscrok, cool
06:38.48MACscrwasnt sure if it would work ok without using ip addresses directly
06:39.12Gabriel25i have like this and is working with no problem
06:39.23MACscrwell, i just used TrixBox and installed everything fine
06:39.42MACscrhavent decided on a voip provider yet
06:39.48MACscri wish i could use vonage
06:40.46Gabriel25don`t use the word vonage noone like that here
06:40.47Gabriel25:))
06:41.00Gabriel25I was almost banet for this word
06:41.22MACscrthats retarded. They have decent service, the coverage is one of the largest, and they are cheap
06:41.28MACscri have 3 lines with them
06:41.51Gabriel25https://www.teliax.com/newaccount/
06:41.54Gabriel25some prices
06:42.03MACscryep, im looking at them
06:42.18MACscri hate going with any company though that doesnt have 24/7 phone support
06:42.45Gabriel25you are right about this one :)
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06:43.07MACscrive heard good things, but a lot of voip providers run their sites like they are a 1 person company
06:43.19MACscrand built it with frontpage
06:43.41Gabriel25:))
06:43.47MACscri dont know. Im a web host and designer, so im pretty critical
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06:43.58VoIPMastaHi there
06:43.58Gabriel25I didn`t try
06:44.07Gabriel25wow nice
06:44.18MACscri worked in telecom for 2 years as well, so im pretty familiar with how things work
06:44.27MACscrnot to much voip experience though
06:44.33Gabriel25where you are from ?
06:44.43MACscrPeoria, IL, USA
06:44.50VoIPMastaI have a question for those experienced with dialplans... let's say that I have 2 devices, one is IAX and the other one is SIP. Now I want to receive calls to my extension regardless of wether device I'm using to connect... any ideas?
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06:45.39Gabriel25I know some people in verizon bussines ex MCI in New York
06:45.40Gabriel25:))
06:46.30MACscri worked with SBC (AT&T) mostly
06:47.03MACscrwhats harder about finding a good provider is that i want to find one that i can port my numbers too
06:47.18MACscrteliax is one that can, but there arent many that i have found that can
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07:13.53x86MACscr: no way, I'm from peoria as well ;)
07:14.04MACscrlol, really?
07:14.05x86MACscr: east peoria anyway
07:14.09x86yeah man :)
07:14.10MACscroh, im sorry
07:14.11MACscrjk
07:14.13x86hehe
07:14.14MACscr=P
07:14.17x86the good side of the river ;)
07:14.20MACscrim originally from washington
07:14.23x86ah ok
07:14.31x86right down the road from me ;)
07:14.38MACscrso you want to configure my asterisk system for me so i can go to bed?
07:14.40MACscr=P
07:14.40x86you work for cat?
07:14.44MACscrlol, no
07:14.50x86hehe
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07:15.09MACscrim an IT director for a brokerage firm
07:15.46nextime'morning
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09:01.02Tmobhey guys... i have a sip capable phone.. i'm not too familiar with phone tech.. so curios if anoyne here knows where i can buy some accont so i can make calls using sip
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09:01.24Tmobsaves me a lot of $$ onphone bills if i cna do that .. esp when i'm travelling out of country :)
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09:08.01robin_szthere are thousands of SIP providers
09:09.31Tmobrobin_sz, i am tryig to search for ones which let me use my phone to connect to them
09:09.40Tmobrobin_sz, i found a few but they have their own applications
09:09.46Tmobrobin_sz, can you recommend any?
09:09.57{zombie}look on voip-info.org
09:10.07{zombie}there are several lists of VoIP providers
09:10.15robin_szwhat he said, there a list on voip-info.org
09:10.25Tmobthanks guys :)
09:10.28Tmobi'll check it out
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09:12.23robin_szprobably wont save you a lot though .. as you travel around the world, finding an network conneciton is ususally more expensive than just using a mobile to make calls
09:12.41robin_szbut, in theory, you should save
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09:17.32Tmobrobin_sz, hehe yea, but you will be surprised how many calling cards i use up when in europe
09:17.53Tmobrobin_sz, i have a theory their clocks run faster than ours ;)
09:19.08Tmoboops.. yes i did fall for that one :(
09:19.09Tmobsorry
09:19.12robin_szhehe
09:19.21Tmobrobin_sz, you work for emc?
09:19.23robin_szim in Geneva :)
09:19.30Tmobah cool..
09:19.44robin_szemc? the storage people?
09:20.02robin_szemc the linux based machien tool controller?
09:20.11Tmobrobin_sz, heh thought it was storage.. nm
09:20.17robin_szeither way, no
09:20.32Tmobrobin_sz, i'll be in zurich in 3 weeks :)
09:20.37robin_sznice,
09:20.45Tmobbut far away  from geneva
09:21.02robin_sznot so far, its a very small country
09:21.06Tmobusually nothing to do in geneva i think.. except diplomats doing useless meetings.. hehe
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09:21.11robin_szin 3 weeks I will be back in the UK
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09:53.52Eric-xxhow to upgrade asterisk?
10:00.24TiliEric-xx: download new tarball u want. delete everything in /usr/lib/asterisk/modules. Extract new tarball somewhere. make and then make install
10:30.56x86Tili: what about /var/lib/asterisk ?
10:31.04x86or /usr/bin/asterisk? heh
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10:31.22Tilix86: dont need to do anything in there
10:31.41Tilix86: /var/lib/asterisk holds the agi and sounds etc. these are not such integrated parts of *
10:31.54Tilix86: /usr/bin/asterisk will get overridden when you do make install
10:32.28Tilithe only thing about modules directory is that modules are loaded from that directory.
10:32.31x86so will /usr/lib/asterisk/modules ;)
10:32.38Tilix86: no
10:32.42Tilimodules dir wont
10:32.53x86no? read the make file and try again :)
10:32.57Tilionly those modules that were in old * and are in new * will get overriden
10:33.04Tilix86: I know what I am doing
10:33.15x86read the make file, because you're wrong
10:33.21x86ah right
10:33.34Tilibut if some module is not supported anymore, it will still be loaded in *
10:33.36x86only same module names will be overridden
10:33.41Tiliyeah
10:33.57Tilithis issue arises mostly when u downgrade. but can happen in upgrade also
10:34.01x86well, you didnt tell him to remove /etc/asterisk/*, so the old module will be in modules.conf still ;)
10:34.05Tiliunless make file removes all modules in modules dir
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10:34.38Tilix86: modules.conf is no problem usually. it tells not to load a module
10:34.47Tiliand removing /etc/asterisk/* is dangerous
10:34.52x86very
10:34.52Tilias you will lose all ur earlier settings
10:35.00x86unless you use realtime ;)
10:35.07Tilix86: true
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11:24.19jhiverhi all
11:24.39jhiveris there a way to detect if a gateway has timed out from the dialplan?
11:25.27jhiverlike you do Dial(SIP/number@gateway) <------- if a call couldn't get through because the gateway has timed out, i want to do something about it
11:26.16jhiverthe idea would be to suspend gateway which time out for 60 minutes, and then suspend them permanently and send an email alert on the second time
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11:35.35jhiverI guess there isn't heh :)
11:35.42jhiveror maybe this channel is dead :)
11:35.53jhivervi rocks, emacs sucks! should sort it out
11:36.09robin_szshh, sleeping!
11:37.55drraythree chers for sed!
11:38.04jhiverlol
11:38.09robin_szbah, hex ed
11:38.12robin_szhexedit
11:38.39jhiversee? vi vs emacs always works to wake up a channel :)
11:39.22jhiverlet's do something more outrageous
11:39.28jhiverOpenPBX rocks!
11:40.01jhiverhopefully that'll wake up some hardcore digium supporters and then somebody will know about my poor issues :)
11:40.44jhiveroh no! they go away now :)
11:41.06robin_szjhiver, are you familiar with IRC commands?
11:41.21jhivernot very much no
11:41.30robin_szwait, its ok ... I found it .. /ignore
11:41.35jhiverlol
11:41.43jhiverallright I'll shut up
11:41.54jhiverjust faffing about really :)
11:42.00robin_szno kidding
11:42.29jhiverwell since everybody's asleep a bit of sillyness doesn't hurt does it
11:43.38drraymaybe if you spent your time reading the wiki
11:43.50jhiverI do
11:44.03jhiverI've been on Dial and RetryDial pages but haven't seen anything so far
11:44.36drrayyou know about the t extension?
11:45.03jhiverbut that's not timing out because the gateway doesn't answer is it
11:46.17drrayok, so why does the timeout variable in the dial command not work for you?
11:47.11jhiverisn't it timing out if a call hasn't been answered under 't' seconds?
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11:47.29drrayyes
11:47.40jhiverthat's not what I want to do
11:48.09jhiverI want to detect if the call hasn't gone through because the GATEWAY has timed out (i.e. it's down for instance)
11:48.30jhiverso I can suspend it temporarily
11:49.01jhivernow if somebody lets a phone ring for 60 or so seconds, it's fine by me
11:49.01drrayI don't suppose least cost routing would help
11:49.29jhiverbut if the gateway doesn't answer to INVITE requests under 5 seconds, I want it out
11:49.34jhiverdoes it seem reasonable?
11:50.55jhiverdrray, how would it help?
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12:33.12synthetiqanyone know what would cause a 503 (service unavail) error?
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12:54.45coplandHello,  if a call is coming in on a trunkA that says gets fowarded by follow me to a pstn number that it always uses TrunkB and not any of my other trunks
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13:04.27queuetueHow would I build a tool that allows me to access CID info on my desktop machine?
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13:13.14x86queuetue: most likely one such tool already exists
13:13.20x86queuetue: google for asterisk screen pop
13:13.29x86queuetue: i got it setup on my Mac very easy
13:13.53queuetuex86, Good, mac integration is what I'm looking for - will it pause iTunes? :)
13:14.38x86lol not that i know of
13:14.47x86it has Growl integration which is quite nice
13:15.10queuetuex86, Do you know the exact name?  asterisk screen pop doesn't seem to be doing it...
13:16.59x86http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Notify
13:17.07x86you'll need the app_notify.so module
13:17.24queuetuex86, Ok, that's what I was already investigating. :)  Thanks.
13:17.28x86http://www.voip-info.org/wiki/view/Asterisk+call+notification
13:17.47x86this one shows desktop apps that tie into the asterisk management API and give screen pops
13:18.34x86wait no
13:18.36x86go here:
13:18.42x86http://www.mezzo.net/asterisk/app_notify.html
13:18.48x86there is a section on MacOS X client
13:20.02x86Growl + Address Book integration
13:20.06x86cant be beat :)
13:20.15x86http://www.mezzo.net/asterisk/AsteriskNotifyClient-1.0rc5.dmg
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13:24.59x86the dialing from Address Book seems to be broken though
13:25.14x86but not a big deal for me, as I have custom software I use for that anyway
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13:34.01asterisk-dudfor some reason i can't get fop to work
13:35.16asterisk-dudindex.php doesn't kick in
13:35.30asterisk-dudi can only see the files in the directory
13:35.32x86sounds like your apache configuration skills are weak ;)
13:36.03x86ask about that in #apache
13:36.05x86not here :P
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14:02.54jhiverhi again all
14:03.04jhiveris there a way to detect if a gateway has timed out from the dialplan?
14:03.10jhiverlike you do Dial(SIP/number@gateway) <------- if a call couldn't get through because the gateway has timed out, i want to do something about it
14:03.23jhiverthe idea would be to suspend gateway which time out for 60 minutes, and then suspend them permanently and send an email alert on the second time
14:03.28jhiverany ideas?
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14:28.25littleballhello, my domain registrar doesn't support DNS SRV record. who can recommend one domain registra?
14:29.12lunkgodaddy
14:29.20lunkbig tit commercials for the win
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14:36.22eKo1Is it possible to configure two spans on the TE410 to share a D-channel?
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14:38.21wunderkineKo1, that is what nfas is for
14:39.48eKo1Hmm...I'm going to have to play around with this stuff.
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14:40.26robin_szlittleball, there are no registrars on the planet that "support SRV record", nor will there ever be.
14:41.05robin_szlittleball, SRV records come out of your DNS, registrars simply add NS records to the root servers
14:41.44robin_sztwo entirely different things
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14:51.11littleballrobin_sz, http://www.voip-info.org/wiki-DNS+SRV
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14:58.12eKo1Is there a way to take out a span from operation within the  * CLI?
15:00.22tzafrir_laptopeKo1, what spam?
15:00.37wunderkinno
15:01.10WiredXhey everyone..
15:01.55tzafrir_laptopeKo1, is that span actually doing something useful?
15:02.44wunderkinyes.. some butt fucker picked up my email from an asterisk mailing list and is now spamming other users on my domain now.. ugh
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15:05.12eKo1tzafrir_laptop: nope, it is complaing about no d-channel
15:05.15eKo1I want to take it down.
15:05.33tzafrir_laptopztcfg -s NUM          ?
15:05.50tzafrir_laptopOr just remove that span from your zapata.conf
15:06.40tzafrir_laptopWiredX, you wanted to ask something?
15:06.47eKo1I did remove it from zapata.conf. That didn't work. Doesn't ztcfg -s shutdown all the spans?
15:07.04tzafrir_laptopeKo1, did you restart asterisk?
15:07.45tzafrir_laptopWithout restarting asterisk: you can "destroy" zap channels, but I'm not sure about spans
15:08.49eKo1tzafrir_laptop: I don't want to restart asterisk as I have calls going through the other spans.
15:09.09eKo1I guess it can't be done.
15:09.22tzafrir_laptopYou can destro the channells of the irrelevant span
15:09.41tzafrir_laptopWorks on analog. Never tried it with ISDN
15:09.53tzafrir_laptopzap destroy channel NNN
15:10.06WiredXtzafrir_laptop: no :) just saying hi and making myself feel welcome
15:10.50jhiverhey, how come when a destination is circuit-busy it executes congestion instead of doing the next Dial() command as specified in my extensions.conf ?
15:10.59*** join/#asterisk acidchild (i=ash@unaffiliated/acidchild)
15:11.02tzafrir_laptopBTW: areyou related to JCraft's WierdX ?
15:11.08*** part/#asterisk acidchild (i=ash@unaffiliated/acidchild)
15:12.51jhiverCan anybody take a look at my macro and tell me what's wrong?
15:13.01jhiverhttp://pastebin.ca/63732
15:13.17*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
15:13.39Qwelljhiver: What is it (not) doing?
15:13.50jhiverwhen it hits exten => s,100,Dial(SIP/0${ARG1:3}@83.206.114.85) and if it's circuit busy, then it executes congestion instead of jumping to 200
15:14.18jhiverideally I'd like that if the response code is anything but answered, it goes to the next step
15:14.23jhiverI use 1.2.7.1
15:14.45jhiverand I have autofallthrough=yes
15:15.13jhiverbecause I have this:
15:15.15jhiver#
15:15.16jhiverexten => s,100,Dial(SIP/0${ARG1:3}@83.206.114.85)
15:15.16jhiver#
15:15.16jhiverexten => s,101,Goto(200)
15:15.16jhiver#
15:15.17jhiverexten => s,200,Macro(loadbalance-orange,${ARG1})
15:15.18jhiver#
15:15.20jhiverexten => s,201,Congestion()
15:15.22jhiveroops
15:15.24jhiversorry
15:15.30jhiverthat was supposed to be only 4 lines
15:16.15robin_szso the s,101 line is redundant right?
15:16.23robin_szgoto the next line?
15:16.45blitzragepriority numbers need to be sequential
15:16.47jhiverWell, I don't think it would jump straight from 100 to 200 would it?
15:16.50blitzrageso its not redundant
15:16.56robin_szk
15:16.56Qwellblitzrage: notice the goto(200)? :p
15:17.00blitzragebut you should be using 'n', not actual numbers
15:17.06blitzrageQwell: yes, I do
15:17.14Qwell100>101>200>201
15:17.20jhiveryeah well old habits from asterisk 1.0.9
15:17.23blitzrageright... thats why the Goto()
15:17.24Qwelljhiver: check the DIALSTATUS var
15:18.35jhiverI know it's a circuit busy
15:18.49jhiverthat's what it says on the CLI
15:19.15*** join/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net)
15:19.18SexyKenHey there fellas!
15:19.21jhivertrouble is I can't faff about too much with it because it's in production, I guess I'll have to wait during night time to see what's going wrong
15:19.36Qwelljhiver: You assume it'll do priority jumping
15:19.51jhiverbut anyway, isn't asterisk supposed to go to the next step when a call isn't answered?
15:19.58SexyKenGuys -- we're all a pretty interactive community here!  We help each other with issues and whatnot....
15:20.05blitzragejhiver: yes
15:20.14jhiveris there some option I haven't set or something?
15:20.29robin_szjhiver, just type extensions reload a few times and you'll soon solve the "in production" annoyance ;)
15:20.33SexyKen....maybe you guys would be interested in donating some change for the Relay for Life that I'm doing?  It's to raise money for cancer research.
15:20.39jhiverrobin_sz, ?
15:21.17robin_szSexyKen, go bother the govenrment about it
15:21.17jhiverI _do_ do reloads rather than a restart... can't really afford to restart the box except maybe at 4:00 am or something
15:21.53robin_szjhiver, well, eventually, reloads seem to make things unstable
15:22.07jhiverdoes it?
15:22.09Qwellrobin_sz: eh?
15:22.10jhivercrap
15:22.13Qwellno..
15:22.19jhivermight be why it's not working then :)
15:22.24robin_szshrug .. thats my experience
15:22.28blitzragejust reload the module you need
15:22.34blitzragereload chan_sip.so for example
15:22.34Qwellrobin_sz: got a bug number?
15:22.41Qwellblitzrage: sip reload!
15:22.47Qwell(is there actually a difference?)
15:22.51blitzrageQwell: I perfer just to reload the module :)
15:22.55blitzrageQwell: nah... no difference
15:22.56robin_szand thats guaranteed memeroy leak and after effect free?
15:23.06jhiverok, well, I'll do some tests tonight at 4am GMT + 4
15:23.19*** part/#asterisk SexyKen (n=Ken@c-24-5-129-114.hsd1.ca.comcast.net)
15:23.27jhiverthe channel should be still pretty active so I guess I'll ask when I have some time to fuck about with my Asterisk box
15:23.51jhiverbecause right now there is too much traffic for testing
15:23.55robin_szpersonally, after a bit of reloading, I do a stop when conveninet and a resatrt
15:24.22ghenryWhere do you set things to put a zap channel on hold, call another user, then put the original caller through to them by hanging up? i.e. call transfer and call forwarding? zapata.conf? Also, is this the same as Parking a call? I don't think so, as that's to move on to another phone and retrieve the call, correct? Thanks. brb
15:24.26robin_sz"stop when convenient" is pretty kewl, unless it keeps you sitting there waiting for it to die for an hour
15:24.35jhiverit's a pretty strange issue though....
15:24.57jhiverrobin_sz, it would only work at night time for me anyway
15:24.58jhiverso...
15:25.15robin_szpeople talk too much huh?
15:25.44jhiveranyway, has this anything to do with autofallthrough=yes?
15:26.27jhiver; If autofallthrough is set, then if an extension runs out of
15:26.28jhiver; things to do, it will terminate the call with BUSY, CONGESTION
15:26.28jhiver; or HANGUP depending on Asterisk's best guess (strongly recommended).
15:26.31Qwelljhiver: priorityjumping
15:26.53robin_szQwell, http://www.voip-info.org/wiki/view/Asterisk+administration
15:27.05robin_szA repetitive reload will not be sufficient, and can actually cause more harm (instability, memory not being released, see bug tracker) than it does good
15:27.10robin_szok?
15:27.18jhiverI don't have that set
15:27.18Qwellwtf
15:27.22jhivermight be it
15:27.26Qwellthere is a ton of BS on this page
15:27.50robin_szthere it is .. in the wiki .. whenever I ask I always get told "read the wiki" I did .
15:27.55QwellWTF
15:28.29jhiverDial sets DIALSTATUS to indicate its success. However, under some circumstances, execution will jump to priority n+101 in the current context
15:28.31QwellNote that it is recommended to do a clean CVS checkout instead of a CVS update!
15:28.32QwellBULLSHIT
15:28.52ghenryIs it config option as well, that when a zap/sip user puts a call on hold, they hear music?
15:28.53robin_szwell, feel free to edit it :)
15:29.00jhiverso
15:29.18jhiverI guess I have to add priorityjumping=yes to the [general] section
15:29.47jhiverand then change the extensions.conf to have something smart at extension 201 (since I attempt the dial at 100) ?
15:29.47Qwelljhiver: Do you want it to jump to 201?
15:30.02QwellYou said it already is
15:30.06Qwellwhich means you have that set
15:30.25jhiverit isn't!
15:30.25Qwellso then check DIALSTATUS
15:30.25jhiverit executes Congestion when everything is fast-busy
15:30.30QwellSo then it is
15:30.40QwellSo remove that option, and check DIALSTATUS
15:30.40jhiverI thought it would go to +1 but apparenty not
15:31.47*** join/#asterisk parag7732 (n=root@de1-b1453.alshamil.net.ae)
15:33.09DaminThat's complete BULLSHIT! You have no idea what the hell you are talking about!
15:33.27QwellDamin: troll
15:33.28Qwell:P
15:33.30parag7732Hi...I m wondering that i m able to configure the "external voip provider account" with Linksys SPA942 and Pap2.....but its not working with asterisk....!!!!!!!!!  why
15:33.39QwellThere we go.  Line 3.  http://www.voip-info.org/wiki/view/Asterisk+administration
15:33.49parag7732i created the right trunks
15:33.56parag7732and peers
15:34.45robin_szQwell: thats a start, now repeat the process for the rest of the wiki
15:41.23stephane_re
15:45.49Dr-Linuxre
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15:48.47*** join/#asterisk Borgon (n=l3orgon@host-69-59-103-160.nctv.com)
15:49.11Borgonif am using a voip software that uses sip, can i make it connect to my asterisk by sip then use asterisk to dial using iax2?
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15:52.41grabowskiBorgon: of course
15:52.54Borgongrabowski: hrm ok
15:53.24jaikeanyone running asterisk on AMD dual cores?
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15:58.22Dr-Linuxwhy WIKI suggest to restart the asterisk regularly?????
15:58.26Dr-Linuxhere:
15:58.28Dr-LinuxRegularly restart (better: stop and start) your PBX during off-hours. A repetitive reload will not be sufficient, and can actually cause more harm (instability, memory not being released, see bug tracker) than it does good. If you run Asterisk provisioned for automatic reloading this could be as simple as placing a cronjob to execute asterisk -rx 'stop gracefully'.
15:59.43QwellDr-Linux: It's wrong
16:00.06kumamotoDamn Qwell you are still here?
16:00.19Dr-LinuxQwell: i'm reading the link, that you pasted above
16:00.22Qwellkumamoto: sure
16:00.27Dr-Linux<Qwell> There we go.  Line 3.  http://www.voip-info.org/wiki/view/Asterisk+administration
16:00.29QwellDr-Linux: somebody else pasted it
16:00.34QwellDr-Linux: yes, read line 3 ;)
16:00.40kumamotoI appreciate your help in this channel
16:00.54Qwellkumamoto: donations accepted :p
16:00.58Dr-Linuxok :P
16:00.59*** join/#asterisk zotz (n=zotz@24.244.133.115)
16:01.07QwellDr-Linux: me, saying all the stuff there is wrong, heh
16:01.33Dr-Linux:P
16:01.36jaikefrom experience, better restart rather than just reload
16:01.43Dr-LinuxQwell: what's the line 3 ?
16:01.49kumamotoQwell, once I get into the business of voip full time I will be here fulltime and maybe we can talk on donations.
16:01.58QwellDr-Linux: "edit: Half of the stuff in here is completely false. Take it all with a grain of salt."
16:01.59kumamotoAt this time I am still learning
16:02.15QwellDr-Linux: That's line 3
16:02.46Dr-Linuxok
16:03.06kumamotoI was wondering those IP phones with 2 nics is one intended for POE or for bridging
16:03.18Qwellkumamoto: It's a builtin switch
16:04.24kumamotoaha so not a POE thing? That is good to know I always thought it was a POE port
16:04.26kumamotothanks
16:04.38Qwellkumamoto: poe uses the same port
16:04.44Qwellif the phone supports poe, that is
16:05.03kumamotoso it is interchangeable if it supports POE
16:10.07*** join/#asterisk anonymouz666 (n=anonymou@200.218.193.6)
16:12.56[TK]D-Fenderkumamoto : Some phones have switches, some have PoE, others both or neither.
16:13.32[TK]D-FenderPoE is only ever enabled on 1 of the 2 ports on the phone if present at all.
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16:16.29kumamotoaha nice
16:18.04[TK]D-Fenderkumamoto : What models are you looking at?
16:20.06kumamotoBudgetstream GS-102
16:20.25kumamotoor the polycom 103
16:21.09wunderkinheh.. you have that all backwords
16:21.28kumamotoi thought no one will catch it hehe
16:22.20kumamotoGrandstream Budgetone 102
16:22.36Qwellyuck
16:22.42Qwellbarbietone
16:22.52wunderkin:D
16:22.54kumamotoPolycom SoundPoint IP 301
16:23.12kumamotoaaah what is now wrong with the grandstream
16:23.43QwellMy First VoIP Phone (TM)
16:24.35*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
16:24.49kumamotowas it the 101 or 102?
16:24.56kumamotoor GXP2000
16:27.11jaikepolycoms very sturdy
16:27.46Qwellcisco very purdy
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16:28.14kumamotoSo what y'all saying is rather spend extra on the polycom than the grandstream
16:28.20Qwellyes
16:28.41kumamotook how does the faxing work
16:29.54[TK]D-Fenderkumamoto : Forget Grandstream altogther.  Cheap crap.
16:30.37kumamototrue it is cheap
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16:31.47anonymouz666grandstream firmware is not good.
16:31.50[TK]D-Fenderkumamoto : And believe me the "crap" is just as true.
16:32.06[TK]D-Fenderkumamoto : Polycom is great stuff....
16:32.07kumamotohow about the sipura phone
16:32.27[TK]D-Fenderkumamoto : Sipura/Linksys SPA-94X can be ok, but ti depends.
16:32.33[TK]D-Fenderkumamoto : Where are you located?
16:32.46kumamotoin dallas
16:33.13[TK]D-Fenderkumamoto : Then Sipura's price point devalidates it.  Polycom is a superior product at a price on par with Sipura
16:33.20[TK]D-Fenderkumamoto : www.atacomm.com
16:34.24kumamotodamn thanks for that links I always seem to stick to voipsupply.com
16:35.46kumamotoAny reviews on the Snom phone?
16:36.12[TK]D-FenderSnom is better than the cheap crap, but still unstable firmware, and crappy LCD usability
16:36.22[TK]D-Fenderkumamoto : Stick with Polycom, trust me...
16:37.35[TK]D-FenderSo far the only other phones in Polycom's class are Cisco, and they are PRICY.
16:37.35kumamotoisn't snom the one running with a linux based firmware?
16:37.56kumamotoI guess for now polycom wins
16:37.57[TK]D-Fenderkumamoto : Yup... great web interface and all that, but when it comes to using the phone... cik...
16:38.08[TK]D-Fender(in reference to SNOM)
16:38.22*** join/#asterisk JakBeatZ (n=JakBeatZ@trek.tor1.ebit.ca)
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16:38.55kumamotopolycom will work with asterisk@home (tribox)?
16:38.58[TK]D-FenderPolycom is a bit more complex on setup, but it pays, and is stable and a sturdy product.
16:39.23kumamotoWhat is the software that does faxing using voip?
16:39.32[TK]D-Fenderkumamoto : Like any other SIP phone yes, but Trixbox is a craptastic cookie-cutter system that will box you in.
16:39.36kumamotoI guess polycom it is
16:39.40JakBeatZFolks.. having a weird issue with a 7960.   have a bunch of SIP extenstions configured on the phone and call waiting is turned on, on the phone, but when I try to dial one extension from the other (on the same phone) I get a 486.  I'm running 8.2 SIP on the phone.. anyone seen anything like that before?
16:39.48[TK]D-FenderYou're better off learning how to run * for yourself.  there really isn't that much to it.
16:40.27*** join/#asterisk awe6 (n=lba@user-12lml5g.cable.mindspring.com)
16:41.04kumamotoIt is those configuration files that seem to be intimidating but I will get over it
16:41.54[TK]D-Fenderok, ewll good luck to you, I'm off...
16:41.57[TK]D-Fenderback later....
16:42.17kumamotothanks to all your help and good information
16:42.22kumamotoI will be back later
16:42.25*** part/#asterisk kumamoto (n=eryco@68-189-215-167.dhcp.ftwo.tx.charter.com)
16:45.44*** part/#asterisk parag7732 (n=root@de1-b1453.alshamil.net.ae)
16:47.04*** join/#asterisk doolph (n=doolph@200.75.204.169)
16:47.12doolphhow can I compile without zaptel
16:47.26Qwelldoolph: same way
16:47.45doolphin asterisk dir I get chan_zap.c: In function `pri_dchannel':
16:47.56Qwelluninstall zaptel
16:48.02Qwellor upgrade
16:48.25doolphI tried to upgrade zaptel and i get another error
16:48.28doolphhow can I uninstall it
16:49.15doolphI dont have zap hardware though
16:50.00doolpherm nvm the problem was zaptel head
16:51.41*** join/#asterisk litage (n=nick@203.220.55.70)
16:56.17ghenrywith faxdetect, is it easy to redirect this call to a fax machine?
16:56.47*** join/#asterisk litage (n=nick@203.220.55.70)
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17:00.45dstr_2How do I pass the pundkey through asterisk? When calling another pbx they ask me to press 1 and the pound key, which drops me into my own asterisk
17:05.09*** join/#asterisk litage (n=nick@203.220.55.70)
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17:09.45jaikechange the transfer key in features.conf? not sure though
17:10.11grabowskidstr_2: just dont add the T option to your outbound Dial cmd
17:10.12*** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net)
17:10.59grabowskidstr_2: This disables the feature that allows you to transfer on your own PBX with the pound key tho (for that outbound dial cmd)
17:11.32dstr_2grabowski : is there a way to work around it?
17:12.04grabowskidstr_2: besides what jaike said and change the transfer key no
17:12.26*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220)
17:12.35grabowskidstr_2: You can still transfer on a phone that has a transfer button..
17:13.09dstr_2all I find is blind transfer
17:13.36grabowskiI think pound is a blind transfer
17:13.54dstr_2;blindxfer => #1                ; Blind transfer
17:14.04grabowskidstr_2: http://www.voip-info.org/wiki-Asterisk+config+features.conf
17:14.34grabowskidstr_2: ;blindxfer => #1                ; Blind transfer, default is #
17:15.14grabowskiso you could uncomment that line and that should make the new blind transfer #1
17:16.28JakBeatZHey.. anyone have any ideas why a 7960 won't let me dial from one extension to the other on the same phone even though call waiting is enabled?
17:16.40QwellJakBeatZ: different contexts?
17:17.03*** part/#asterisk jaike (i=jaike@210.213.168.167)
17:17.04dstr_2grabowski : ha! it works now!
17:17.09dstr_2i changed it to *9
17:17.55JakBeatZQwell:  No, same context.  I get a 486 busy here.  It's like the cisco doesn't know call waiting is enabled
17:18.43dstr_2Perhapps i should just add a dialplan for the banks phonenumber, without the T?
17:18.48*** join/#asterisk litage (n=nick@203.220.55.70)
17:19.24grabowskiJakBeatZ: I have a 7960 beside me.. when I call myself its rings and I have done nothing fancy in the phone config..
17:19.55dstr_2but *9 wont work.  :/
17:20.36grabowskidstr_2: I think you need to press it quickly
17:20.38JakBeatZgrabowski:  I know, that's the funny part..   I don't understand what's going on.  I can dial from a soft phone to the 7960 no problem and vice-verse.  I can call soft phone to soft phone with no issues, but the 7960 won't do it..  What code are you running on your 7960?
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17:21.23grabowskiJakBeatZ: what do you mean by code? Its the SIP firmware P0S3-04-4-00
17:22.09grabowskiJakBeatZ: want to take a look at my SIP-mac-addy.cnf file?
17:22.14JakBeatZgrabowski:  code/firmware.. sorry.. that term is interchangable for me..  Wow, 4-4-00.  I'm up to 8-2 and it's worked fine so far
17:22.39JakBeatZgrabowski: except for the call waiting, that is..  hmm...
17:22.39grabowskiJakBeatZ: any reason I should upgrade?
17:23.21JakBeatZgrabowski:  if it works, all the power to ya :)  I'm just a child of 'latest and greatest' so that's why I'm up to 8-2.  Unless there are new features in newer code you want, but I dunno.
17:23.23dstr_2ah! 500ms was a bit to narrow
17:23.45*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
17:23.49JakBeatZs/code/firmware :)
17:24.25grabowskiJakBeatZ: you need a cisco login to get the latest sip firmware do you not?
17:25.11JakBeatZgrabowski:  I believe so, yes.  And you may need a support contract number tied to the login to be able to download firmware.
17:25.37JakBeatZgrabowski:  but I hear support contracts on phones are dirt cheap so it may be worth while.
17:26.19grabowskidstr_2: just tryed #1 worked fine for me
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17:26.47grabowskiJakBeatZ: so yea it may be something with the new firmware I guess
17:27.09JakBeatZsheeit.
17:27.37grabowskiJakBeatZ: again, if you want Ill post my SIP.cnf file
17:29.47JakBeatZgrabowski:  Sure, I'll check it out, if you don't mind.
17:29.58asterisk-dudi'm trying to recompile asterisk with spandsp and i'm getting compile errors
17:30.19asterisk-dudany advice?
17:32.33thermfasterisk-dud: what are the compile errors?
17:33.46grabowskiJakBeatZ: http://pastebin.ca/63787
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17:33.53asterisk-dudthere are a whole bunch of no member named errors
17:34.07asterisk-dudincomplete pointer type
17:34.25grabowskiJakBeatZ: Im going to look though my SIPDefault.cnf as well.. 1 sec
17:34.27robin_szdang those pointers
17:34.31asterisk-dud't30_flsuh' defined but not used
17:34.33robin_szthey really should learn to get along
17:34.49asterisk-dudapp_rxfax.c is creating the errors
17:34.54thermfah
17:34.55JakBeatZgrabowski:  Ya, nothing out of the ordinary there.  Mine is very similar.
17:35.08thermfare you compiling with the correct spandsp version?
17:35.48grabowskiJakBeatZ: in my SIPDefault.cnf I have call_waiting: "1"
17:35.49asterisk-dudi have 0.0.3 pre20
17:35.55grabowskiJakBeatZ: # Call Waiting (0-disabled, 1-enabled, 2-disabled with no user control, 3-enabled with no user control)
17:36.15*** join/#asterisk Splat (n=Splat@220-253-105-26.TAS.netspace.net.au)
17:36.45JakBeatZgrabowski:  ditto.  except that's in my SIPmac.conf
17:36.49thermfasterisk-dud: where did you get the app_rxfax from? it is probably made for 0.0.2
17:36.57asterisk-dudthemf: thats the prob, thanks
17:37.09grabowskiJakBeatZ: maybe it needs to be in the SIPDefault.cnf as well?
17:37.37JakBeatZgrabowski:  Not sure.. I'll check.
17:37.56thermfasterisk-dud: did you just download 0.0.3pre20 today? i actually have never seen that version
17:38.27asterisk-dudyes i did, but i'll download the pre26
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17:42.15thermfdoes anyone have the version of app_rxfax/app_txfax that makes use of t.38?
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17:46.07grabowskiJakBeatZ: any luck?
17:48.33JakBeatZgrabowski:  No.  I just posted to the Cisco NetPro forums..
17:49.06JakBeatZgrabowski:  I never used a SIPDefault.cnf file and call waiting worked previously many moons ago when I tried so I don't think it's that.
17:49.40grabowskiJakBeatZ: Well, good luck with that.
17:50.26JakBeatZgrabowski:  thx for your help.
17:53.08dstr_2Which IP-phone is the cheapest/best for use with asterisk?  could this be a good pick? http://www.ntsweden.se/public/voipsortiment/IP300.pdf
17:53.56dstr_2i want something simple at home
17:54.11*** join/#asterisk shmur (n=blern@157.130.10.166)
17:54.13grabowskidstr_2: cheapest and best do not coincide
17:54.17dstr_2hehe
17:54.25grabowskidstr_2: the best of the cheapest maybe..
17:54.45dstr_2just cheapest then... since it's for home.
17:56.03*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
17:56.05grabowskidstr_2: The Grandstream budgettone (I think) is about the same quality of your average residential phone. For something a little better without much more maybe get a polycom.
17:56.51grabowskidstr_2: Take a look at http://voip-info.org/wiki/view/Cheapest+ATAs+and+Service in the Phones section.
17:59.27grabowskidstr_2: I guess get a Grandstream BT-102 so you have the dual RJ45
18:01.35dstr_2looks nice
18:01.36grabowskidstr_2: The polycom you linked there is a good quality phone.
18:01.55dstr_2i think the polycom is about $130 here in sweden
18:05.19JakBeatZgrabowski:  looks like it's firmware.  put P0S3-07-5-00 on it and it works fine.
18:05.27asterisk-dudspandsp and asterisk recompiled without errors but application rxfax is still not found
18:05.38asterisk-dudhow can i make sure it's loaded
18:05.43*** join/#asterisk salviadud (n=ralfalfa@201.133.207.93)
18:05.52grabowskiJakBeatZ: Yet another reason not to upgrade.
18:08.09*** join/#asterisk nassy (n=nassy@207-38-197-201.c3-0.wsd-ubr1.qens-wsd.ny.cable.rcn.com)
18:10.05thermfasterisk-dud: asterisk -r -x 'show application rxfax'
18:11.52*** join/#asterisk yxa (i=lonari@cm121.gamma228.maxonline.com.sg)
18:12.01*** join/#asterisk DarKnesS_WolF (n=wolf@196.218.41.192)
18:12.52shido6rxfax and spandsp, eh ? :)  you make the changed to your Makefile in the apps directory?
18:13.08yxawhat does this mean: Got SIP response 485 "Ambiguous" back from x.x.x.x
18:13.59asterisk-duddoes asterisk overwrite my extension.conf file when i recompile to a newer version?
18:14.10shido6as long as you dont do make samples
18:14.15asterisk-dudok, thanks
18:14.43mitchelochey shido6
18:14.47*** join/#asterisk anderiv (n=anderiv@207-67-87-34.static.twtelecom.net)
18:14.53shido6hey
18:15.01mitchelochow have you been?
18:16.04shido6ok||ko
18:17.18*** join/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
18:17.21TripleFFFFhey akk
18:18.08TripleFFFFok where can i find old specs on the TE110P
18:19.54yxaanyone knows what the SIP response 485 means?
18:20.07TripleFFFF4xx is authorization
18:20.26*** join/#asterisk NDT (n=noone@cpe-72-228-10-145.nycap.res.rr.com)
18:21.37NDTAnyone running 2 or more TE410P cards in a dell 2850?
18:22.23TripleFFFFbad idea
18:22.35TripleFFFFso where can in find the te 110 ancestor
18:22.54justinu|laptop485 ambiguous means there wasn't enough digits in the address to route the call
18:23.08NDTI have 1 card that runs fine added another and it doesn't see it...even change the dial on them to 0 and 1 respectively
18:24.25yxajustinu|laptop means i'm missing something?
18:25.34*** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg)
18:28.04justinu|laptop21.4.23 485 Ambiguous     The Request-URI was ambiguous.  The response MAY contain a listing of    possible unambiguous addresses in Contact header fields.  Revealing    alternatives can infringe on privacy of the user or the organization.    It MUST be possible to configure a server to respond with status 404    (Not Found) or to suppress the listing of possible choices for    ambiguous Request-URIs.
18:28.39justinu|laptopExample response to a request with the Request-URI    sip:lee@example.com:        SIP/2.0 485 Ambiguous       Contact: Carol Lee <sip:carol.lee@example.com>       Contact: Ping Lee <sip:p.lee@example.com>       Contact: Lee M. Foote <sips:lee.foote@example.com>
18:29.21thermfdoes anyone have the version of app_rxfax/app_txfax that makes use of t.38?
18:29.55TripleFFFFhey is t38 avail / ?? boutny is sitll up lol
18:30.14*** join/#asterisk ikey (i=ikey@220.226.47.200)
18:33.44*** join/#asterisk salah (n=salah@216-30-75.0505.adsl.tele2.no)
18:36.07shmurhi everyone, im looking at setting up an asterisk box at work, and Im wondering about support for a specific dialogic card? i see the support for 240 on the website, but its not the specific model. is the support for dialogic cards fairly universal?
18:36.36*** part/#asterisk salah (n=salah@216-30-75.0505.adsl.tele2.no)
18:38.07asterisk-dudtiff will not biuld: [tif_stream.lo] Error 1 \
18:38.18shmurthe specific model is D/240sc-T1 Rev 3
18:38.26asterisk-dudline 837: g++: command not found
18:38.33asterisk-dudi have gcc installed
18:38.43justinu|laptopshmur: a little advice... forget about the dialogic card, and get a Digium or Sangoma T1 card
18:38.57justinu|laptopsingle T1 card is only about $459
18:39.24shmurI would like to buy a fully support card, but for right now the boss kind of wants to use what we have in house which is why im wondering
18:39.44websaewho can fuck me?
18:40.25websaei want to fuck everyone
18:41.41QwellI want to do something..that matters..
18:42.19salviadudnin song?
18:42.25Qwell:D
18:42.37Qwellsalviadud: was wondering if anybody would catch that
18:42.57salviadudQwell, I like NIN, especially the quake soundtrack
18:43.01Qwellmmhmm
18:43.31QwellFor some reason, I don't actually have that
18:44.04*** part/#asterisk littlejohn (n=little@host217-58.pool8717.interbusiness.it)
18:44.10salviadudwell, you can down it from galbadia hotel, or just get a copy of quake 1
18:44.20Qwellyeah..
18:44.35shmursince the advice is to forget about the dialogic card, can i assume that the support for it in asterisk isn't the greatest?
18:44.38ghenryhow do you transfer a call on a SIP phone?
18:44.46Qwellghenry: transfer button?
18:44.51salviadudQwell, it's worth your while dude, it's the best MOH if you want them to hang up quick
18:44.55*** join/#asterisk postel (n=jp@unaffiliated/postel)
18:45.00Qwellheh
18:45.04ghenryYeah ;-) I see a XFer button
18:45.06SRCRI'm unable to get inbound SIP calls see http://pastebin.ca/63826 for details.
18:45.54ghenryQwell: what about on an anagloue phone?
18:45.55justinu|laptopSRCR: check line 50
18:46.29SRCRjustinu|laptop: i have a entry provider-in in extensions
18:46.41justinu|laptopan entry called "provider"?
18:48.03TripleFFFFthe ancestor of digium there was the code on google once
18:48.14TripleFFFFanyone remmber ? think it was on a spanish site
18:48.48thermfdoes Mithraen visit here?
18:49.20SRCRjustinu|laptop: I have added a [provider] in my extensions.conf.. this should do the trick ?
18:49.34justinu|laptopsrcr: no
18:49.45justinu|laptopyou need an entry in your [provider-in] context
18:49.49justinu|laptopsomething that looks like:
18:50.10justinu|laptopexten => provider,1,Playback(file)
18:50.28*** join/#asterisk anthm (n=anthm@000-445-169.area4.spcsdns.net)
18:50.28*** mode/#asterisk [+o anthm] by ChanServ
18:50.57SRCRjustinu|laptop: ok did that i'll test
18:51.36justinu|laptopSRCR: file isn't really a file you can play, replace it with something in /var/lib/asterisk/sounds
18:51.54SRCRjustinu|laptop: i know.. :) thanks
18:52.07justinu|laptopnever can tell with people in this room :P
18:52.54SRCRjustinu|laptop: it almost works.. i forgot to fireup the softphone :|
18:55.21SRCRjustinu|laptop: it works.. i stil had entries like 'exten = > s,1,Answer'
18:55.59justinu|laptopthe "s" exten is only really used for macros and channel tech that doesn't provide DNIS info
18:56.26SRCRjustinu|laptop: you lost me.. but i'll figure it out someday
18:57.18justinu|laptopDNIS is a way that you can tell what number the calling party dialed
18:57.28justinu|laptopPOTS lines for example won't tell you that
18:58.03*** join/#asterisk darby_t (i=darby_t@aaph227.neoplus.adsl.tpnet.pl)
18:58.06justinu|laptoptry the bot for acronyms you don't know
18:58.09justinu|laptop~dnis
18:58.11jbotfrom memory, dnis is A telephone service that identifies the number that the caller dialed for the receiver of the call. DNIS is a common feature of 800 and 900 services, and can identify the number originally dialed when multiple 800 or 900 numbers terminate on the same destination trunks. DNIS works by passing the dialed number to the destination device, which ...
18:58.14justinu|laptop~pots
18:58.16jbotit has been said that pots is Plain Old Telephone Service as in "Old Analogue Crap"
18:58.29TripleFFFFfoud it
18:59.12mitchelocthere should be a help doc generated off of jbots vocabulary =)
18:59.23justinu|laptop~status
18:59.24jbotSince Tue Jun  6 21:15:44 2006, there have been 76 modifications, 803 questions, 0 dunnos, 0 morons and 637 commands.  I have been awake for 3d 21h 43m 39s this session, and currently reference 110978 factoids.  I'm using about 18976 kB of memory. With 0 active forks. Process time user/system 11268.56/926.45 child 0.04/0.01
18:59.29justinu|laptopjbot, laskjdakdj?
18:59.53NDTanyone have 2 TE410P with 1st gen and 2nd gen firmware woring in the same box? I am starting to think that is my issue
19:00.57justinu|laptoppretty slim chance of you finding someone with that specific config, NDT
19:01.22justinu|laptopdoes it not work right with both cards?
19:01.37*** join/#asterisk gr0mit_home (n=Tim@extrt.txrx.org.uk)
19:02.20NDTWell...the older one was in the dell 2850...added the newer one and it won't detect the old one
19:02.39justinu|laptopi assume when the new card comes out, the old one starts working again?
19:02.44NDTlspci only shows the new one...take the new one out the old one shows
19:02.53NDThehe yep
19:02.57justinu|laptopthat's too bad
19:03.10justinu|laptopyou might wanna talk to digium about it
19:03.16justinu|laptopseems broke
19:04.09NDTyeah...what a pain in the ass...friggin datacenter is 4 hours away...There is a 2nd gen in anotehr machine...maybe I can bring it down and swp em
19:04.14NDTerr swap em
19:04.30justinu|laptopsounds like your weekend is toast
19:05.05*** part/#asterisk TripleFFFF (n=TripleFF@147-102.mc.cite.net)
19:05.11NDTI didn't think they would make them incompatible ughhh
19:06.02NDTYou can flash sangoma cards right?
19:06.28justinu|laptopnot sure... i've only worked on one system with a sangoma
19:06.32justinu|laptopA101 card
19:07.26NDTI saw this on a page
19:07.28NDT> The 2nd gen firmware has field-upgradeability. The 1st gen firmware does
19:07.28NDT> not, unfortunately. There is not currently any 3rd gen firmware, but
19:07.28NDT> when there is, you'll be able to do it yourself
19:07.37NDTwonder if I can downgrade it then LOL
19:10.10NDTsomewhere else says it isn't upgradeable...blah...don't think digium is even there on the weekend are they?
19:10.26justinu|laptopprolly not
19:11.04asterisk-dudwell, i can't get asterisk to compile with rx_fax app, it gets all kind of errors, i have checked to see if I have the same versions
19:11.06NDTnah says they aren't on the site...
19:11.10asterisk-dudanyone have this trouble?
19:11.45*** join/#asterisk L|NUX (n=linux@202.5.145.56)
19:12.45asterisk-dudanyone with spandsp experience
19:17.07reza_anyone buy from discountvoipoutlet.com before?  reputable?
19:20.22NDTnever...I know I haven't found anyone cheaper on TE410P cards though then netxusa
19:20.43docelm0Where can I find cheap TDM400P's?
19:21.22asterisk-dudmake[1]: *** [app_rxfax.o] Error 1
19:22.22NDTI buy my quad TE410P cards from http://www.netxusa.com/ guy named rick...for $1262 cheapest I have found so far
19:22.35NDTmaybe they have TDM400s too dunno
19:22.54NDTyeah says they do on their site
19:24.24asterisk-dudmake[1]: *** [app_rxfax.o] Error 1
19:25.27docelm0does netxusa sell to end users or just businesses?
19:28.37docelm0asterisk-dud quit being a dick..  What do you want?
19:30.22thermfasterisk-dud: rxfax/txfax usually compile without much trouble if you have the right spandsp version installed on your system
19:30.40thermfasterisk-dud: make sure that you remove all versions of spandsp and then "make install" spandsp 0.0.2
19:30.43*** join/#asterisk litage (n=nick@203.220.55.70)
19:31.48NDTdocelm0: think both
19:39.12*** join/#asterisk litage (n=nick@203.220.55.70)
19:40.26senglandAnyone know why I cant get the context to work under Realtime configs and IAX ?
19:40.41*** join/#asterisk MatsK (i=MatsK@83.233.97.229)
19:41.28*** join/#asterisk Whoopie_ (n=Whoopie@p54A7AFF5.dip0.t-ipconnect.de)
19:43.02Whoopie_Hi, I'd like to uninstall asterisk 1.0.10. Compiled it myself, but want to switch to debian packages (asterisk 1.2.x.x). Is there somewhere an uninstaller? there's no possibility to do "make uninstall"?
19:45.35*** join/#asterisk Ciber311 (i=Ciber@user-1087e94.cable.mindspring.com)
19:45.55Strom_CWhoopie_: just compile 1.2.9.1 from source
19:46.02Strom_Cthe packages are likely to be out of date
19:47.17*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
19:48.09asterisk-dudHunk #2 failed at 100 while trying to patch asterisk for spandsp
19:48.18asterisk-dudwhat coudl be wrong
19:48.44Whoopie_Strom_C: But are all files replaced or are there then some files from 1.0.10?
19:52.46Ciber311so what voip providers are you guys using for your asterisk boxes?
19:55.36*** join/#asterisk Jedirl (n=asdf@154.Red-217-127-168.staticIP.rima-tde.net)
19:55.38JedirlHi!
19:55.52JedirlI'm getting this: "Rejecting call on channel 0/8, span 1"
19:55.53Ciber311OMG hi2u2
19:55.56Ciber311:P
19:56.11Jedirlspan 1 is always the first E1 in my TE405 card?
19:56.32Jedirlthat message would make sense if coming from the second E1, not the first one
19:57.36Jedirlspan 1 is *always* the first E1?
19:58.57Dr-Linux~soccer
19:58.59jbotgooooooaaaaaogoalgoalgoggogogogogogogogogoaooaaaaaaaaal... oops sorry, you said soccer, didn't you.
19:59.47senglandOk what am I missing here. Under realtime I am setting up iax users. I can authenticate to the database and when I do a realtiem load from the cli the correct data is returned. However when I try and make a call the user is not placed in the correct default context. Any ideas?
20:04.49*** join/#asterisk darby_d (i=darby_t@aapf23.neoplus.adsl.tpnet.pl)
20:05.46asterisk-dudHunk #2 failed at 100 while trying to patch asterisk for spandsp
20:08.21Jedirlhuh
20:08.28Jedirlany new release of spandsp?
20:09.03Jedirl0.0.3 still not production, right?
20:09.13Ciber311so can anyone in here recommend a reliable voip provider for asterisk?
20:09.37*** join/#asterisk litage (n=nick@203.220.55.70)
20:09.43*** join/#asterisk darby__t (i=darby_t@aapj121.neoplus.adsl.tpnet.pl)
20:10.56justinu|laptopreliable compared to what? ILEC?
20:10.56grabowskiCiber311: VoicePulse (http://connect.voicepulse.com) seem reliable
20:11.10grabowskiCiber311: cheap for outgoing US48/Canada too!
20:13.13*** join/#asterisk obiwanmikenolte (n=obiwanmi@24-107-22-85.dhcp.stls.mo.charter.com)
20:14.37Ciber311seem? you actually use them right?
20:14.40Ciber311Download Speed: 4769 kbps (596.1 KB/sec transfer rate)
20:14.40Ciber311Upload Speed: 356 kbps (44.5 KB/sec transfer rate)Download Speed: 4769 kbps (596.1 KB/sec transfer rate)
20:14.43Ciber311Upload Speed: 356 kbps (44.5 KB/sec transfer rate)woops
20:14.44Ciber311wtf
20:14.47Ciber311Upload Speed: 356 kbps (44.5 KB/sec transfer rate)Download Speed: 4769 kbps (596.1 KB/sec transfer rate)
20:15.01filewhat's the matter?
20:15.05NDTjustinu|laptop: I have that 2850 here with me cause I brought it back from the datacenter and was supposed to drive it back down asap...heh well...the old card that was working is actually dead...doesn't get any power...so the shmuck I had put the new card in musta fucked something up..
20:15.08grabowskiCiber311: only been using them for a little while
20:15.11Ciber311accidently pasted that
20:15.23Dr-LinuxCiber311?
20:15.29NDTerr fudged something up heh
20:18.41senglandCiber311 voicepulse is really not that reliable.
20:18.54justinu|laptopNDT, :/
20:19.13Ciber311so what do you recommend sengland?
20:19.33senglandFor DID try voxbone
20:19.35grabowskisengland: have you been having problems? I have not had any of yet
20:19.40senglandThen have several outbounds
20:19.42*** join/#asterisk darby (i=darby_t@aapi69.neoplus.adsl.tpnet.pl)
20:20.15senglandgrabowski yes. They tend to have alot of circuits busy issues
20:21.42*** join/#asterisk charles___ (n=charles@fw.invosat.com)
20:21.45charles___Hey Guys
20:21.56charles___Does anyone uses  CISCO 7960 ?
20:22.25senglandgah fixed the realtime issue, Turn off rtcachefriends.
20:22.36Ciber311sengland: where are the prices? :P
20:23.10*** join/#asterisk litage (n=nick@203.220.55.70)
20:23.54ghenryIs this the way everyone else sets up Call Forwarding? http://www.voip-info.org/wiki-Asterisk+call+forwarding
20:24.00ghenrythanks, bbl
20:26.23senglandThey only handle DID not termination, I use several carriers for outbound Broadvoice, Junction Networks.
20:26.44senglandTo get DID prices you have to sign up
20:27.01senglandWhat area are you looking for?
20:27.07Ciber311nyc
20:27.23senglandarea code?
20:27.27Ciber311212
20:29.05senglandinbound DID is $9 setup 7.50 monthly unlimited inbound
20:29.20justinu|laptopthe markup on DIDs is insane
20:29.37justinu|laptopour wholesale voip provider charges us $0.50/mo for DIDs
20:30.21senglandis there an inbound per minute charge?
20:30.25Ciber311someone has to pay for the traffic :P
20:30.42senglandAnd how many do you have to buy to get that rate.
20:31.09*** join/#asterisk Assid (n=assid@210.18.143.29)
20:31.29senglandThe nice thing about voxbone is they have did's all over the world and they have a port pooling system
20:31.45justinu|laptopwell, the wholesale arrangement required a big monthly commit
20:31.47grabowskijustinu|laptop: yep its worse then getting IP addresses from a colo/dedicated company.
20:31.53justinu|laptopsomething like 20grand a month/
20:32.50senglandif you need more ports you can buy them for a region (a region is the US, Canada, South America and Europe) and they dynamically allocate them based upon realtime load.
20:32.50justinu|laptopand yeah, there's an inbound charge
20:33.03senglandjustinu|laptop thats the kicker
20:33.04justinu|laptopit varies from something sub penny/min to maybe 2.5/min
20:33.15charles___Does anyone uses  CISCO 7960 ?
20:33.15justinu|laptopdepending on the ILEC that serves that DID
20:33.23justinu|laptopthere's a whole tier chart
20:33.28grabowskicharles___: yes
20:33.48senglandSee with voxbone its a flat rate. For most small installs they just want a flat rate.
20:33.58justinu|laptopthe most expensive places to get DIDs are the smallest indepedent LECs out there
20:34.21justinu|laptopthen comes former GTE/Contel territories
20:34.24grabowskisengland: is voxbone just reselling DIDs from everywere or do they have actual connectivity everwhere they offer DIDs?
20:34.31justinu|laptopthe RBOCs are the cheapest
20:34.51*** join/#asterisk Snake-Eyes (n=blog@203.220.55.70)
20:34.59senglandRussia wants 250 a month for a single did
20:35.09senglandI dont offer service in Russia as a result
20:35.22justinu|laptop250USD?
20:35.24senglandBoth I think
20:35.28senglandYes :)
20:35.37justinu|laptopthat's a joke
20:36.17senglandIf your a Russia company you pay much less........
20:37.28senglandI have only had one issue with voxbone and they cleared it up right away. They maintain a 24/7 support staff, which most providers dont do.
20:38.10senglandAnd you can add and remove dids in realtime from the website.
20:38.48*** join/#asterisk knarfly (n=Knarf980@c-69-180-98-189.hsd1.fl.comcast.net)
20:38.59knarflyhello out there?????
20:39.07justinu|laptopi had quality issues with voxbone
20:39.12justinu|laptopthey wanted me to change to g729
20:39.37knarflyAnyone setup with FreeBSD and IPFW?
20:40.17knarflyMy setup only works halfway and I think I have a natd problem.!
20:40.32*** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net)
20:41.17knarflyCan anyone offer advice on simple sip setup to FWDNET?
20:42.40grabowskiknarfly: It's on the voip-info.org wiki
20:43.11grabowskiknarfly: http://www.voip-info.org/wiki/view/Asterisk+How+to+connect+to+FWD
20:43.27docelm0is trixter in here?
20:43.31knarflyThanks, but I have read through that numerous times and think I'm following it ok - but I can't hear anything.
20:43.33docelm0~seen trixter
20:43.45jbottrixter <n=trixter@65-165-167-217.du.volcano.net> was last seen on IRC in channel #asterisk, 1d 17h 36m 37s ago, saying: 'chino_: depends on how you connect'.
20:43.58grabowskiknarfly: when you dial a FWD number you can't hear anything?
20:44.38*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
20:44.38grabowskiknarfly: is everything going though just fine on the asterisk CLI?
20:44.50knarflygrabowski: I hear a dial tone and can dial a # and hear it ring. Then when it appears someone answers I can't hear anything.
20:45.29grabowskiknarfly: Did you try the 613 echo test?
20:45.33knarflythe CLI looks good but honestly I wouldn't really know if something bad flew by...but there are no warning messages as I attempt the calls.
20:45.40grabowskiknarfly: could it be a one way audio problem?
20:46.03grabowskiknarfly: is your asterisk system behind a NAT?
20:46.16knarflyI did try the 613 and got it to ring but could not hear anything.
20:46.46knarflyYes, I am behind another FreeBSD server running IPFWE and NATD.
20:47.10knarflyI've tweaked the rules and the redirects to some degree and gotten some of this working.
20:47.12justinu|laptop~nat
20:47.15jbotnat is probably Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
20:47.36justinu|laptoppay attention to that... externip, localnet are important to making it work
20:47.49grabowskiknarfly: I suggest you setup your FWD using IAX (which they offer)
20:48.08grabowskiknarfly: You just need to enable it in your FWD account on one of the options pages on their website.
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20:49.02knarflyI'll have to double check if I enabled it. I believe I did. But I went through the setup there too and it's still a no go.
20:51.25L|NUXcan some one tell me how can i ring extensions b/w 7am to 7pm
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20:56.07grabowskiL|NUX: http://www.voip-info.org/wiki/view/Asterisk+tips+openhours
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20:57.42grabowskiL|NUX: theres some other pages on the wiki about this type of setup, just search around
20:58.53grabowskiL|NUX: actually this page http://www.voip-info.org/wiki/view/Asterisk+day+night+mode+example is better
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20:59.38thermfhey... does Mithraen hang out around here?
21:02.00L|NUXok
21:02.09L|NUXgrabowski : thanks bro
21:02.13L|NUXi will look into this now
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21:04.34asterisk-dudhas anybody installed fax support for asterisk?
21:05.33grabowskiasterisk-dud: yes but not T.38
21:05.37*** join/#asterisk arekm (n=arekm@pld-linux/arekm)
21:06.31asterisk-dudwhat is t.38, forgive my ignorance?
21:06.48Jedirlstore&forward faxing over IP
21:07.02asterisk-dudok, i don't need that
21:07.09grabowskiasterisk-dud: http://www.voip-info.org/wiki/view/T.38
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21:07.12asterisk-dudi have asterisk 1.2.9.1
21:07.29arekmhi, http://pastebin.ca/63901 , when I call 100 it's going trough Zap/1 while IMO it should go via SIP - any ideas why? 1.2.7.1 here
21:08.21asterisk-dudi have spandsp 0.2 ref26, with all rx_fax and tx_fax and apps_makefile from same version
21:08.25anthmnever put any extens with _ in them in with any other stuff put it in it's own context and include it
21:08.30[TK]D-Fenderarekm : Change the order of _X. and your include and reload
21:08.42asterisk-dudi have trouble patching the makefile
21:09.01asterisk-dudi manually edited it and then asterisk compiled and install successfully
21:09.10asterisk-dudbut the applications are not installed
21:09.13anthmif you made a [zap] context with that _. then you would be fine cos the includes have sequential pref
21:09.20grabowskiarekm: You should read http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns so you understand why that is happening.
21:09.20asterisk-dudy is there a problem patching the file?
21:09.33anthmand _ ext in the same context always beats anything included
21:10.13arekmgrabowski: there is exactly such example there and according to it asterisk should go via sip - it doesn't here
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21:10.28arekm[TK]D-Fender: no change after order change
21:10.31asterisk-dudgrabowski: do you have any idea what could be wring?
21:10.33asterisk-dudwrong?
21:11.14anthmhmm i guess i'm wasting my virtual breath
21:11.28grabowskiasterisk-dud: I guess you need to load the modules..
21:11.46anthmarekm quick go read the detailed explanation i just wrote write in front of you.....
21:12.21asterisk-dudi have an asterisk@home box, and in the apps folder there are three different files for each app, but with the asterisk box i'm working now this is not the case
21:12.32asterisk-dudshould the three files be present?
21:12.47asterisk-dud.o, .c,
21:12.47grabowskiarekm: read http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
21:13.34grabowskiasterisk-dud: let me check what I did
21:13.36asterisk-dudi cannot figure out why it's giving me an error when i patch the makefile
21:13.41asterisk-dudok
21:13.42asterisk-dudthanks
21:13.54arekmah, now it's clear, thanks
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21:16.13grabowskiasterisk-dud: just the so file the compiled version
21:17.00asterisk-dudwhat version did u use, u remember
21:18.15asterisk-dudgrabowski
21:19.09grabowskiasterisk-dud: I just followed the entry on the wiki. I can't remember exactly what version as it was 5-6 months ago.
21:19.54asterisk-dudwell, i'm trying to follow the wiki but it won't patch
21:20.35grabowskiasterisk-dud: the diff wont patch?
21:20.50asterisk-dudmakefile
21:21.05asterisk-dudhow do i load modules or applications into asterisk
21:21.12charles___<PROTECTED>
21:21.19charles___Sorry about but I miss a SBN file
21:21.39grabowskiasterisk-dud: Just put them in the /usr/lib/asterisk/modules
21:21.57drraywon
21:22.16asterisk-dudso if i put the rx_fax.so file into that dir and reload it should work?
21:22.39grabowskiasterisk-dud: you will need to do a shutdown of asterisk
21:23.00grabowskiasterisk-dud: yes, if it does not you could add the module to the modules.conf file but it should take it automaticly
21:23.34charles___Hey Guys, anyone uses CISCO 7960 phones ?
21:25.50asterisk-dudloading modules failed
21:26.05grabowskiasterisk-dud: what was the problem?
21:26.42asterisk-dudgrabowski: undefined symbol: fax_set_phase_d_handler
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21:27.28charles___Hey Guys, anyone uses CISCO 7960 phones ?
21:27.29grabowskiasterisk-dud: try google, it brings up a bunch of results
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21:27.49asterisk-dudif i just recompile asterisk to a newer version it will ovewrite the old one, correct?
21:28.06asterisk-dudi know, i've tried most of them
21:28.49mishehuI make packages
21:28.51grabowskicharles___: It looks like no one else that is here has a cisco 7960 why don't you try again later? You don't need to keep repeating yourself.
21:29.03Ciber311i make smoothies
21:29.31mishehucharles___: yes, anyway, you might want to ask the real question and not "who uses XYZ?"
21:29.38grabowskiasterisk-dud: yep, just dont do make samples
21:29.46mishehudataja - don't ask to ask, just ask
21:30.42asterisk-dudgrabowski: do u remember which wiki u followed?
21:31.35grabowskiasterisk-dud: http://www.voip-info.org/wiki/view/app_rxfax+and+app_txfax and http://www.voip-info.org/wiki/view/Asterisk+fax
21:32.33charles___grabowski:  I know dude
21:32.38charles___mishehu: shhhh
21:34.16knarflyI want to update Asterisk on my FreeBSD server. If I just compile the new source it won't go in the right directories. Is there an easier way?
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21:35.43grabowskiasterisk-dud: I gtg, good luck getting it to work.
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21:36.56vetoIs it possible to number portability a number out of a wireless carrier (Sprint) to voip?  Is anyone setup to do this?
21:38.17justinu|laptopmy wholesale voip carrier won't let me do LNPs from wireless phones
21:39.57vetojustinu|laptop, I assume it's a PITA for the destination carrier?
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21:52.51justinu|laptopveto: i'm not sure what the issue is
21:53.02justinu|laptopwe use a major carrier tho
21:54.32vetoI find very little info on this via google, i guess not many people are trying to do it.
21:55.09justinu|laptopi would check into the laws that mandated LNP
21:55.24justinu|laptopmaybe there's a loophole that says wireless carriers don't have to participate with wireline carriers?
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22:06.36grabowskijustinu|laptop: Wireless carriers were given a LNP extension (at least here in Canada under the CRTC). I believe its somewhat the same situation under the FCC.
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22:06.59justinu|laptopthe thing is, i know you can do LNPs from wireless to wireless carrier
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22:07.25ghenryIs this the way everyone else sets up Call Forwarding? http://www.voip-info.org/wiki-Asterisk+call+forwarding
22:07.26ghenrythanks, bbl
22:07.31Dr-LinuxLNP?
22:07.38Dr-Linux~LNP
22:08.13grabowskijustinu|laptop: I think they don't have to fully comply till 2008 or something?
22:08.25[TK]D-FenderLocal Number Portability
22:08.27grabowskiLocal number portability
22:08.28grabowskiyea
22:08.52justinu|laptopgrabowski: ic
22:09.13ghenryanyone do it that way?
22:10.14thermfdoes anyone have the version of app_rxfax/app_txfax that makes use of t.38?
22:11.15grabowskighenry: What exactly do you need?
22:11.16[TK]D-Fenderghenry : Similarly, yes
22:11.53ghenryI just need to set it up for a client, and wanted to check if I was way off by following that guide
22:12.19[TK]D-Fenderghenry : Think of it as "inspiration" and you should tweak it to fit your needs.
22:12.27filecan you feel the love tonight?
22:12.33ghenryAy [TK]D-Fender
22:12.57ghenryIt's just nice to have a verbal on the stuff you read on all the * sites and books ;-)
22:13.11ghenryTo make sure it
22:13.18grabowskijustinu|laptop: Actualy the 2008 deadline might be LNP from Wireless carrier to Wireless carrier may be even longer before we see LNP from Wireless to a CLEC
22:13.19ghenry's not out of date etc.
22:13.44grabowskijustinu|laptop: or LEC.
22:13.57asterisk-dudthis wiki is telling me to find a symlink, any suggestions on how to go about doing that?
22:15.03ghenryasterisk-dud: How so?
22:16.02ghenry[TK]D-Fender: How many * rollouts does it take to know when to "tweak" or when other guides etc. read wrongly?
22:16.38asterisk-dudghenry: http://www.asteriskguru.com/tutorials/spandsp.html troubleshooting step 2
22:17.47[TK]D-Fenderghenry : I never used guides for much of anything... I find the best way if to just learn the scope of commands at your disposal and implementations will simply come to you.
22:18.00ghenryasterisk-dud: symlink --help in that directory, if on GNU/Linux
22:18.14ghenry[TK]D-Fender: COol.
22:19.07ghenryasterisk-dud: symlink will fix brokn symlinks
22:19.44asterisk-dudthnks
22:19.52ghenrynp
22:20.00ghenry.me off for a cupa
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22:28.08ghenryhow do you hook flash on an analogue phone?
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22:33.03teniarput in on hook then off hook quickly
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22:37.40knarflyfreevsd_fan: Do you have your Asterisk in full working order?
22:38.38knarflyI could use some insights into the natd settings
22:38.39Ciber311sengland: you there?
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22:52.04ghenrythanks teniar
22:52.33ghenryis that the normal way to transfer a call on a modern analogue phone?
22:52.43ghenryif that makes sense teniar ;-)
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22:53.11[TK]D-Fenderghenry : Analog has always been the same...
22:53.34teniarghenry, is it using an ATA? i know sipuras have some special code to transfer
22:53.54[TK]D-Fenderghenry : Sipuras work multiple ways.
22:54.17ghenrythis is a bog std analogue phone on a zap chan
23:02.54teniarghenry, yeah then just hook flash it
23:03.05teniarghenry, as if you were picking up call waiting
23:03.10ghenrythank.
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23:04.44Dr-Linuxlitage: :)
23:07.19Dr-Linuxanybody active and free? :)
23:09.50RoyK<PROTECTED>
23:10.27Dr-LinuxRoyK: hey there :)
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23:11.22Dr-LinuxRoyK: i wanna discuss about my new asterisk setup. if you got some time
23:11.25Dr-Linux:)
23:11.54RoyKpork?
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23:13.20Dr-Linuxpork? :S
23:13.32Dr-Linuxyou mean pig's meat? :S
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23:36.26JickDoes anyone know if a call queue can be configured such that, when an agent is called by the queue to receive a caller, a set of dialplan instructions can be execution before the call is transferred?
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23:40.14asterisk-dudwhen recieving fax i get error" TIFFOpen: :Cannot open
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23:48.57JickDoes anyone know if call queues can be configured to execute a set of dialplan instructions for the agent before connecting the agent to the next caller?
23:49.23[TK]D-FenderJick : Use Local/ agents and put it in there
23:50.26JickHmmm
23:52.11*** part/#asterisk websae (n=websae@209-252-79-66.ip.mcleodusa.net)
23:58.26JickD-Fender, okay. Let me see if I can understand how Asterisk would handle that...
23:59.21JickSo an agent would be configured at extension, say, 105 in the [agents] dialplan context...

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