IRC log for #asterisk on 20080509

00:00.14alrser, 9112
00:00.18Qwellhttp://www.ctiusa.com/solutions/customer-eng-solutions.asp
00:00.18alrsI don't know why I'm stuck o 9116
00:00.19Qwellhotel phone
00:00.20dFencedrmessano: the phones we have in our 2nd house are much more interesting for guests and the guests there are more likely to steal them (analog siemens phone with fancy lights)
00:00.38dFenceguess how many have been stolen ever: 0
00:00.45drmessanodFence: Excellent.. good luck
00:01.00dFencedrmessano: the phones getting stolen is my least concern
00:01.16JayTee52if you want to make sure none of your VOIP phones ever get stolen just buy Grandstreams.
00:01.24dFencelol
00:01.27drmessanoStolen isn't the problem
00:01.33drmessanoAbuse is.. It's been said 3 or 4 times
00:01.52dFencesame
00:01.54drmessanoBut again.. good luck on your install
00:01.55alrsdrmessano: you're underestimating the 9112, they aren't flimsy like your linksys
00:02.12drmessanoThe linksys isn't flimsy
00:02.15JayTee52so just buy 10 Grandstreams for the price of 3 good Polycoms and you'll do ok and then tack on 10 buck to the nightly room rates to cover the damage.
00:02.58drmessanoBut you're not gonna convince me a $50 SIP phone will last in a hotel
00:03.42alrsdrmessano: the $50 one was analog, the 9112s are closer to $100
00:03.57drmessanoHave you ever tried to beat a hooker with a cordless phone?
00:04.01alrsand why you think a proven design like the Aastra (nortel) can't hack it is beyond me.
00:04.13dFencedoes anyone know a phone of the linksys spa922 caliber? (needed features: interal switch & poe)
00:04.19drmessano(nortel) <-- not impressed
00:04.41alrsI see decade-old Nortel handsets all over the place
00:05.15drmessanoI see 15 year old NEC handsets all over the place too
00:05.18drmessanoDoesn't make it right
00:05.49dFencedrmessano: who's beating hookers!?
00:05.59dFenceand why the hell did noone inform me!?
00:06.49alrsdFence: Uh, Aastra 9112i?
00:06.51`SauronBecause you're lame.
00:06.58drmessanoIf you smack a hooker with an Aastra, she's gonna shiv you.. Plain and simple
00:07.33alrsdFence: though it looks like you have to go up to the 9133i to get the eth bridge
00:07.37dFencehm... never tried an aastra to be honest... gonna add that to the list ;D
00:08.58dFencethe 2nd eth-port is the n1 reason for the whole *-idea... would come way cheaper to equip every room with an ip-phone and extra ethernet port than a wifi-solution like that zyxel-stuff
00:09.33drmessanoOr you could put in a few Cisco APs
00:10.24dFencedrmessano: ill be leaving in september and once i'm gone that whole thing has to run by itself. i got panic-calls at 3am china-time because the router's lights were blinking ORANGE instead of green...
00:11.22dFencei worked my way through almost all major vendor sites, the only considerable solution is the hospitality-thing from zyxel for ~1200 bugs
00:12.10ftp3anyone know someplace besides didx that I can get a good deal on wholesale usa dids?
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00:18.17dFencewhen talking about POE does "inline" referr to "midspan", "endspan" or "go figure"?
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00:20.43dkwiebeThe Aastra phones aren't very heavy but they feel "good"
00:21.17dFencedkwiebe: is that also when whacking prostitutes or for regular use? :D
00:21.46dkwiebedfence: prostitues are for whacking? lol
00:21.55dkwiebedfence: regular use
00:22.05dFencedkwiebe: why, what did you think they're for!?
00:22.24dFencedkwiebe: ok... btw: aastra is not www.aastra.com, is it?!
00:23.03*** join/#asterisk Mavvie (n=edwin@ppp121-44-49-247.lns10.syd7.internode.on.net)
00:23.05dkwiebeyes it is
00:23.18dFenceaaaa *panic*
00:23.26dkwiebelol, what's up?
00:24.49dFencedunno the exact organization of aastra but in germany they're somehow related with detewe... had 2 detewe devices, both ended up costing us twice as much for maintenance as product-price itself
00:24.56dFencehm... should lay off the vodka
00:25.35dkwiebelike a maintenance agreement.  You don't need those with their devices in Canada at least
00:36.00unpaidbillso do coupon codes actually exist for the digium store
00:36.14unpaidbilli want to get a discount on this here codec
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00:48.54unpaidbillyea that's what i thought
00:51.11drmessanoDid you just ask for a discount code for buying G729?
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01:01.01logi4023calls to my sip softphone are generating call progress (ringing signal) to the caller. Anyone knows why this is happening?
01:01.10logi4023are not generating
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01:15.40VoiceCXdo you guys ever think that someone may write a module for your trixbox or FreePBX to login to Sirius Radio Online to be used as a MOH module
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01:27.12profoundedhey quick question, i have a server setup in a remote location and i want to connect a sip phone to it.  I would think it would be a matter of opening up port 5060 for udp and tcp and then port 1252 (which is the port that shows up when i run sip show peers).. what am i missing?
01:28.03plik10000-20000 udp for rtp
01:28.42profoundedthanks plik
01:28.57plikwelcome
01:35.28profoundedplik, the sip phone can be behind a firewall, its just the server that needs to have the ports open correct?
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01:39.48lanningprofounded, yes, just the server (the sip phone's firewall must allow general outbound traffic, as usual)
01:40.02profoundedthx lanning
01:40.02lanningalso, you don't need port 1252
01:40.38profoundedfigured, just added it because i was stuck, ty
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01:50.22ManxPowerVoiceCX: Most of us hate TrixBox and FreePBX
01:53.21BeeBuuhate for what?
01:53.25*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
01:54.06ManxPowerBeeBuu: I'm sure each does for their own reasons.  Personally, I hate them because their users come here and end up disrupting the channel when their users should be using the product support methods for the product they are using.
01:54.24ManxPowerAlso, only girlymen use a GUI
01:54.48JayTee52lol
01:55.56JayTee52I need help installing AsteriskNOW on Red Hat 3 but I can't find all my floppies, can someone help me? <<<< Joke example
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01:58.41ManxPowerSomething like that.
01:59.00JayTee52ManxPower, were you in here the other night for that?
01:59.22outtoluncnash!
01:59.32JayTee52hehehe
01:59.34ManxPowerThere are many things to dislike about the GUI Asterisks, but the one that really gets my tail fur in an uproar is their users on this channel being offtopic.
02:00.19ManxPowerIf they don't want to support their product then they should close the channel and everyone can come here for all forms of Asterisk.  But I'm pretty sure they don't want their channel flooded with Asterisk specific questions either.
02:01.28JayTee52do the GUI based systems that are based on Asterisk still used the .conf files or have they replaced them with database or XML?
02:01.30errrIm having an issue with my voicemail. When it shows up in my email inbox the message attachment is a 0 size but it plays just fine and is not 0 size on the pbx
02:01.33ManxPowerLazyness is not a reason to come to a channel and ask off topic questions
02:01.53JTJayTee52: freepbx uses a sql backend which generates .confs
02:01.57errrany idea why I keep getting a 0 sized file in email instead of the actual message
02:02.04ManxPowerJayTee52: Everyone that I've heard of has used complicated sets of include files and incredibly complicated scripts, macros, and AGIs
02:02.37ManxPowererrr: Whatever is used to attach and build the message did something wrong 8-)
02:02.45JayTee52my boss has a hardon for GUI's and keeps pushing me to find one for *. I keep telling him that it's not worth the effort.
02:02.57ManxPowerI take it you have examined the full headers and MIME structure of the message?
02:02.57errrManxPower: where can I kick it to make it do the right thing?
02:03.14ManxPowerJayTee52: Is your boss a geek or a geek wannabe?
02:03.53ManxPowerIf so, challange him to build an Asterisk system prototype.
02:04.01JayTee52my boss is an idiot that thinks Microsoft rules and is constantly bitching about their stock price. He wants to roll out Vista to everyone even though it won't run half our legacy apps.
02:04.03ManxPowerUsing A GUI
02:04.16ManxPowerThen try to get all of the features of your current Asterisk to work in the GUI
02:04.28JayTee52he also believes the world is 6000 years old and that dinosaurs coexisted with man.
02:04.39ManxPowerJayTee52: He could either be an idiot or a geek wannabe
02:04.52ManxPowerJayTee52: You are serious?
02:04.59JayTee52his ego is constantly writing checks his intellect can't cash.
02:05.11JayTee52yeah, he's born again hard core
02:05.11ManxPowerThen perhaps he would take a challange
02:05.29ManxPowerHonestly, I could not work with such a person.  I'm a Devout Atheist.
02:06.03errrJayTee52: use freepbx to give it a gui then just do all the work in the conf files removing the files the gui will edit. If he just wants something to look at that will keep him busy ;)
02:06.05JayTee52then we have half a thing in common, I'm just not devout about anything
02:07.03JayTee52I wish sipX had support for IAX2, then I could have all my phones register to it and connect to Exchange UM and use * as my main server and PSTN gateway.
02:07.35ManxPowerI'll tolerate people's beliefs but if they want to flash it in my face, I shall have a hard time keeping quiet.
02:07.59errrManxPower: I dont know much about email headers but these seem to be right and there is a scetion for the msg001.wav in them
02:08.02ManxPowerJayTee52: Why not use SIP
02:08.15ManxPowererrr: pastebin the headers and I'll take a look
02:08.18errrok
02:08.25JayTee52he was born and raised in Indiana but want's to join the Sons of the Confederacy because he is such a civil war buff and refers to it as "The War of Northern Aggression"
02:08.34JayTee52ManxPower, I do use SIP
02:08.47ManxPowerJayTee52: I moved to the south from the north and MANY people down here call it that.
02:09.16JayTee52I use SIP now to go between UM and * but I have to use sipX to transform UDP to TCP for UM because UM only speaks TCP.
02:09.46ManxPowerOh!  Yes, I remember something like that being said somewhere.  One of the stupidest things I've heard of in *years*
02:10.35ManxPowerJayTee52: look at the 1.6 featureset.  I'd not use it in production yet, but you can play with it and set up a prototype system.  When 1.6 is stable, you will already know all you need to know about and I think 1.6 may support TCP
02:10.55JayTee52if I could figure out how to do a transform from UDP to TCP using * 1.6 which supports SIP/TCP I'd do it to replace sipX
02:11.23JayTee52ManxPower, I've already got a box setup with 1.6 but the docs are kinda scant
02:11.51ManxPowerI can see the reason to use SIP/TCP.  SIP is just call setup and teardown, that can be delayed slightly and should be reliable.
02:12.09JayTee52according to the comments in the sip.conf file for 1.6 it does support sip tcp and tls
02:12.18*** join/#asterisk andrewn (n=andrew@76-191-151-229.dsl.dynamic.sonic.net)
02:13.42JayTee52I don't mind the Web GUI setup for sipX and I think it would be great for a non-technical person to use to administer the phone side of things but sipX won't use Digium hardware and I've already got 2 TDM04B cards and a TE212P card invested.
02:14.11errrManxPower: http://fluxbox.pastebin.ca/1012305
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02:17.05JayTee52I only have 4 more DVDs to watch to complete the entire 5 season Babylon 5 series.
02:17.13gitguylooks like the problems i had this morning with the pins and stuff were bandwidth problems
02:17.34ManxPowerJayTee52: A nontechnical person will screw it up.  Even if you train them, that training will eventually be lost along the long string of people that try their hand at admining the pbx
02:17.39ManxPowerBut I'm not cynical or anything
02:17.53[TK]D-FenderJayTee52, I borrowed the complete series and first 5 movies off a friend and had another rip it & I burned them.
02:17.57gitguynow i don't lose digits anymore with Read() and NoOp()
02:18.04ManxPowerYou have to *understand* telecom to admin any pbx
02:18.04[TK]D-FenderJayTee52, Working my way through 5 years of Andromeda right now :)
02:18.45JayTee52oooh! I loved that show!!!
02:18.51gitguyManxPower: was that for me?
02:19.04[TK]D-FenderJayTee52, I getting to.  B5 sets the bar VERY high though...
02:19.20JayTee52I got the first season of Babylon 5 used for 24 bucks then found the other seasons for 19.99 on sale at Best Buy
02:19.40errrManxPower: so were the headers formed correctly?
02:19.52JayTee52my all time favorite series only lasted 1 season. Firefly
02:20.26[TK]D-FenderJayTee52, thats next on my list to watch.  Everyone says its awesome.
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02:21.01JayTee52[TK]D-Fender, the lead actor is a Canadian. Nathan Fillion, he's from Edmonton, Alberta
02:21.37ManxPowererrr: If you use WAV49 format the files will basically be GSM (MUCH smaller) wrapped up in a .wav file that windows media players can play
02:21.37errrManxPower: I tried using that the attachments are still 0 size
02:21.51[TK]D-FenderJayTee52, "thats nice" :)  Don't care where they're from, only that they are good at what they were brought in to do.
02:21.58ManxPowererrr: What mal reader are you using?
02:22.07[TK]D-FenderJayTee52, loved Farscape too.... miss that one.
02:22.16JayTee52it's the characters and dialogue that make Firefly such an awesome show
02:22.26errrManxPower: thunderbird. but even in the gmail web interface they are 0 size
02:22.31JayTee52hehe, yeah. Friend of mine named her dog Frell
02:23.18ManxPowererrr: I would have to refresh my limited memory of MIME but I don't know if you are supposed to have duplicate boundary tags, other than that it looks perfectly good to me
02:23.54errrManxPower: if I use mutt from the cli to send the same voicemail it works fine and the message is not 0 size.. its only when asterisk sends it
02:24.00errrthat it is 0 size*
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02:24.40ManxPowerFigure out what the difference between the two messages are
02:27.03errrManxPower: the only diff is that the attachment is not 0 size so the header is much much larger sice it has all the data in it
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02:27.56*** mode/#asterisk [+o putnopvut] by ChanServ
02:38.39*** join/#asterisk zackz (n=zack@64-136-221-136.dyn.everestkc.net)
02:40.48zackzis there a way i can make a "dialplan reload" reload like a | more in linux?
02:42.44JayTee52zackz, why not just connect to the * server using putty or some other ssh client. The terminal window will be scrollable.
02:43.30zackzi cant scroll back far enough to see everything
02:44.00rob0I don't think the asterisk console has a pager command. Increase your terminal's scrollback buffer.
02:44.57errryou could > the putput into a file
02:45.01errroutput*
02:45.19errruse asterisk -rx then redirect to a file
02:46.37zackzerrr: i did that but nothing shows up
02:46.54profoundedim having a hard time getting asterisk to work with a remote sip phone.  whenever i dial a number using the remote sip phone, i get an echo on the phone number i called.  does sound like an rtp issue?  i cant seem to configure port forwarding correctly if this is the case and im blaming it on my router
02:47.20errrah you are right
02:47.34zackzi also tried piping it to tee but nothing hsows up that way either
02:49.27zackzand i odnt see anywhere in putty to increase the buffer DOH
02:49.40profoundedputty can be increased
02:49.54profounded1 sec
02:50.39profoundedright click toolbar, change settings, window, and in there the line numbers
02:50.58profounded"lines of scrollback"
02:51.45zackzyoure right
02:52.16zackzprofounded: wha tkind of router do you have
02:52.52ManxPowerEcho has nothing to do with any of that.  Echo must be removed where the call is converted between PSTN/VoIP
02:53.50errrManxPower: http://fluxbox.pastebin.ca/1012341 my logs seem to be indicating that app_voicemail may be trying to send a file that is not there instead of using the temp  file that was actually created..
02:53.51profoundedversalink/westel clone d90-327... verison dsl router wifi point
02:54.45zackzya i ManxPower is right
02:54.50profoundedmanxpower: i meant "echo test", when i speak into the phone that i called, my voice is repeated to me
02:55.12profoundedi cannot hear the other party
02:55.30*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
02:55.35lanningyou mean one way audio
02:55.50profoundedyeah, it loops back
02:56.25profoundedonly for the receiving end, i get nothing for the party that made the call
02:56.27lanningare you calling an echo test extension?
02:57.07profoundedno, works fine when i am inside the network, making me think its a port forwarding issue
02:57.11zackzone way audio is your router blocking rtp ports
02:57.53drmessanoor NAT not set correctly
02:58.18zackzasterisk uses 10000-20000 for rtp i believe
02:58.21zackzudp
02:58.26zackzactually you can set it
02:58.30profoundedthe router first of all has a retarded dmz option that doesnt work, so i cant make use of that and use a better router/firewall
02:58.34zackzto whatever you want but that is default
02:58.39drmessanoLOL
02:58.47drmessanoediting rtp.conf is difficult?
02:59.08profoundedi purposely set it to 10000-10010 in rtp.conf and port forwarded all those ports using udp just to test and still wont work
02:59.13drmessanoforward 5060 UDP and whatever is in your rtp.conf
02:59.23drmessanoThen set localnet and externhost/externip
02:59.27drmessano~nat
02:59.28jbotwell, nat is Network Address Translation  Usable in Asterisk sip.conf file with externip, localnet, and localmask setup properly.  See docs.
02:59.51drmessano~sipnat
02:59.52jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
02:59.56drmessanoThere you go
03:00.58profoundeddrmessano, ill read into it, ty.  i think im missing the set localnet and externhost/externip part of the equation. thx
03:03.09drmessanono probs
03:04.34JayTee52well, I've been debating whether I should go to bed or install Red Hat 3 on my 486/33mhz system with 64MB of RAM and then compile Asterisk 1.2 on it.
03:04.59drmessanoGot Floppy?
03:05.09JayTee52got all my floppys :-)
03:05.29VxJasonxVI've created extensions by username. Is there a way, in the users.conf, to "alias" a numerical extension?
03:05.38VxJasonxVOf is that something that would have to be designed in the dialplan?
03:05.55MooingLemurdialplan. :)
03:05.55VxJasonxVs/Or/Or/*
03:05.59VxJasonxV:(
03:06.11VxJasonxVof or blah
03:06.53VxJasonxVthat sucks, and is exceedingly annoying
03:07.13rob0JayTee52: Bedtime for you young man.
03:07.19JayTee52yep
03:07.26JayTee52nite all
03:07.30MooingLemurVxJasonxV: I'd do it the other way around
03:07.31rob0:)
03:07.38VxJasonxVhmmm
03:07.42MooingLemurextensions by number, and name as the alias in the dialplan
03:07.49JayTee52a young man who can actually remember when Eisenhower was President :-)
03:08.05MooingLemurhaha
03:08.14JayTee52I turn 54 a week from this sunday
03:08.29MooingLemurso the 52 isn't quite right
03:08.40JayTee52it was when I registered
03:08.46MooingLemureither as the year or age
03:08.57MooingLemur(I would have guessed the year)
03:09.13JayTee52I wanted to just have jaytee like I do on Blitzed but Freenode ops are making me wait. the guy who has it registered hasn't logged in for over 12 weeks.
03:09.36MooingLemurhah
03:09.43JayTee522 more days and I'm asking them again
03:10.05JayTee52hey, I have lemurs at my work
03:10.16MooingLemurI bet they don't moo.
03:10.20JayTee52ring tailed adn red ruffed
03:10.25JayTee52nope they don't moo
03:10.27rob0was born in Kennedy presidency
03:10.40JayTee52he was a good President. Inspiring
03:10.41MooingLemurCarter :(
03:10.47JayTee52not so much
03:10.53MooingLemurI'm young in this crowd :P
03:11.04JayTee52nothing wrong with that
03:11.39JayTee52just don't burn the candle at both ends all the time, try to save some wick for when you get to be my age
03:12.46MooingLemurI think my stress levels are reasonable most of the time :)
03:12.52JayTee52anyhow, hope all of you have a pleasant evening and a great day tomorrow. Good nite!
03:13.01MooingLemurtake care
03:13.04drmessanoOhhhhh "wick"
03:13.05MooingLemurd'oh
03:13.09MooingLemurhahah
03:13.37*** join/#asterisk nighty^ (n=nighty@210.188.173.246)
03:15.57zackzhahahaha, i was watching FSN, and there was a message that came on that said "due to the length of the previous program, we now join the regular program already in progress" and it went to colored bars and tone
03:17.56Braxuslol
03:18.45profoundedwow drmessano, that did it!! thank you very much!
03:20.30zackzi went through two different linksys routers trying to run sip over vpn tunnel, neither one worked, im never buying linksys again
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03:24.30mmlj4this is definitely a strange twist: VoIP over ham radio: www.arrl.org/qst/2003/02/VoIP.pdf
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03:26.36gitguycan asterisk do sip over tcp?
03:27.10mmlj4yes, if you like lag and stuff
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03:27.48bkw__mmlj4: wtf you talking about sip over TCP? and lag? you are talking out your ass
03:27.59bkw__TCP is a MUST in the RFC not an optional thing
03:28.27mmlj4bkw_: fragmentation for starters... there's a reason why UDP is preferred
03:28.35bkw__fragmentation?
03:28.44bkw__mmlj4: you seem to not know what you're talking about
03:28.46mmlj4never mind
03:28.56bkw__let me outline the dumbest thing in sip
03:29.00bkw__the 64k thing
03:29.03mmlj4one of use seems that way, yes
03:29.10bkw__if the packet is over 64k it falls to TCP if it can't then it falls back to UDP
03:29.21bkw__thats the dumb one
03:29.26bkw__but their is NO lag on in TCP vs UDP
03:30.15bkw__the other nice thing about TCP is the connection stays open
03:30.25bkw__which can help with nat in some cases
03:30.46*** join/#asterisk dawebber (n=dawebber@adsl-75-24-187-185.dsl.ltrkar.sbcglobal.net)
03:31.00bkw__does asterisk do TCP yet?
03:31.19bkw__I did know their was a patch for it but haven't been following to see if it was added
03:31.33mmlj4gitguy: as i was saying, sure, SIP over TCP works, but you won't want to use it, the practical quality verses UDP just isn't there
03:31.54bkw__again
03:31.56bkw__talking out your ass
03:32.12bkw__mmlj4: please go get a clue about it before you start yapping your trap
03:32.37mmlj4bkw_: please list one streaming protocol in widespread use that does TCP
03:32.53bkw__again you have no clue what you're talking about
03:32.53mmlj4this is silly
03:32.57bkw__SIP is Signaling
03:32.59bkw__it goes over TCP
03:33.02bkw__the Media is still UDP
03:33.14mmlj4now you're starting to make sense
03:33.22bkw__apparently you missed that fact
03:33.34bkw__TLS is always TCP till DTLS is standard
03:33.41drmessanoWho gives a crap about SIP quality?
03:33.47drmessanoSIP is just signalling
03:33.54drmessanoIt can be so-so for all I care
03:33.56bkw__drmessano: I think he was confusing media with signaling
03:34.00drmessanoGive me reliable RTP
03:34.06bkw__drmessano: then you need SCTP
03:34.15bkw__but reliable media is one of those tricky things
03:34.19bkw__you have to accept losses and move on
03:34.27drmessanoThats not my point
03:34.51drmessanoArguing over the quality of SIP is stupud
03:34.55drmessanostupid too
03:35.02bkw__tcp vs udp isn't about quality
03:35.18drmessano[23:26] <mmlj4> bkw_: fragmentation for starters... there's a reason why UDP is preferred
03:35.25drmessanoThats not what I was reading
03:35.46bkw__drmessano: read the RFC it has this really stupid rule about packets over the MTU
03:35.56*** join/#asterisk cmantito (n=gphreak@pool-71-188-82-138.cmdnnj.east.verizon.net)
03:36.02drmessanoI have better things to do than read the RFC
03:36.15bkw__thats fine they are mind numbing
03:36.23*** part/#asterisk dawebber (n=dawebber@adsl-75-24-187-185.dsl.ltrkar.sbcglobal.net)
03:36.23bkw__but i you implement anything you had better know them well
03:36.58drmessanoDeploying and implementing are two different things
03:37.07bkw__well the other thing is TCP isn't optional
03:37.12bkw__its a requirement .. aka a MUST
03:37.22bkw__you can't optionally support TCP if you wish to be compliant
03:37.23mmlj4yes, i knew SIP was just signalling, but yes, i got them confused, my bad
03:37.32bkw__mmlj4: thats ok.. its late
03:37.40drmessanoHmm
03:37.48drmessanoNot sure how that matters here
03:38.06bkw__TCP has a few things going for it.. the reliable delivery is one
03:38.14drmessanoIm pretty sure my phones don't care that Asterisk hasn't supported TCP until now
03:38.15bkw__and sip info over TCP is going to be ordered properly
03:38.16zackzUDP is faster becuase it doesnt check to see if the packet was delivered
03:38.30bkw__zackz: not 100% true
03:38.38zackzwell, the protocol doesnt check at all
03:38.38bkw__you can't tell speed differences in UDP vs TCP
03:38.42zackzyou can do it in software
03:39.00bkw__the bit thing is the connection being open and active all the time is a plus in some cases
03:39.07bkw__and TCP is good over laggy satellite links
03:39.12bkw__where UDP would fail
03:39.29bkw__each has some advantages
03:39.37bkw__having both is nice
03:39.49zackzudp is not generally used for data that needs to be 100%
03:39.57bkw__that is true
03:40.06bkw__but i'm not sure about you but I like my phone calls going thru 100%
03:40.12bkw__:P
03:41.49*** part/#asterisk bkw__ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net)
03:42.06zackzyou must not own a cell phone then :)
03:42.30drmessanoHe's a troll anyway
03:42.51drmessanoI was waiting for the FreeSWITCH rhetoric
03:43.48fileyawns
03:43.55mmlj4who? not me, I hope
03:44.48drmessanoNo
03:45.05drmessanoThe one that just PM'ed me to warn me he has eyes everywhere
03:45.17drmessanoand to "watch it"
03:45.24filedrmessano: hrm?
03:45.53drmessanoYeah
03:46.15rob0I'm an Asterlink customer, very pleased with it, so I stay out of the fights. :)
03:46.27mmlj4ok, so s/SIP/IAX2/ and I'll restate my argument
03:46.28drmessanoI was also told that this channel is logged publicly
03:46.33drmessanobecause, you know.. I never knew that
03:46.36drmessanoOh, wait..
03:46.59gitguydidn't bkw contributed to asterisk?
03:47.09rob0quite a bit IIRC
03:47.10mmlj4it is logged, just about all networks/channels of consequence were stealth-logged for years
03:47.12filehe did, long ago
03:47.16drmessano[23:41] <bkw__> troll I am not... better watch it.. I have eyes every where
03:47.29jblackso do potatoes.
03:47.54drmessanoand it's "everywhere"
03:47.57drmessanoone word
03:48.01zackzeverwhare
03:48.02rob0I will say it makes sense to me to use TCP for the control channel, but I don't know.
03:48.14mmlj4rob0: I'd agree
03:48.17drmessanoWell
03:48.23drmessanoAsterisk has been around for........?
03:48.30zackz1999
03:48.30fileawhile.
03:48.32rob0the media pretty much HAS to be udp
03:48.48Corydon76-digHe unfortunately burned his bridges when he left, and only relatively recently has tried rebuilding them
03:48.49drmessanoand I haven't heard a lot of cries of "ZOMGGGGG MY CALLS ARE DROPPING BECAUSE ASTERISK USES SIP UDP AND NOT TCP"
03:48.53drmessanoSo...
03:49.15Corydon76-digUnfortunately, nobody wants want to be on the bridge again when he sets fire to it
03:49.16drmessanoI welcome the addition of TCP to 1.6... but come on..
03:49.56drmessanoActing like "Yay, Asterisk finally supports the RFC" really discounts the fact that Asterisk has been kicking ass for years
03:50.23rob0googles for fire resistant hang gliders
03:50.36Corydon76-digThat's because as a design goal, we have interop above strict compliance
03:51.11drmessanoCorydon76-dig: But aren't people whining about the RFC much more important than asterisk actually working??
03:51.25*** join/#asterisk bkw__ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net)
03:51.47bkw__get this straight I don't care to rebuild any bridges with anyone.. just help people build really kick ass things related to telephony
03:51.52bkw__NEXT!!!
03:52.13bkw__I"m glad I burned them.. or C4'ed them.
03:52.48bkw__anyway  l8tr
03:52.48*** part/#asterisk bkw__ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net)
03:53.13Corydon76-digand he wonders why people are reluctant to work with him
03:53.30drmessanoO.o
03:53.40Corydon76-digfeeds the troll some more
03:53.54fileis glad he defected
03:54.03*** join/#asterisk tengulre (n=tengulre@125.71.208.16)
03:54.22drmessanojust wants someone to build a good Win32 PBX system he can run on Windows 98
03:54.28rob0haha
03:54.34drmessanoCan't we all just get along for a good cause?
03:54.34rob03.1
03:54.53Corydon76-digthought Win98 and "good" were opposites...
03:55.10zackzwin98 is better than vista
03:55.15drmessanolol
03:55.22jblackso, is bkw a loser now?
03:55.30jblackis he still contributing at all?
03:55.42Corydon76-digHe is not contributing to Asterisk at all, no.
03:55.45filehe does not contribute any longer
03:56.13rob0they're working on freeswitch?
03:56.14Corydon76-digI wouldn't call him a loser, though.  Diva, yes.
03:56.27filenods to rob0
03:56.33Corydon76-digPrima donna, you betcha.
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03:58.59keith4__so, I can get an extension to play out the console using ALSA, no problem. how do I play DTMF or say digiits or playback a gsm file out the console?
03:59.01paulproteusIs it possible to get a US-based service provider that allows me to receive SMS messages to a phone number served by my Asterisk?
03:59.47*** part/#asterisk zackz (n=zack@64-136-221-136.dyn.everestkc.net)
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04:06.10logi4023calls to my sip softphone are not generating call progress to the caller. anyone knows why?
04:06.23logi4023using dial() cmd.
04:06.34logi4023softphone is xlite
04:06.53Corydon76-digkeith4__: just create a new extension with what you want to do
04:07.10Corydon76-digkeith4__: and you can create extensions from the CLI, too
04:08.27Corydon76-digkeith4__: "dialplan add extension....."
04:12.11keith4__The extension alread exists. Currently, if I call '88', Console/dsp auto-answers. and anything I say into the phone, comes out line-out of the soundcard. All I have at extension 88 is Dial(Console/dsp) and then Hangup
04:12.16keith4__this setup works fine
04:12.50keith4__but I want to play a tone over Console/dsp too
04:13.05Corydon76-digkeith4__: Take a look at the AMI Originate command
04:13.18Corydon76-digkeith4__: or the CLI originate command
04:18.17Nasraany1 know of a video for instruction to install and running Asterisk from scratch with procedures since I am new ?
04:19.17LARefugeeNasra:  See youtube
04:19.36Nasrayoutube?
04:19.45LARefugeeyoutube.com
04:20.05Nasrawell....let's just give a try then....
04:20.08Nasraand thanks alot
04:20.42mmlj4Nasra: or you can use trixbox or another of the cute-and-fuzzy distros that basically Just Work and are easy to administrate
04:21.01[TK]D-Fendervideo?  Ancient useless junk.  Only one I recall was a revision3 thing from over 3 years ago
04:21.15Nasrawhat I want is to learn wih my own mistakes from scratch
04:21.21LARefugeeHere's one: http://youtube.com/watch?v=SQb71Y_X4yo
04:21.33[TK]D-FenderRead the book that people here kindly contributed 2 editions of and we can download freely.
04:21.35Nasraokay
04:22.14NasraD-Fender I am reading the book that is why I don't participate in the chat just reading..
04:22.22Nasralearning alot though
04:22.59mmlj4reading is fundamental]
04:23.06Nasraand the needs more diagrams for newbies
04:23.21drmessanodiagrams?
04:23.37Nasralike pictures etc....
04:24.15drmessanoWhy does everyone find it necessary to define terms when the validity of the use of the term was questioned, not the meaning of it.
04:24.53Nasradrmessano: there are ppls like me....
04:25.04adeelis it possible to hide the notify updates in the verbose CLI output? e.g.logger.c    -- Extension changed 6014 new state Idle for Notify User 6036
04:25.26drmessanolower your verbosity?
04:25.35adeeldrmessano, i have it on 3
04:25.41drmessanoadeel: try 2
04:25.52drmessano2 < 3
04:26.33adeeli wish there was a table that indicated at what verbose levels different components will log to
04:26.46Nasrammlj4: just burned centOs 5.1 to test and learn....so I am getting there slowly....currently running Ubuntu 8.04
04:26.46adeeli'm sure i can figure it out by diving/grepping through the source
04:27.23adeeldrmessano, seriously? 2<3?? wow...that explains why i failed 2nd grade multiple times...i always thought 2>3 ....next thing you're going to say is that 1+1 doesn't equal 3!
04:27.26adeel=cp
04:27.42keith4__ugh. centos
04:28.03drmessanoadeel: I truly believe you
04:28.12adeeldrmessano, hahahah
04:28.17drmessano;)
04:28.37drmessanoand for the record
04:28.47drmessanoI don't think boolean terms came until 3rd grade
04:28.59adeeldepends on the city/county/state
04:29.05drmessanoWell...
04:29.06*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
04:29.15adeeli've gone to a lot of different schools
04:29.34adeel10 different schools in 12 years
04:29.41*** join/#asterisk Lin (n=igormorg@unaffiliated/lincity)
04:29.57drmessanoIn NJ, where the school system I was in was selected as one of the top 6 in the nation, it was 3rd grade... here in Georgia, which is 48th in Education, I think that was a college prep class in 11th grade
04:29.57Lingood morning(ugt)
04:30.14adeelhell, in 9th grade we were covering decimal places again and people were failing those tests
04:31.07adeelalthough, to be fair, that was in Canada
04:31.40drmessanoIn NJ, we had school system that had a school set up for all of the 6th graders in the township, so Jr High was 3 years, high school ended up being 3 years..   I moved here in 9th grade.. the first week.. I went from being a "senior" in Jr High to a real HS Freshmen...
04:31.51drmessanoWell... First week I am here, we have an english test
04:32.05drmessanoFrom the McGraw Hill 7th grade english book
04:32.13drmessanoSaid it at the bottom of the page
04:32.16drmessanoI cried
04:32.18adeelhahaha
04:32.47adeelhigh school was a joke, and i was in the best school in the state
04:33.03drmessanoI slept from 10th grade thru 12th grade
04:33.13adeellikewise
04:33.29adeelbut even then, i STILL can't * to behave properly with DTMF
04:33.31drmessanoI graduated with a 69.something average.. which was barely passing..
04:33.46drmessanoand now I make more money than most of the people I went to school with....
04:34.28adeelno offense, but that typically isn't that difficult...hell call girls make more money than most people
04:34.34drmessanolol
04:34.35adeellike the chick with the mayor of new york
04:34.43adeelerr governor
04:35.22drmessanoWell, my inadequencies make it necessary for me to bolster my self esteem with comparison of monetary compensation
04:35.34adeelfair enough
04:35.36drmessanolol
04:36.01adeelseriously though....on a  PURE SIP connection, * will randomly ignore some DTMF inputs
04:36.07adeeland will log that it's ignoring them
04:36.45drmessanoParticular ITSP?
04:36.50adeele.g. [May  6 13:22:56] DTMF[29746] channel.c: DTMF begin ignored '5' on SIP/6010-b68f8508
04:37.07adeelwell i don't think it's an ITSP problem
04:37.29adeelthese are all being done on Polycom 601's or 330's with the latest firmware
04:37.46adeelusing RFC2833
04:37.50filewhat is the channel doing?
04:38.20adeelfile, nothing out of the ordinary...there's nothing in the sip history or messages that seems incorrect
04:38.32filelet me rephrase...
04:38.34adeeli think it's just that * ignores some inputs
04:38.48fileis it two channels bridged together? it is in an IVR?
04:38.48fileerm is it in
04:38.58adeelohh, 2 channels bridged
04:39.01adeeloutbound call
04:39.09fileboth sides set to dtmfmode=rfc2833?
04:39.42adeelone side is rfc2833 and one side is set to auto
04:40.08adeelbut other DTMF signals are passed through
04:40.11adeelin the same call
04:40.23fileare the DTMF digits being hit fast?
04:40.45adeelthe users claim they're not
04:41.00adeelbut i've looked at the minimum time, and it's 40 ms....there's no human way to do get that speed
04:41.04adeelunless polycom is buffering them
04:41.13filedoes it show it receiving an end DTMF?
04:41.18adeelyes
04:42.18filefile an issue report with full log with the DTMF logging and I'll look at it tomorrow
04:42.44adeelsure
04:42.46adeelthanks
04:43.58*** join/#asterisk s0lid (n=s0lid@210.213.198.7)
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04:48.23drmessanoI guess now is a good time to pack up the asterisk source and post it on Bittorrent
04:49.04drmessanoAsterisk_PBX_SOURCE_CODE_LEAKED-ZoMg
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04:55.15[TK]D-Fenderdrmessano, * has tons of leaks as it is... most of them memory....
04:55.31adeelhaha
04:56.30drmessanoIt also has limited hardware support.. for example, I still can't use my USR Robotics Sportster 14.4 WinModem as an FXO device
04:56.33drmessanoThat... is lame
04:59.18keith4__any way to play audio out Console/dsp from the dialplan?
04:59.31keith4__perhaps during a call to Console/dsp ?
04:59.45*** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl)
05:01.02adeelthere's supposed to be
05:01.10unpaidbillwhat the heck... how does asterisk pick the codec to use, is by the order they're allowed or is there something else?  it's defaulting to ulaw if i put both ulaw and g729 in the sip config
05:01.23unpaidbillbut if i have only g729 i cant call my other asterisk server because of no compatible codecs
05:01.40unpaidbilland i want it to use g729 when it dials to the voip provider :/
05:02.03unpaidbillbut fall back to ulaw when dialing that provider if the endpoint doesnt support g729!
05:02.04adeelunpaidbill, are you using sip?
05:02.08unpaidbillyeah
05:02.16adeelunpaidbill, the order in which they're listed in sip.conf matters
05:02.28adeelso try listing g729 before ulaw
05:02.36unpaidbillwell that is confusing me because it always picks ulaw, and ulaw is listed after g729
05:02.43unpaidbillif i only allow g729 it uses it
05:03.43adeelunpaidbill, i'd recommend double checking to make sure something else isn't over-riding the g729...otherwise, it could be the endpoint tries to force ulaw rather than g729
05:03.48keith4__adeel: can ChanSpy or ExtenSpy play sound out one end of a call?
05:04.19adeelkeith4, i'm not sure i follow...'play sound out one end of a call'?
05:04.20*** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net)
05:04.43keith4__uh... let's say I pickup a zap channel and dial an extension that just dumps to Console/dsp, which is set to autoanswer
05:04.55keith4__now I have a call to the soundcard's audio out port
05:05.02adeelsure, you can do that
05:05.09keith4__I am doing that. it works perfectly
05:05.31adeelok, so then what are you trying to do?
05:05.52keith4__but now I want to, maybe... SayDigits() out the audio port
05:06.02keith4__or, play an mp3 out the audio port
05:06.48keith4__like, to announce a page or something
05:06.58adeelkeith4, while a call is active?
05:07.15keith4__yes
05:07.24adeeli mean, while you already have sound being dumped to Console/dsp?
05:07.27keith4__or before it's active
05:07.43adeelif the soundcard is not being used, then you should be able to pipe anything you want to it
05:07.57keith4__oh, like just use a system call?
05:08.06drmessanoyes
05:08.16keith4__hmmm
05:08.17adeelif it is in use, then you'll probably need to setup DMIX, if using alsa,
05:08.25keith4__yes, using alsa
05:08.30*** join/#asterisk [hC] (n=hardcore@190.10.9.126)
05:08.39keith4__i was just about to ask about a possible contention problem...
05:08.40[hC]is it possible to get asterisk 1.2 chan_sip to listen on two ports?
05:08.54adeelotherwise you run into the typical contention problem
05:09.28keith4__http://alsa.opensrc.org/DmixPlugin says dmix is enabled by default
05:09.37adeelkeith4, but i'm not too sure how *s internals are in terms of console audio
05:09.48*** join/#asterisk mikong (n=gones@203.193.37.251)
05:09.50adeel[hC], you can try setting the port= value again
05:10.25adeelkeith4, best advice is to try it out...i don't really foresee why you shouldn't be able to do what you want
05:10.27[hC]adeel: that doesnt work.
05:10.30[hC]adeel: it just takes the first one.
05:10.47adeel[hC], hmmm....well unless you're able to bind to another IP then i don't think it can listen on 2 different ports
05:10.57[hC]iptables to the rescue.
05:10.58[hC]:)
05:11.03adeel[hC], yep
05:11.44keith4__adeel: alright, thanks. I'll try it
05:12.40adeelnp
05:13.49mikongHI , have somebody use the * for callback server ?
05:14.15adeelmikong, as a callback server? no...haven't used it as a server, but i have used the callback functionality and it works
05:15.00keith4__what's a callback server?
05:15.11adeelyou call the box, it hangs up and calls you back
05:15.14mikongHow to do this ?  the callback functional ?
05:15.23keith4__ah, interesting
05:15.37adeelor can connect 2 different people together by initiating the calls
05:15.40adeeland then bridging it
05:18.12*** part/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com)
05:18.22mikongBut the * how to  connect 2 different people ?
05:18.53adeelusing the app_callback iir
05:18.56adeeler iirc
05:20.21mikongIs it include the normal version ?
05:22.33adeelshould be
05:25.47mikongsure ?
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05:32.58*** join/#asterisk shital (n=shital@122.167.99.32)
05:37.45adeeli don't see why it wouldn't...unless you didn't enable it while compiling
05:37.45adeelfile, http://bugs.digium.com/view.php?id=12615
05:37.45adeelthanks
05:38.22unpaidbillapparently 1.2.27 doesnt choose the codecs correctly
05:38.54unpaidbillit isnt honoring the priority i set in sip.conf general, but it works on a 1.4.19 server
05:39.14drmessanounpaidbill
05:39.16mikongthanks
05:39.34drmessanoIf you need g729 for a specifier peer, ONLY ALLOW G729 for THAT PEER
05:39.41drmessanoSpecific
05:39.50drmessanodisallow=all and allow=g729
05:39.52drmessanoSimple
05:40.33[hC]anyone know of a sip client on osx that isnt as bad as x-lite?
05:40.45[hC]x-lite seems to not even send packets out anymore for some reason
05:40.59drmessanoMaybe your OSX is broken
05:41.03drmessanoHard to believe, I know
05:41.17[hC]Its not likely. X-lite has a history of doing shit like this in osx
05:41.24unpaidbilldrmessano no, that isnt it at all.
05:41.25[hC]xmeeting (sip client) works, but has no dtmf pad,.
05:42.10drmessanounpaidbill: What are you trying to do?
05:42.35unpaidbillim trying to set it to use g729 by default, but fail over to alaw if the endpoint doesnt support g729
05:42.38unpaidbilldisallow=all
05:42.44unpaidbillallow=g729,alaw
05:42.53unpaidbillin general, and in the specific config
05:42.58unpaidbilland it ALWAYS uses alaw
05:43.08unpaidbillbut on 1.4, it goes by the order
05:43.19unpaidbilland for this specific machine i cant use 1.4
05:43.27drmessanoTry putting it on two lines?
05:43.33jblack<PROTECTED>
05:43.39unpaidbillyeah on 2 lines it doesnt do it either
05:43.47unpaidbillyes i can use g729 fine if it's the only codec i specify
05:44.23unpaidbillsip show settings...   Codecs:                 g729,alaw,ulaw,gsm
05:44.41drmessanohmmm
05:44.53unpaidbillso i would assume that means priority for g729.. but no... 1.2 wont have it!
05:45.04unpaidbillunless there's something im missing that isnt needed in 1.4
05:45.55drmessanoDo you have the codec specified in some other place?
05:46.35unpaidbillit's in [general] and the user specific configuration
05:46.38unpaidbillin sip.conf
05:46.41unpaidbilland it's ordered the same way in both
05:46.49unpaidbillg729,alaw,ulaw,gsm
05:47.04unpaidbillim gonna buy another damn g729 license and test this on the 1.4 box
05:47.12unpaidbillstupid g729.
05:47.14unpaidbillhehe
05:47.24drmessanoI dont ever specify a codec in a peer definition unless im overriding something
05:47.36unpaidbillwell
05:47.41unpaidbilllet me try removing it and see what happens
05:47.53unpaidbilli was just doing it because nothing else was working and i started going nuts with the config
05:48.18[hC]ok so i found x-lite's problem
05:48.35[hC]its sending my local ip in the sip headers instead of my public ip since presumably it coudlnt figure out what my public ip was.
05:48.39[hC]and there's no way to force set it
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05:51.00unpaidbillyea that didnt fix it
05:51.11unpaidbilli am a sad man.
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06:40.18LinHi there. Im having problems with SIP behind NAT ADSL routers. I have configured my router to FORWARD everything coming from 5060-5070 and 10000-20000 udp to my SIP Phone, but Asterisk keeps complaining about  Maximum retries exceeded on transmission, testing with IAX2 works flawlessly. Anything else should be done? best regards.
06:49.08drmessano1. Stay in channel more than 8 minutes.. this isnt Dell 24/7 tech support
06:49.15drmessano2. ??????
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07:09.09adeel2. if you want priority service, it's $50 bucks an hour
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08:24.49tzafrir_homeOT: any idea how I can rename a page in voip-info.org ?
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08:38.53E-bolaCan somebody help me figure out how to get the following functionality in my dialplan. I want a msg to be played when ppl call int that says "All lines are busy please wait, or press 1 to leave a message" which obviously should let the caller press 1 to go to voicemail or let him wait for somebody to answer the phone
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08:42.35mikongBackground()
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08:45.05badcfehello. i would like to set a 420p wildcard in E1 mode by software (without touching the jumpers).  how do i do this?
08:45.36badcfethe only configs for such a card is zapata.conf and zaptel.conf right?
08:47.12E-bolamikong: I tried using background but it never moves on, it just seems to wait for something to be entered forever
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08:47.19E-bolado i need to use some sort of timeout command or similar?
08:48.12mikongshow your dialplan
08:48.59E-bolahttp://pastebin.com/m619aeba1
08:49.11E-bolaIts my first time trying to make an IVR sort of thing so it might be rubish
08:49.56jblackYou're on the right track
08:50.28jblackI'd suggest adding:
08:50.29E-bolaive seen the ResponseTimeout and digittimeout commands, but im nto sure where to palce them
08:50.40jblackexten => s,n,WaitExten(20)
08:51.14jblackAlso, instead of doing 2,1  2,2   2,3 so on, you can do 2,1  2,n  2,n 2,n ... Saves you from having to count.
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08:51.34E-bolaya but its easier to locate a line if you specify a number like if you need to make a goto
08:51.38E-bolaatleast thats what ive experienced
08:51.45E-bolaBut where would you have me put the WaitExten?
08:51.51E-bolabelow background or above or?
08:52.13krdianis back. what's up? o/~
08:52.14jblackbelow.
08:52.16krdianhi
08:52.29jblackPlay your sound first, then wait for them to enter a digit. ;)
08:52.45E-bolajblack: so i change background to playback?
08:53.17jblackyou can leave that as background.
08:53.39E-bolaalright, ile test it now....
08:54.11krdianis there any ip vide phone suports amr codec ?
08:54.41krdians/vide/video/g
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09:02.46Rico29hello ! does anybody know where I can find the second edition of the o'reilly book "asterisk the future of telephony" in pdf version ?
09:03.13jblack~book
09:03.14jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
09:03.40jblackI really suggest you buy the book though
09:04.34E-bolajblack: thanks it works perfectly now :)
09:04.43jblackwelcome. You're on the right track
09:05.09jblackI can tell you're reading the book. :)
09:05.24Rico29thanks
09:05.53jblackI was referring to e-bola. Rico, I can tell you're working on getting the book. =)
09:06.01jblackWhich is a great start to the day, indeed.
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09:09.45E-bolaisnt reading the book
09:09.45E-bolahehe
09:09.55E-bolai did read most of it though about 1 year ago
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09:12.18adeelcan someone explain to me what the hell this means, "You hereby grant Digium a perpetual, worldwide, royalty-free, irrevocable, non-exclusive, sublicenseable and transferable license under any patent You own or control, now or in the future, to make, have made, use, sell, offer for sale, or import Submissions or any modifications thereof, including without limitation any combinations of the Submissions or modifications thereof with soft
09:12.18adeelware, technology or services of Digium or its affiliates.:
09:12.57adeelis that more or less saying that anything i ever make/patent can be used by Digium irrespective if it has nothing to do with *?
09:13.01jblackwhere did you read that?
09:13.07adeelhttp://bugs.digium.com/license_agreement.php
09:13.30jblackThat's the license agreement for the bug tracker?
09:13.38adeelfor submitting patches
09:14.04jblackYes. That sounds right, then.
09:14.23jblackThat's why a lot of the good stuff goes into callweaver, which is really promising, but never seems to make a release.
09:14.35E-bolalol
09:14.40E-bolathats an insane license agreement
09:14.42adeeleh, releases are overrated...so long as the thing compiles i'm fine
09:14.53jblackSome people (quite sensibly!) dont' want to give away all rights to their work.
09:14.58E-bolaI dont think its even legal?
09:15.11adeelthey might as well toss in that you're first born is legally digiums'
09:15.11jblackI never noticed the "any patent" part.
09:15.14E-bolaif i understand it correctly you sign over all rights to anything you will ever make?
09:15.20adeelthat's what i got out of it
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09:15.27E-bolaIt would more or less make you a slave of digium for life?
09:15.27adeelremind me to never submit patches to digium
09:15.28E-bolarofl
09:15.30jblackYeah, it looks like a pretty wide carte blanche to me.
09:15.54E-bolaHavent license agreements such as that and eula's etc
09:16.02adeelthat's a nice way to kill an 'open source' project
09:16.10E-bolabeen voted invalid in courts numerous times? for the simple reason that nobody reads them?
09:16.18jblackI knew their submission agreement was onerous. I didn't realize it was that bad.
09:16.19adeelnot that i'm aware of
09:16.20E-bola(except adeel) :)
09:16.29adeelaren't you glad that i do?
09:16.35adeeland i read them SPECIFICALLY for clauses like that
09:16.36E-bolaIndeed I am :)
09:16.46E-bolaI still dont think its legal though
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09:16.54adeeldo you want to spend the time and money to find out?
09:17.02jblackLike I said, I knew they were nasty already, which is why I wouldn't submit the couple small patches I made to them.
09:17.07adeelbegins the process to migrating to a 'safer' code base
09:17.36adeeli think i'll just start my own repo of patches that won't be subject to their rediculous license
09:17.52adeelsomething to add to my todo list
09:17.53jblackThat's where callweaver came from, but they've diverged quite a bit.
09:18.13adeeli've got a couple of minor patches i'd submit...but not after reading that
09:18.44E-bolaactualy thought asterisk was gpl....
09:18.45jblackThe fax stuff is in callweaver, there are conference apps built in that don't require zaptel...
09:18.57jblackasterisk is free software.
09:19.04jblackIt's actually dual licensed, I believe.
09:19.06adeelE-bola, it's supposed to be....but digium owns the copyright and can change terms
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09:19.33adeeljblack, yes, but getting 3rd party supported programs for callweaver is a little troublesome
09:19.40E-bolaSo the sourcecode is GPL but patches you submit arent?
09:19.42jblackand yeah, it's gplv2
09:19.43E-bolamakes no sence...
09:19.43adeelany idea what bug tracker digium uses?
09:20.00adeelE-bola, well it states that digium can re-issue your patches to whatever license scheme they want
09:20.04jblacke-bola: That's something that confuses people. Licensing, and ownership, are two different things.
09:20.37adeelE-bola, the owner defines the license, and as such, can change it whenever they wish
09:20.47jblackIf they accepted gpl'ed patches without taking ownership, then they wouldn't have the option to make a seperate release of asterisk, with a nonfree license.
09:21.03adeeljblack, not entirely
09:21.08E-bolaadeel: well thats not possible
09:21.13E-bolayou cant change the licens of GPL code
09:21.23adeelE-bola, i can if it's my code
09:21.25E-bolaso if you submit GPL license patches
09:21.25jblackIt's complicated enough for him without covering edge cases.
09:21.30E-boladigium cant use them?
09:21.47adeelE-bola, read the agreement, you sign over ALL rights and license terms to digium
09:21.54adeeldigium will then define the terms of the patch
09:22.06jblacknot in a differently licensed product, not without your permission, e-bola.
09:22.10E-bolathats bs
09:22.14adeelbut from my understanding...if you RELEASE software under a certain license term, you cannot retroactively change that license
09:22.18E-bolahow can digium proove it was the owner of the code who submitted it?
09:22.40adeelE-bola, with this clause:
09:22.41adeelYou hereby grant Digium a perpetual, worldwide, royalty-free, irrevocable, non-exclusive, and transferable license to use, reproduce, prepare derivative works of, publicly display, publicly perform, distribute the Submissions, and to sublicense such rights to others. The rights granted may be exercised in any form or format, and Digium may distribute and sublicense to others on any licensing terms, including without limitation: (a) open
09:22.42adeel<PROTECTED>
09:22.44jblackYeah. Post-fact revocation doesn't exist in the US, unless it's in the contract. Which it's not in the gplv2
09:22.53jblackplease don't flood.
09:22.57adeelsorry
09:23.46jblackAnd technically, you're not giving them full rights.
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09:24.26adeelwell you're not transferring ownership
09:24.34jblackcorrect.
09:24.43E-boladoesnt matter
09:24.45adeelbut you're giving them free reign to do with your work whatever they want, without having to pay for you
09:24.55jblacke-bola: Oh, trust me, it does.
09:24.58E-bolalol
09:25.02E-bolaso if i submit the apache2 codebase
09:25.05E-boladigium then owns it?
09:25.12adeelno, it's not yours
09:25.28E-bolathats my point
09:25.35E-bolawhy would they think anything i submitted was mine ?
09:25.47E-bolaif its gpl'd its likely its publicly availeble
09:25.48adeelE-bola, read the damn license and then ask questions
09:25.52jblackE-bola: Shhhhhhhhhhhh
09:25.55adeelit's the 4th paragraph that covers it
09:26.06E-bolaIt doesnt matter what it reads
09:26.10adeelyes it does
09:26.11E-bolaits by logic not possible
09:26.13adeelit's a legally binding contract
09:26.17E-bolano its not
09:26.18E-bolathats my point
09:26.21E-bolawhere is my signature?
09:26.23E-bolathey got no proof at all
09:26.30adeelyour an absolute idiot
09:26.35jblackE-bola: You're right on the nose on you're thinking about third party software. That's gotten some projects that have dual licensed in the past before.
09:27.00jblackForcing them to drop proprietary projects.
09:27.10E-bolaadeel: you seem to think everything falls under american law, either that or you do not get it at all
09:27.24E-bolaI cant sign shit by clicking a button
09:27.27E-bolaif you think so ur an idiot
09:27.40jblackActually, you can.
09:27.48E-bolaNot where I live
09:27.54E-bolaits not valid as a court contract
09:27.57jblackWe're talking US laws, of course.
09:27.58E-bolathats been proven numberous times
09:28.21E-bolaif thats the case in america their law is rediouclous, but then again that wouldnt supprise me. They have DMCA as well
09:28.52jblackThe US unfortunately, based upon case law, decided that "shrink wrap" EULAS are legal contracts, which extends to active user consent.
09:29.19jblackThe same thing that makes clicking "I agree" when you install american software legal in the US is the same thing that makes it legal on a website.
09:29.46E-bolaI once saw a study of a test of the EULA system. It sid somewhere burried int he text that if you call the following number you would get 500$ in return. out of 50000 downloads only 2 people called
09:29.52jblackAnyways, going back to your more interesting question, what happens if I submit your GPL code to a project that demands I grant them non-exclusivity rights...
09:30.20jblackIn the US, a contract is enforceable if a party agrees. Reading is not a requirement.
09:30.42E-bolaHere it was decide that simply clicking "next" in an isntaller isnt agreeing
09:30.46E-bolasince nobody ever reads it
09:31.00jblackAnd?
09:31.05E-bolaWhich makes sence, any country which doesnt work int hat way is obviously screwing their own citizens over
09:31.48jblackUnfortunately, most first world countries work exactly that way, and the trend is towards that behaviour, not away. You may not have those rights forever (but I hope you do)
09:32.24jblackAnd? government isn't about your rights. It's about protecting the power of those that have it
09:32.37E-bolaThats true if your a pessimist
09:32.50jblackregardless, it is what it is
09:32.58E-bolaSome countries outside america actualy have working democracies :)
09:33.15E-bolawithout insane presidents and giant power lobbies
09:33.37jblackI'm not really interested in a chat about the flaws in the american (and british, and to a lesser extent, the entire european) legal system.
09:33.51E-bolalets shut it down then :)
09:34.19jblackGreat. Now, if you want to get on your donkey, and go tilt at windmills, then I wish you godspeed, because the world can use more of those sorts.
09:34.55jblackbut sitting here, fuming about the great arms swinging around, isn't gona do any good
09:35.15*** part/#asterisk bkw_ (n=brian@adsl-70-234-182-53.dsl.tul2ok.sbcglobal.net)
09:35.22E-bolaI just feel sory for the americans
09:35.46jblackThere's places not so bad off, and others much worse.
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09:59.01badcfehello. i would like to set a 420p wildcard in E1 mode by software (without touching the jumpers) .. i hope its possible?  how?
09:59.40JTi doubt it
10:00.09Rico29does somebody knows how to write something on an IPphone lcd screen (thomson st2030)
10:00.19Rico29looks stupid, i know
10:00.21Rico29:)
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10:31.27whymarkwhhi all
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10:49.07badcfein asterisk cli i do zap show status and the span i configured is OK but the IRQ listed is 0 .. does that mean its not up?
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10:54.30MicWhi
10:54.53MicWshoud it be possible to use video conferencing with asterisk and openwengo (as client)?
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11:10.45Rico29MicW>  i tried
11:10.58Rico29but wengophone never registers on my asterisk
11:11.04Rico29and i dont know why
11:11.11Rico29but a friend of me did it
11:17.48whymarkwhbadcfe: still here?
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11:33.17tzafrir_homebadcfe, it means it does not lose interrupts
11:33.38tzafrir_homeWhich is generally a Good Thing
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11:42.15Mahmoudany one knwos why can't I see "dialplan reload" ?
11:42.19Mahmoudusing the latest SVN
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11:45.41shastaeeek, [May  6 10:15:42] NOTICE[14574] chan_sip.c: Peer 'øK´¶È^C°¶' is now Reachable. (8ms / 100ms)
11:45.42*** join/#asterisk XnOSX (i=d491af1a@gateway/web/ajax/mibbit.com/x-5b15ee08daecb8b6)
11:45.50XnOSXgood morning!
11:47.06XnOSXanybody here have some problem with compile a zaptel driver with make b410p in debian 2.6.24-1-686? i have a problem for compile zapata
11:47.23XnOSXi need to install a b410p wildcard
11:47.46XnOSXhere is the log http://pastebin.ca/1012613
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11:50.33stelioskXnOSX : Its not zaptel that has a problem its misdn
11:59.07Rico29where can I find the asterisk headers ?
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12:02.44Rico29help please
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12:08.58ManxPowerRico29: /usr/include/asterisk I believe.  "find / -name asterisk.h -print" should tell you
12:09.37Rico29i did it
12:09.40Rico29i have the lib
12:09.52Rico29but when i want to compile the addons, with many options
12:09.55ManxPowerI don't think Asterisk contains any libraries.
12:10.01Rico29it says : checking for asterisk.h... no
12:10.10ManxPowerRico29: you can't mix 1.2, 1.4, and 1.6 versions
12:10.10Rico29(i use --includedir
12:10.16Rico29i dont
12:10.50whymarkwhSkinny/SCCP Protocol is this just for cisco phones
12:10.51whymarkwh?
12:10.53ManxPowerWhat versions of Asterisk and Asterisk-addons do you have?
12:11.04ManxPowerwhymarkwh: yes.  The protocol is owned and protected by Cisco.
12:11.42whymarkwhis there any other text to speech enjins that work with asterisk?
12:11.57ManxPowerOther?  Which one do you use now?
12:12.04whymarkwhnone
12:12.28ManxPowerThere are at least two TTS engines that work with Asterisk.  Festival and Cepstral
12:12.39Rico29ManxPower>  asterisk-1.4.19.1         asterisk-addons-1.4.6
12:12.54ManxPowerTry reinstalling Asterisk
12:13.00Rico29sound qualioty with vestival is bad
12:13.02Rico29festival
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12:13.29whymarkwhManxPower: witch is the better of the 2?
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12:14.04ManxPowerwhymarkwh: I like Cepstral.  It sounds much better than any other similar priced system.
12:14.41whymarkwhis it an asteriks addon?
12:15.32ManxPowerNo, it is a commercial product
12:15.47whymarkwhk thx
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12:17.12lirakis_workRico29: there are TONS of configuration options, "voices" etc. with festival .. but its configured in Lisp ( oh dear god )
12:17.20lirakis_workit is very very configurable though
12:17.28lirakis_workits ashame its not "nicer" by default
12:19.31Rico29ManxPower>  an idea for my pb ?
12:22.04ManxPowerRico29: Make SURE you have the
12:22.23ManxPowerMOST RECENT Asterisk-Addons.  There were significant fixes in the build process in the past few versions
12:22.49shido6:)
12:22.58shido6hey ManxPower
12:22.59ManxPowerwaves to Shido
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12:29.11Rico29ManxPower>  i downloaded them this morning
12:29.25Rico29but i installed asterisk in non-root mode
12:30.13ManxPowerRico29: My servers are not exposed to the Internet in any way, so I've not felt the additional hassle of non-root is worth it on my systems.
12:32.43Rico29:(
12:32.46Rico29that sucks
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12:33.05Rico29it would help if I pastebin my ./configure result ?
12:33.23*** join/#asterisk xenonex (n=xenonex@89.218.237.113)
12:34.29ManxPowerIT might help someone, but not me
12:34.37ManxPowerI only to the easy stuff for free 8-)
12:35.08Rico29:)
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12:36.04Rico29i realy dont understand why it doesn't work....
12:36.45jdugganhey guys, are there any brits here using digium FXO (our specific model is TDM404B )? im having problems with the card
12:38.19[TK]D-Fender~ask
12:38.20jbotextra, extra, read all about it, ask is Questions in the channel should be specific, informative, complete, concise, and on-topic.  Don't ask if you can ask a question first.  Don't ask if a person is there; just ask what you intended to ask them.  Better questions more frequently yield better answers.  We are all here voluntarily or against our will.
12:42.16*** join/#asterisk Marquel (n=Marquel@port-418.pppoe.wtnet.de)
12:42.22Marquelmorning
12:42.25*** join/#asterisk nighty^ (n=nighty@KD124213240154.ppp-bb.dion.ne.jp)
12:43.05x86morning
12:43.14x86err.... moin ;)
12:43.20x86wie gehts?
12:43.43*** join/#asterisk RoyK (n=roy@fw.fortel.no)
12:45.32Marquelmoin x86 - gut und selbst?
12:46.46Marqueli have a little problem w/ an internal ZAP-channel. first and most important: call pickup by dialing default "*8" doesn't work. the phone just reports "not available". maybe something w/ my zap-channel configuration?
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12:49.35ManxPowerMarquel: What phone reports "unavailable"?
12:49.40ManxPowerAnalog phones don't do that
12:50.14jdugganwots the best to way to tell if my digium card is recognised and all ports working etc?
12:50.22ManxPowerjduggan: use it
12:50.41jdugganManxPower: well, i want to ensure the hardware is alright before i mess with the config, because its a config issue no doubt
12:50.55ManxPowerwaits to be told what actual card you have
12:51.05jdugganDigium TDM404B
12:51.11jduggan4 port FXO
12:51.16ManxPowerdmesg should tell you when the driver loads
12:51.59badcfewhymarkwh: yes im still here.  and you?
12:52.32jdugganManxPower: http://internetworkpro.org/pastebin/2445
12:52.53MarquelManxPower: an ISDN-phone, connected to a bri-card.
12:52.57badcfetzafrir_home: thanks.  but then i dont understand why i get "unable to create channel.  reason 0" (circa) when i Dial(Zap/1)
12:53.50jdugganManxPower: also http://internetworkpro.org/pastebin/2446
12:53.58[TK]D-Fenderjduggan: want to know if its ok?  USE IT
12:54.17ManxPowerMarquel: I don't know if whatever card you have supports *8, but if it does it would be configured for the driver
12:54.51ManxPowerjduggan: read the error message.  Do what it suggests.
12:55.00MarquelManxPower: thx, that's something a can look up.
12:55.00tzafrir_homeMarquel, something is wrong with the context?
12:55.13ManxPowerjduggan: what color are the modules on the card?
12:55.19tzafrir_home*8 is implemented in the core of Asterisk , right?
12:55.30tzafrir_homeIt does work with analog Zap
12:55.37jdugganManxPower: red
12:55.39ManxPowertzafrir_home: I thought so, but for example, chan_iax2 does NOT support *8 pickup
12:55.40Marquelif i get lost in a few moments, don't bother. i'll be back later.
12:56.17ManxPowerjduggan: I don't see the card driver being loaded
12:56.26tzafrir_homeWell, it seems to actually do something (but not the right thng)
12:56.37ManxPowerzaptel is another required driver -- not hardware specific.
12:56.40jdugganManxPower: hmm, i thought the zaptel driver was the card driver
12:56.41jdugganaha
12:56.44jdugganok, one moment
12:56.57ManxPowerjduggan: no, zaptel is the zaptel driver, your card driver would be listed in the Zaptel README.
12:57.04jdugganwctdm24xxp
12:57.39ManxPowerjduggan: I strongly doubt that
12:57.53jdugganits been autoloaded, i shall read the readme
12:58.26ManxPowerNext time read the README before coming here.
12:59.09jdugganah, it was a pdf i read that suggested that was the correct module for the card
13:00.07ManxPowerjduggan: Many idiots give advice.
13:00.26ManxPowerI trust the docs located in the asterisk, zaptel source as the primary docs
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13:07.55JayTee52mornin *'ers
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13:16.03M1s3ryjduggan, I may have missed earlier information, however is this the TDM400? or the TDM410?
13:17.36ManxPowerjduggan: Digium TDM404B
13:18.28*** join/#asterisk Marquel (n=Marquel@port-418.pppoe.wtnet.de)
13:18.31Marquelre
13:20.11Marqueltzafrir_home: what could that be? the context simply includes the context for sip-phones (working very well).
13:21.00ManxPowerM1s3ry: IIRC, the 410B uses the tdm24xx driver, right?  I know the 400B uses wctdm
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13:21.13M1s3rycoorect
13:21.17M1s3rycorrect*
13:21.30M1s3rythat's why i asked... I couldn't tell which one he had...
13:21.35jdugganwctdm24xxp <- thats what's loaded
13:21.35ididitwithsuseanyone usingxchat irc program?
13:21.52tzafrir_homeMarquel, if you set verbosity to at least 3, what do you see in the CLI?
13:21.54jdugganis that right?
13:21.55M1s3rywishes the naming scheme allowed for easier determination on which card a customer was using
13:22.09tzafrir_homeididitwithsuse, I am
13:22.11ManxPowerjduggan: any digium driver will load even if the card does not exist in the system
13:22.17M1s3ryjduggan, what card is it?
13:22.31jdugganFound a Wildcard TDM: Wildcard TDM410P (4 modules)
13:22.32Marqueltzafrir_home: incoming call on zap (that's okay ), and an overlap dialed call coming from
13:22.34jdugganin dmesg
13:22.35Marquel*narf*
13:23.06Marqueltzafrir_home: incoming call on zap (that's the one to be picked up), and an overlap dialed call coming from the zap user in question (his MSN) to "<unspecified>".
13:23.07ManxPowerjduggan: next time be more accurate.  The TDM404B is a different card than the TDM404B you reported just a few mins ago
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13:23.46ManxPowerSorry, the 414/410
13:23.49jdugganManxPower: i took the card details from the shop i bought it from.. i guess its actually a different model.. which i had no way to know until i loaded its module
13:24.01jdugganand dmesg told me
13:24.23mwallinghans, why did you kill your wife?
13:24.29mwallingdmesg told me to!
13:24.32destructureindeed
13:24.34mwallingi had to !
13:28.28[TK]D-FenderReiser performs well with fragmentation....
13:28.49[TK]D-Fender</pun>
13:28.56rob0ouch
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13:29.54ManxPowerI used to have a client that was in criminal law defense.  The only confidence I have in the prosecutor's office is that they will lie, cheat, and steal to win their case, regardless of if the defendant is innocent or not
13:30.50ManxPowerIn one case the person was on death row and the state had evidence that PROVED the person could not have committed the crime he was being executed for.
13:30.53rob0agreed, I think lawyers tend to be the lowest form of human life, and prosecutors are the lowest form of lawyer
13:31.21rob0too bad they can't be held accountable
13:31.40ManxPowerRegardlesss of if you are pro or anti death penalty, I most people will agree that is terrible.
13:32.05ididitwithsusewhat is considered the most resent stable version of asterisk?
13:32.15rob0/topic
13:33.53ididitwithsuse'
13:34.33ididitwithsuse.
13:34.59*** join/#asterisk stkn|work (n=stkn|wor@gentoo/developer/pdpc.active.stkn)
13:35.37[TK]D-Fender"There were not big enough changes for Asterisk 1.6 to require a major ABI change release of libpri, so instead most of the 1.6 specific functions were back ported to the 1.4 branch of libpri (including BRI support, as well as a few other things such as TBCT for Q.SIG)"
13:36.54*** join/#asterisk mort_gib (n=mjensen@16.Red-83-36-63.staticIP.rima-tde.net)
13:37.21tzafrir_homeWell, the ABI was technically officially changed:
13:37.44tzafrir_homein 1.4.3 the library had the SONAME 1.0 . In 1.4.4 it is 1.4
13:39.17[TK]D-Fendertzafrir_home: but not "major"
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13:41.26JayTee52rob0, did you ever hear this joke? Daughter: "Mom, can I get pregnant from having anal sex?" Mother: "Of course, dear! Where do you think all the lawyers come from?"
13:41.39*** join/#asterisk henrique (n=henrique@unaffiliated/henrique)
13:42.02rob0lol, I have some lawyer friends to tell that one to!
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13:43.15JayTee52rob0, "Why don't sharks eat lawyers?"
13:43.27rob0Professional courtesy.
13:43.32JayTee52hehe
13:44.22tzafrir_home[TK]D-Fender, changing the SONAME is generally considered a major change
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13:50.50ididitwithsusell
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14:04.07ididitwithsusein witch order is it best to install :zaptel ,, libpri,,addons,,asterisk or doesn,t it matter?
14:07.05russellbzaptel and libpri first (doesn't matter which order)
14:07.07russellbthen asterisk
14:07.07russellbthen addons
14:08.43mort_gibAnyone knows what chan_sip.c: Maximum retries exceeded on transmission means??
14:09.21russellbmeans that the maximum number of retries on transmitting a packet has been exceeded ...
14:09.34*** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com)
14:09.41russellband the packet was never acknowledged by the other end
14:09.50mort_gibThat's what I though, so network issue??
14:10.35zktechbug 0009896   http://bugs.digium.com/view.php?id=9896  has been closed. I check the changelog for the 1.4.20.rc-2 and did not see it in there. I reall nead this in the branch releases.
14:11.01zktechHow can I tell what the state is on the bug? Thanks
14:11.40fileit was added to trunk and will be in 1.6.1
14:12.20zktechWhat about 1.4 ? I am running
14:12.26fileit will not be in there.
14:12.39jasonwoothey, what's the ISBN number of the Book?
14:12.40Juggieits not a bug its a feature
14:12.43shastaanyone know if there was a bug, fixed between 1.4.17 and 1.4.20 that could cause such things? NOTICE[14574] chan_sip.c: Peer 'øK´¶È^C°¶' is now Reachable. (8ms / 100ms)
14:12.44Juggie~book
14:12.45jbotAsterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook
14:12.56`Sauron~buybook
14:12.56jbotYou can buy "Asterisk the Future of Telephony" at http://www.oreilly.com/catalog/9780596510480/ so go buy it SERIOUSLY
14:12.58jasonwootdanke
14:13.36zktechHow can I get a patch version into 1.4  right now I have one for 1.4.17 and I am stuck there until I can get something for the current branches.
14:14.05ManxPowerzktech: if it did not make it into 1.4.x it never will.
14:14.27ManxPoweryou can manually apply the patch to your 1.4 if you want, or you can hope to find a 1.4 version on the bug file attachments
14:14.35*** join/#asterisk Tourinho (n=tourinho@201.37.118.16)
14:15.21zktechThe version 12 of the patch worked on 1.4.17 but I could not get any of version of the patch to run on 1.4.19 or 1.4.20 rc
14:15.38Tourinhohello people.. if I have an application that receives a call, play prompts end dial to another place using SIP and bridge calls, should I have 2 g729 codecs license?
14:17.43ididitwithsusewhere can i find a list  of dependecies for asterisk?
14:17.54[TK]D-Fenderididitwithsuse: www.asterisk.org
14:18.02[TK]D-FenderTourinho: 1
14:19.14zktechWithout the patch 0009896 I can not reboot any ATA that is secure. This is major for us.
14:19.14zktechWhen would it be likely that a version 1.6.1 would be available and stable for a production server?
14:19.33filewe can not tell the future.
14:20.11anonymouz666zktech: maybe when reach the version .20
14:20.24anonymouz666:P
14:21.06zktechanonymouz666  I is not in the changelog for .20
14:21.27Tourinho[TK]D-Fender: thank you
14:21.45anonymouz666zktech: I mean 1.6.20
14:21.56russellbit's not going to ever be in 1.4
14:22.02russellbit's a new feature, and was only merged into 1.6
14:27.03zktechI will move to 1.6 then. Are the betas stable enough for production and am I likely to get hit with major dial plan incompatibilities? Where is the best source of info on making the jump from 1.4 to 1.6?
14:27.35zktechI am also using the TC400 cards is 1.6 likely to give me any issue there?
14:29.08filewhat you want is not yet in a 1.6 release
14:29.33Nuggetbuilt 1.6 yesterday
14:29.54JayTee52has 1.6.0beta8 running on a test box
14:33.34russellbyay for people actually testing 1.6
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14:36.27ididitwithsuseif you have a system that already has asterisk installed suse you can select it from the instalation package how do update it to latest version, can one do this is the question?
14:39.45russellbthat didn't make any sense
14:39.54russellband is probably a #suse question or something similar
14:44.01ManxPowerzktech: upgrade.txt in the asterisk source
14:46.03zktechManxPower I am downloading it right now Thanks
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14:48.57ManxPowerzktech: There SHOULD be info for upgrading from 1.0.x to 1.2.x as well as from 1.2.x to 1.4.x.  If there is not then download an asterisk 1.2 and look at that upgrade.txt too.
14:49.18ManxPowerThings that were deprecated in 1.2 were removed in 1.4 and that can bite you if you don't have both upgrade.txt files.
14:50.05ManxPowerI've lobbied to have both files included in 1.4 and 1.6, and both are in 1.6, IIRc, but I don't know about 1.4.
14:50.11ManxPowerrussellb: can you comment on that?
14:50.34russellbi have no comments on anything
14:50.49russellbi think they're all in 1.6
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14:50.53russellbdon't think they're all in 1.4
14:51.56ManxPowerhands russellb some /* and */s
14:52.37Tourinho1/1 encoders/decoders of 10 licensed channels are currently in use << this means that Im using 2 licenses?
14:52.58fileyou are using 1 license.
14:53.26Tourinhofile: thanks
14:53.35Marqueltzafrir_home: no more ideas?
14:58.52ManxPowerTourinho: A license is for one encode, one decode simul
15:01.10zktechIs there a posted timeline for the 1.6 releases?
15:01.21*** join/#asterisk ScarEye (n=scareye@12.27.87.66)
15:04.38[TK]D-Fenderzktech: Never
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15:06.13drfreezeHi
15:06.57drfreezeI have a T1 with an Adtran board that provides 8 phone lines
15:07.07drfreezeI route these lines to 2 TDM cards
15:08.12drfreezeIs it ok to also connect an analog phone to these lines? Currently, I have one line that I don't connnect to the TDM and use for other purposes. I am wondering if I can add it to Asterisk as well
15:09.02XnOSXanybody have a problem with the zaptel (make b410p) in debian unstable kernel 2.6.24?
15:09.30[TK]D-Fenderdrfreeze: Thats just retarded.  Get a T1 card an connect it direct to *
15:10.18drfreeze[TK]D-Fender: at the time this was configured, the provider didn't support T1 card
15:10.40[TK]D-Fenderdrfreeze: and when the GOT the T1, why did they get a channel bank for it?
15:10.41drfreeze[TK]D-Fender: if it works, why call it retarded?
15:11.13[TK]D-Fenderdrfreeze: Spend money converting it BACK to analog only to take it into *.  Thats like questioning why 10th generation photocopies suck.
15:11.26drfreeze[TK]D-Fender: I was all for the T1 card, but was told not to
15:11.45[TK]D-Fenderdrfreeze: See above.  Wrong move.
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15:12.09drfreezeIt's been a while, does the T1 support Fax?
15:12.15jdugganguys, when calling via the analogue fxo card, the sip client is hearing himself in the handset, like its looping back his own voice (doesnt seem like echo), what causes this?, if its sip-sip then its fine
15:12.57[TK]D-Fenderdrfreeze: T1 is just a carrier tech.
15:13.21drfreeze[TK]D-Fender: I need at POTS line for Fax
15:14.12drfreezeso, need to get a good analog signal somehow
15:14.41tzangerchannel bank :-)
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15:14.49[TK]D-Fenderdrfreeze: completely separate analog circuit direct from telco
15:15.03drfreeze[TK]D-Fender: you mean, like I am now doing. :)
15:15.15drfreezewell, that would be....retarded
15:16.43drfreezeAnyway, never got the question answered. I suppose the TDM card is smart enuf to recognize if an analog phone is in use on the same line
15:17.16badcfemy wildcard pri cpe span shows up with zap show status in * cli but it doesnt seem up.  how do i verify on a lower level?
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15:18.15drfreezebadcfe: what does zap show channels say
15:18.24badcfelszaptel also shows it as up (tho the channels are marked as Clear)
15:18.51badcfedrfreeze:  pseudo            pstn                       default
15:19.20drfreezenothing, eh
15:19.39badcfeoh?
15:19.51badcfepseudo is nothing .. ?
15:20.08badcfeits supposed to say something else huh?
15:20.24drfreezebadcfe: I have wildcard tdm, but it shows channels
15:20.33drfreezewhat is a wildcard pri?
15:20.52jasonwootchanges topic to 'what should I get my mom for mother's day?'
15:21.10badcfedrfreeze: te420b
15:21.39drfreezebadcfe: I have ound a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
15:21.50drfreezewhat does zttool from the commandline say
15:22.36badcfeit opens a curses menu listing amond the other spans:  OK              T4XXP (PCI) Card 0 Span 1
15:22.38[TK]D-Fenderbadcfe: .... and zapata.conf?
15:22.42[TK]D-Fender~pb
15:22.43jbot[~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
15:22.44[TK]D-Fender^^^^^^^
15:23.33*** join/#asterisk JayTee52 (n=jforde@207-67-84-181.static.twtelecom.net)
15:23.52badcfehttp://yourbackuponline.net/pastebin/20080509172345.txt
15:24.00ididitwithsusewaht is the linux distro of choice for the instalation of asterisk if this is going to be my   first install need to know whats best   ?
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15:25.12badcfehttp://yourbackuponline.net/pastebin/20080509172504.txt
15:25.13Nuggetlinux is linux is linux.  Use whatever you're most familiar with.
15:25.17tzafrir_homeMarquel, again, what do you see in the CLI?
15:25.23drummond_i rolled it out to CentOS
15:25.33JayTee52I second what Nugget said
15:25.35Marqueltzafrir_home: incoming call on zap (that's the one to be picked up), and an overlap dialed call coming from the zap user in question (his MSN) to "<unspecified>".
15:25.39tzafrir_homeididitwithsuse, if there's a linux distro you're familiar with, use it
15:25.44zktechI got disconnected. Is ther a posted timeline for the 1.6 releases? Thanks
15:26.21badcfe[TK]D-Fender, drfreeze: and the net people tel me that this seems good but they say theres a protocoll alarm and ask me to verify that this is VN4
15:27.07JayTee52I've run the compiled version of 1.2 and 1.4 on RHEL 5, CentOS 5, Debian 4 and Ubuntu 6.06 and 7.10 server with no problems and the RPM builds from Livna on Fedora 6 and 7 with no problems.
15:27.48drummond_i rolled it out to centos 5, but built it from scratch
15:28.06JayTee52that's what my main production server is running now
15:28.31JayTee52and my other * system is running RHEL 5 64bit
15:29.27badcfe[TK]D-Fender, drfreeze:  does it seem correct?  bu the way .. is there some low level debug possibilities at my side?
15:29.37[TK]D-Fenderbadcfe: Not sure.
15:31.15tzafrir_homeMarquel, could you please enable debug logging in logger.conf and in the CLI and post the full log as well?
15:32.33Marqueltzafrir_home: that'll have to wait until tomorrow until i can reproduce the necessary circumstances. but i'll return with the logs.
15:36.30drummond_jay, how isthe performance under 64bit?
15:37.39Juggiedrummond_, we run all 64bit in prod, its fine.
15:37.54JayTee52drummond, with a Quad Core Xeon and 4GB of RAM and 2 mirrored SAS drives it absolutely flies.
15:38.05drummond_cool.
15:38.39JuggieYa, we are running Dual Dual Core Xeons w/ 4gigs and Raid5
15:38.43Juggieand its fast :)
15:38.49JayTee52this is on a Dell 2950 and RHEL 5 64bit is what you get if you ask for RHEL preinstalled at the factory. If you want to run 32 bit you load it yourself.
15:39.07drummond_i have it running on a POS dell dual dimension dual core machine, along with splunk, nagios, nfs, mysql, and a bunch of other stuff that hogs memory.  i runs with out problems
15:39.08*** join/#asterisk dFence (n=chatzill@ings-d93223ec.pool.mediaWays.net)
15:39.25*** join/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
15:39.45Zeeekyo ho ho
15:40.19badcfewhat is VN4?
15:45.26ZeeekThe 3rd party ecosystem around asterisk is the subject of todays VoIP Users Conference live in 15 minutes. Join #voip-users-conference and talk/listen by reading http://x2z.eu
15:46.09ftp3anyone know someplace besides didx that I can get a good deal on wholesale usa dids?
15:50.04ftp3guess not :-D
15:50.26ftp3how about a linux utility to join to mp3s for my asterisk moh
15:50.35ftp3to=two
15:50.49[TK]D-Fenderftp3: Audacity, SOX, etc
15:51.30*** join/#asterisk timgws (n=LivedTyp@202.172.97.51)
15:52.10ftp3thank you :-)
15:57.06Uatechi there
15:58.05Uateci have a SIP session that's coming in to asterisk from our sip provider
15:58.35Uateci now want to pass it over to my openser proxy
15:58.38Marquelcu all
15:59.24Uatecbut when asterisk tries to send the invite to openser it sends the URI sip:012345678990@127.0.0.1
15:59.51Uatecwhich is the number presented on the incoming call
16:00.03Uateci need to override that as a URI and pass it my own details?
16:00.06Uatechow can i do this?
16:00.14MahmoudWhy can't I find "dialplan reload" command in the CLI?
16:02.51Mahmoudwhat is responsible to show "dialplan reload" command?
16:03.20Uatectype 'help' and see what's listed...
16:05.28*** join/#asterisk RoyK (n=roy@cnbokcafe.uio.no)
16:05.34[TK]D-FenderMahmoud: CLI is different between * versions.
16:06.30ZeeekJoin #voip-users-conference and talk/listen by reading http://x2z.eu - see you there
16:06.35*** part/#asterisk Zeeek (n=IceChat7@pdpc/supporter/active/Zeeek)
16:07.57Mahmoud[TK]D-Fender, I'm using the latest SVN
16:08.19[TK]D-FenderMahmoud: Of which branch?
16:08.51Mahmoud[TK]D-Fender, http://svn.digium.com/svn/asterisk/trunk/
16:09.00[TK]D-FenderMahmoud: Direct answer please.
16:09.42[TK]D-FenderMahmoud: "looks" like 1.6.  Indeed, type "help" and see what CLI options you've got.
16:10.57Mahmoud[TK]D-Fender, I have only "dialplan {set | show}"
16:11.27[TK]D-FenderMahmoud: look harder.
16:11.58Mahmoudwell, I have only two arguments for "dialplan"
16:13.00Mahmoudany modules to be loaded? currently autoload=yes in modules.conf
16:14.31[TK]D-FenderMahmoud: try looking under "reload"
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16:17.00Mahmoud[TK]D-Fender, i see list of modules, the closest thing I see is extconfig
16:17.19[TK]D-FenderMahmoud: "reload dialplan"
16:18.03Mahmoud[TK]D-Fender, reload d<tab> shows only dnsmgr and dsp (no dialplan)
16:18.52Mahmoud[TK]D-Fender, restarting asterisk, and then executing "dialplan show" doesn't show my dialplan
16:19.48Mahmoudthis is all what I see by "dialplan show" http://pastebin.com/d5878b26b
16:20.03Mahmoudwhich is indeed not my dialplan
16:20.26[TK]D-FenderMahmoud: You've wasted our time in describing it like you din't have the OPTION at all.
16:21.01*** join/#asterisk NRich (n=NRich@72.37.252.50)
16:21.12Mahmoud[TK]D-Fender, well, I have "dialplan" but I don't have the "reload" argument. This is what I said and my appologies if I wasn't clear
16:21.24[TK]D-FenderMahmoud: Go pastebin "cat /etc/asterisk/extensions.conf" and show us the perms on it as well.
16:21.39Mahmoudperms?
16:21.43NRichI have a machine with asterisk, it's using 97% cpu for unknown reasons.. can anyone help with this?
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16:21.52[TK]D-FenderPERMISSIONS
16:22.10JayTee52NRich if you run top what process is hogging the CPU?
16:22.30NRichasterisk
16:22.41NRich4310 root      25   0  4932 1276 4524 R 94.1  0.6   1148:52 asterisk
16:22.41Mahmoud[TK]D-Fender, for testing, it's exactly copy and paste from configs dir
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16:22.58Mahmoud[TK]D-Fender, permission: -rwxr-xr-x  1 root  wheel
16:23.01[TK]D-FenderMahmoud: I don't want your description, I want proof of tis condition.
16:23.16JayTee52NRich, is this a production system or a test system?
16:23.29NRichproduction
16:24.22Mahmoud[TK]D-Fender, http://pastebin.com/m348fc370
16:24.33JayTee52NRich, what version of * and what linux distro and version?
16:25.25NRichthis is a pbxtra box
16:25.44[TK]D-FenderMahmoud: And you're running * as?
16:25.51Mahmoud[TK]D-Fender, root
16:26.11[TK]D-FenderMahmoud: "module load pbx_config.so"
16:26.51JayTee52NRich, I'd call Fonality
16:27.11Mahmoud[TK]D-Fender, done, so far no change, no output either
16:27.11jjshoe_NRich Hey, I just sat down, developer at Fonality, what's up?
16:28.24[TK]D-FenderMahmoud: pb "ls -l /etc/asterisk"
16:29.35Mahmoud[TK]D-Fender, http://pastebin.com/m797b7c0d
16:29.48jjshoe_NRich feel free to pm me your four digit server-id and the issue you're having.
16:30.31NRichjjshoe_: message david kullman - dect on hud (im using his for barging)
16:30.36[TK]D-FenderMahmoud: ....  -rwxr-xr-x  1 root  wheel  25331 May  9 19:56 extentions.conf
16:30.44[TK]D-FenderMahmoud: I don't f'n think so...
16:30.51[TK]D-Fender:p
16:31.01NRichJayTee52: /whois NRich && internet whois my ip =]
16:31.19JayTee52wtf?
16:31.21Mahmoud[TK]D-Fender, I don't get you?
16:31.24[TK]D-FenderMahmoud: "extentions.conf" != "extensions.conf"
16:31.29NRichJayTee52: I am fonality
16:31.39Mahmoud[TK]D-Fender, LOL...
16:31.59[TK]D-FenderMahmoud: BRILLIANT
16:32.02JayTee52well, then I guess you don't need to call yourself then, do you? :-)
16:32.14[TK]D-FenderJayTee52: the voices need company sometimes...
16:32.18Mahmoud[TK]D-Fender, lol, man i spent hours troubleshooting this hehehe
16:32.26NRichI'm brand new here, they just gave me a task to debug and I'm trying my resources =]
16:32.49NRichthe problem has already mostly been debugged by david kullman already
16:33.39b11d`Anyone here ever work with an NEC ElectraElite IPK system?
16:33.48QwellNRich: #asterisk isn't one of your support resources.  That's something you're going to need to learn pretty much immediately.
16:34.04Mahmoud[TK]D-Fender, working great.. thanks man!
16:34.28jjshoe_Qwell I straightened it out with his boss :)
16:34.35Qwelljjshoe_: yeah..
16:35.08jjshoe_Qwell sorry buddy <3
16:35.23Qwellmeh, just..  yeah
16:35.26tzafrir_homewell, it is a great support resource. If you don't rely on it as such...
16:35.36Qwelltzafrir_home: not when your job is supporting trixbox :)
16:35.44Qwellie; paid support.  That isn't going to fly
16:36.01*** join/#asterisk mercutioviz (n=chatzill@66-17-33-47.biz.visl.arrival.net)
16:36.19alrsRats, I was going to have NRich ask Kullmann about Sky Bed
16:36.36jjshoe_alrs sky bed has been seperated.
16:36.45jjshoe_how's it going lars?
16:37.10alrsHanging out at my work-a-ma-job, Joel.
16:37.29alrsJust playing with zaptel-over-ethernet-over-MPLS
16:37.57jjshoe_I'm playing with my rocket ship cup I got from dennys at lunch yesterday. Something tells me I'm having more fun.
16:38.13alrsYou don't have a tattletale coffee cup?
16:38.17jjshoe_"Tell him I said he's welcome to come over and rejuvenate Skybed"
16:38.42jjshoe_I don't need a coffee mug from that bar when my girlfriend got that chick's number for herself that carlo's in love with.
16:39.45JayTee52checks his IRC client to make sure he's in #asterisk and not #trixboxofficegossip
16:39.48jjshoe_he died in a little in side that night :P
16:40.23jjshoe_JayTee52 sorry, I hope I didn't interrupt any string of help you were giving?
16:40.51jjshoe_it's basically dead quiet in here this morning.
16:40.55JayTee52nope, just a joke
16:41.12jjshoe_JayTee52 alrs used to work for us. just saying hey is all.
16:41.39JayTee52like I said, I was joking
16:41.46jjshoe_oh I know :)
16:42.11[TK]D-FenderOk, question I asked a week or so ago : Looking for embedded systems similar to Soekris & Alix.  Anyone got some links for me?
16:42.35Qwell[TK]D-Fender: gumstix
16:42.37alrsNow I face challenges such as http://www.pastebin.ca/1012858
16:42.41Qwelltoo small? :)
16:42.59[TK]D-Fenderrequirements : 1 PCI minimum, 2 PCI is < 3 NICS.  GBIT highly preferred
16:43.37jjshoe_Qwell :P
16:43.47Qwelljjshoe_: ?
16:43.55[TK]D-FenderQwell: Yes... too small...
16:44.05jjshoe_Qwell gumstix.
16:44.14Qwell[TK]D-Fender: ever seen one of KrisK's gumstix Asterisk boxes?
16:44.18jjshoe_alrs most retarded pastebin site I've ever seen. looks like ass in ie7.
16:44.22[TK]D-FenderQwell: Nope.
16:44.22Qwellit's a pretty amazing sight
16:44.51jjshoe_alrs enjoy you digium card.
16:45.00alrsall three of them
16:45.07jjshoe_Qwell / [TK]D-Fender is astlinux any good?
16:45.24Qwellnever actually used it, but I've heard lots of good things
16:45.33QwellI know Kristian is quite competent
16:45.53jjshoe_I was reading up on it, and I was very impressed
16:47.33b11d`TK.. just wanted to say, im quite happy and impressed with the SPA-8000
16:47.58[TK]D-Fenderb11d`: Glad to hear.
16:55.24*** join/#asterisk atis_work (n=atis_wor@193.238.212.171)
16:58.01b11d`I only wish I could set an NTP server on it.. thats the only thing I found lacking.. and thats not a big deal at all
17:00.20JayTee52[TK]D-Fender, you might look at this: http://www.fit-pc.com/new/
17:00.49jjshoe_JayTee52 based on a mini/nano itx I presume?
17:00.59JayTee52paperback size
17:01.05JayTee52run Ubuntu or XP
17:01.14JayTee52*runs
17:01.34[TK]D-FenderJayTee52: No PCI = no thanks
17:01.46[TK]D-FenderJayTee52: I want to build a router
17:01.50JayTee52ah
17:02.03JayTee52and you have to have embedded linux?
17:02.14[TK]D-FenderJayTee52: Many distros out there for this.
17:02.28[TK]D-FenderJayTee52: Soekris is an options, I'm looking for others.
17:03.31JayTee52I haven't seen much in the way of SOC with a PCI connection. A router for VOIP?
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17:27.32jasonwootAnyone remember that guy the other day who was using Asterisk to control access to doorways?
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17:34.28DaveCanoecan one forge a custom insert when using cdr_mysql or cdr_pgsql?
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17:34.55DragoraNhi
17:34.58[TK]D-FenderJayTee52: Slap my Sangoma S518 for ADSL, 3-4 GBIT NICs (on-bourd preferred.  Wifi would be nice, or MiniPCI to add, etc.
17:35.11DragoraNwhich SIP client i should use for my winmobile 6? can someone advice?
17:35.19[TK]D-FenderJayTee52: Frankly a NET5501 is jsut about perfect, but I'm looking for alternatives.
17:35.50mwallingjasonwoot:
17:35.52mwallinger
17:35.54mwalling.irssi/logs/6freenode/#asterisk/2008/05/06:08:52 < Madkiss> jasonwoot: basically I am trying to do this: My companies works on two different floors. every department has its own door, and you can open that door right now via the voip-phones and a wrapper script. i want to define two groups in the database: floor 1 and floor 2. whenever a member of one group calls numer "95", i want the door on the floor to be opened where the user is at that time.
17:36.10mwallingtime stamp in EDT
17:36.24*** join/#asterisk notbright (n=chatzill@unaffiliated/yourname/x-837320)
17:37.29tzafrir_homeI got "strange disconnects" on outgoing Zap/FXO calls. Looking further at them I saw that the reason for disconnecting was the time-out parameter for Dial
17:37.43JayTee52[TK]D-Fender, if I see anything else I'll let ya know
17:38.03tzafrir_homeBut the call has already started. Isn't an FXO call considered "answered" as soon as there's audio?
17:38.28jasonwootTY mwalling
17:38.39mwallingthank grep :)
17:38.49[TK]D-FenderJayTee52: thx
17:40.01[TK]D-Fendertzafrir_home: its answered as soon as Zaptel can reserve the channel or until "callprogress=yes" is set and it stops "ringing" without giving a negative prorgess tone.
17:40.49tzafrir_homecallprogress is not set
17:41.16tzafrir_homeah, there it is: answeronpolarityswitch=yes
17:42.02mwallingjasonwoot: you want the log from that day or do you have it?
17:42.10tzafrir_homewonders if there's any way to detect that one....
17:45.10notbrightDoes anyone have an asterisk safe restart crontab srcript lying around somewhere?
17:46.08DaveCanoesigh.  I feel so invisible on this channel.
17:47.16badcfeanyone knows what a VNx is .. like a VN4 .. ?
17:47.35*** join/#asterisk Strom_C (n=strom@208.127.172.112)
17:47.55notbrightDaveCanoe: I get that feeling myself sometimes.
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17:51.20DaveCanoedoes anyone know how to modify the insert that the CDR interface does to databases.  In this case, I have multiple asterisk servers inserting CDRs into a database.  I'd like one large CDR table, but I want to know _which_ asterisk server inserted the record.  So I need each asterisk server to insert a static field (ie: asterisk_server_number or somesuch)
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17:57.47[TK]D-FenderDaveCanoe: this is already detailed between the WIKI & BOOK.  Go give them botha  good read.
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18:06.03DaveCanoeI'm on the sip wiki ... and I'm reading about cdr... and I don't see anything that implies I can modify the insert statement or how I should do it.  Maybe you have a url I should look at?
18:06.42bobbymin this new asterisk release 1.6 the DeadAGI is going to be deprecated?
18:12.54Juggiethe answer to that is no and yes
18:13.35*** join/#asterisk talntid (n=t@66.208.251.170)
18:15.44bobbymJuggie:
18:15.47bobbymgood ;) haha
18:16.42*** join/#asterisk emist (n=emist@unaffiliated/emist)
18:17.20JayTee52if I need to add 3 digits to the beginning of my dialed number over a PRI trunk do I just add the 3 digits in front of the dialed extension like this: Dial(Zap/g1/630${EXTEN}) ?
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18:19.55Madkissmwalling: what?
18:20.07Madkissmwalling: i got it sorted out by now btw.
18:20.21mwalling13:27 < jasonwoot> Anyone remember that guy the other day who was using Asterisk to control access to doorways?
18:20.25Madkissmwalling: it's relatively simple after all
18:20.34*** join/#asterisk musarati__ (n=musarati@p549247E3.dip.t-dialin.net)
18:20.36Madkissmwalling: asteriskdb and the right magic in extension.ael do the trick.
18:22.09Madkissmwalling: I am not sure I could offer what jasonwoot is looking for, tho. to open the door, i am calling /usr/local/bin/open_that_door_alread_stupid_culprit ... ;
18:22.12Madkiss;)
18:22.21jjshoe_JayTee52 yes
18:23.15JayTee52jjshoe, thanks
18:23.22mwallingMadkiss: i just grepped the logs, although sounds cool
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18:23.50JayTee52I've been digging through the book looking for a way to set a 3 digit variable and concatenate with the $EXTEN variable but I can't find a specific example or a special function.
18:23.57Madkissmwalling: it's simple mechanical stuff. we got a direct connection between the parport of our server and the door, and all i need to do is to call parport_ctrl with the correct parameters.
18:24.19Madkissmwalling: i even extended the solution today so that any employee can use his company mobile phone to open a door.
18:24.43Madkissmwalling: it's damn fscking cool, but as said -- the asterisk-part of it is really easy once you understood how asteriskdb works.
18:25.08[TK]D-FenderJayTee52: Dial(Zap/g1/${myareacode}${EXTEN})
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18:35.49JayTee52[TK]D-Fender, thanks!!! that's what I needed but couldn't find in the book whether I could just put the two together or whether I needed to use something like an & to join them
18:36.05[TK]D-FenderJayTee52: * variables ARE that dumb.  its all just text
18:36.32JayTee52[TK]D-Fender, and I'd much rather use a variable than hard code it even though the nxx # won't change
18:36.51JayTee52hahahah, one of my coworkers switched the keycaps on my keyboard around.
18:37.14JayTee52I think I'll just leave it since I'm a touch typist
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18:38.02*** part/#asterisk gitguy (n=diego@adsl-134-171.click.com.py)
18:38.11[TK]D-FenderJayTee52: Do the same to him... and then remap them on an OS level to something ELSE completely different.
18:39.16JayTee52hehehe, that's an excellent idea. We already put the BSOD screensaver on his computer and it took awhile for him to catch on.
18:39.52[TK]D-FenderJayTee52: OH.. and then really fuck with him and take a screeshot of his destop, and use that as a wall-paper and move all his files / icons off of it.
18:40.12*** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller)
18:40.13[TK]D-Fender"I can't open my programs!!!!"
18:41.41codefreeze-lapDaveCanoe: hint: use cdr-adaptive; and see if CDR variables can get you where you want to go
18:42.15codefreeze-lapDaveCanoe: uh, sorry, cdr-odbc-adaptive, I meant
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18:44.55JayTee52way back in the late 80's I hacked all the intrinsic commands in command.com on my coworker's PC. duh intead of dir crpy instead of copy, etc.
18:46.14*** join/#asterisk defsdoor (n=andy@defsdoor.gotadsl.co.uk)
18:46.32[TK]D-FenderJayTee52: Yup... ahh the good 'ole days
18:46.39bobbymJayTee52: how did you do that?
18:47.09[TK]D-Fenderbobbym: file edit command.com
18:47.13JayTee52bobbym, it was back in the dos days and I used Norton Editor
18:47.18[TK]D-Fenderbobbym: You could search & replace those directly.
18:48.20bobbym[TK]D-Fender: really, i always wondered that if you want to chance those thing inside command.com you should decompile and compile again the com
18:48.21bobbymhaha
18:48.25bobbymwell good to know
18:50.54DaveCanoecodefreeze-lap: thx.
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19:12.18draygonHm. Is the originate feature removed from the newer asterisk version?
19:12.39minteeI need to GotoIf(Exist(soundfile))
19:15.25minteei understand that the EXISTS checks for a variable
19:15.42minteebut I need to check to see if a specific gsm file exists.
19:16.16*** join/#asterisk ajohnstone (n=ajohnsto@cpc3-walt3-0-0-cust880.popl.cable.ntl.com)
19:16.51hmmhesaysSTAT
19:16.54minteeheh, yep
19:16.59minteethanks, just found it myself
19:17.04*** join/#asterisk l2cache (n=chatzill@117.178.101.97.cfl.res.rr.com)
19:17.27l2cacheAre there any good SIP providers that offer unlimited inbound minutes for a fixed rate per channel?
19:19.01rob0well, there's ipkall, which might be good enough for your needs (DIDs in .wa.us)
19:21.34*** join/#asterisk banzaika (n=banzaika@rrcs-208-105-66-210.nyc.biz.rr.com)
19:22.54l2cacheok :)  any others?
19:23.38[TK]D-Fender~itsplist-us
19:23.39jbot[~itsplist-us] Here are some popular ITSPs (USA) starting with the more respected ones : http://www.teliax.com , http://connect.voicepulse.com , http://www.nufone.net , http://www.broadvoice.com, http://www.jnctn.com , http://www.bandwidth.com , http://vitelity.net
19:24.17l2cachethanks
19:25.01banzaikaanybody experienced a burst of noise (flat line) while on the phone ?
19:26.09banzaikamid-call
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19:35.17hohumhow do I change the amount of time which goes by that asterisk waits for further DTMF digits before it times out?
19:39.34banzaikaSet(TIMEOUT(digit)=10)
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19:40.46hohumTHANK OYUthanks
19:41.07banzaikanp
19:41.40LARefugeeI think it just hit me why I was having so much trouble registering to vonage and fwd through sip from behind my nat.
19:42.36banzaikaround 2: anybody experienced a problem of an annoying sound in mid call (randomly) ?
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19:42.46banzaikalike fax line
19:43.16LARefugeeNo one curious?
19:43.22banzaikago for it
19:43.38LARefugeeAnyone use ethernet bonding on their server?
19:44.01banzaikabonding ?
19:44.18LARefugeetwo nics and the bonding kernel module.
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19:44.46nny_1has anyone implemented or tested a US reverse lookup system similar to http://www.monetra.com/~brad/callerid_shell.agi ?
19:45.33LARefugeeguess not. I'll try googling.
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19:48.06banzaikaheh, bonding might be bad for registrations, imo
19:48.11mercutiovizAnybody know of Asterisk users in the central california? (Fresno, etc.)
19:48.33mercutiovizthe central calif area, that is
19:48.55LARefugeebanzaika: why do you say that?
19:48.56rob0IIUC a bonded pair of interfaces would share a single IP address, each taking turns at the ARP layer.
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19:50.07LARefugeerob0: Interesting. One of my nics is a realtek and I'm forced to use balanced-rr instead of balanced-alb.
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20:00.04LARefugeeDigium is taking forever to get back to me. Anyone ever have a problem with an X100M in the first port position on their TDM400P? Lots of noise on the channel?
20:02.14LARefugeeEverybody playing hooky on Friday?
20:02.54banzaikasorry was afk
20:03.20LARefugeeafk? I'm an irc noob. I figured out np.
20:03.56banzaikafrom what i read about bonding
20:03.58banzaikaIn Linux you can use the bonding kernel module for load balancing or Hot standby. The module combines multiple NICs into a single virtual interface.
20:04.18banzaikawell that's different from rob0 said
20:05.24banzaikafrom my understanding you can use 2 physical layers to act as a single interface where one will take over if another is down
20:05.48banzaikai might be wrong though
20:06.20rob0haha the only place I have a TDM400P, I no longer have telco service. But I did have to take out one of the FXS modules, I think it went south.
20:07.26rob0Um, how does that differ from what I said?
20:07.52QwellLARefugee: Have you called Digium support?
20:08.06minteeis there a function that will wait and listen to a key pressed without assuming it's an extension?
20:08.07Qwelloh, you said you did.  nm
20:08.08LARefugeeOh I know that. I'm just wondering if anyone's had trouble with bonding. I'm going for throughput because I use the server for more than *. I have both nics plugged into the back of my buffalo router and peered with my vonage ata and a downlink to another switch.
20:09.48LARefugeerob0: Did you take it out of the first port? Did you hear rumbling, crackling on the channel?
20:10.29LARefugeeQwell: no e-mailed. Does IAXTEL work?
20:10.50fileno, but you can directly call the PBX
20:11.01banzaika<LARefugee> whats your uplink speed ?
20:11.29fileIAX2/guest@pbx.digium.com/s@default will call the main IVR
20:11.33nny_1does this look correct exten => s,2,AGI(callerid_shell.agi|CALLERID(num))
20:11.50LARefugeebanzaika: no sure what you mean. From switch to switch is 100 mb full duplex.
20:11.52nny_1they had AGI(callerid_shell.agi|${CALLERIDNUM}) as an example, but i don't think asterisk supports that
20:12.06LARefugeefile:  thanks.
20:13.38banzaika<LARefugee> you set as away, fyi
20:14.35plikhmmm, any suggestions how I can get rid of "__auto_congest: Auto-congesting call due to slow response" ?
20:15.10LARefugeeblasted pidgin!
20:15.10nny_1anyone wanna make a quick buck?
20:15.29*** join/#asterisk Katty (n=Katty@hera.copi-rite.com)
20:15.34Kattyhai!
20:15.48nny_1trying to implement the caller_id agi, a little unsure of some of the process, never used agi before.. I can pay someone to take a look and tell me what juju is missing
20:15.49plikhai2u kaitai
20:16.08[TK]D-Fendernny_1: exten => s,2,AGI(callerid_shell.agi|${CALLERID(num)})
20:16.15Kattyi has question about voicemail.
20:16.17nny_1[TK]D-Fender: i'll try that thanks
20:16.27Kattythere's all these audio files. unavailable, busy, and temp greeting.
20:16.28[TK]D-Fendernny_1: Don't forget to "reference" your function calls.
20:16.37dkwiebeplik:  I belive you turn "qualify" off
20:16.49dkwiebeplik: it's been a while though.
20:16.53[TK]D-Fendermintee: "core show application read"
20:16.57[TK]D-FenderKatty: Mew.
20:16.57QwellKatty: you can't.  I'm guessing at the question.
20:16.59plikdkwiebe: thanks, I'll have a look
20:17.05Kattyi see there are options to switch between the greetings, u, b, etc.
20:17.06nny_1exten => s,2,AGI(callerid_shell.agi|${CALLERID(num)})     exten => s,3,NoOp(AGI Returned ${lookupname})   exten => s,4,Set(CALLERID(name)=${lookupname})
20:17.07Kattyoh :/
20:17.15Kattyreally? :<
20:17.18QwellKatty: continue though - you aren't asking what I thought you were :p
20:17.20Kattyyou can't do that? :<
20:17.22Kattyoh!
20:17.23Kattyphew.
20:17.27Katty/so/
20:17.30Qwellunless you are, then no
20:17.33Kattyhow do i check what the phone is doing /first/
20:17.48Kattyon reject, go to voicemail(stuff,b)
20:17.50Qwellsee the stdexten macro
20:17.56Qwellit does exactly that
20:17.59Kattyjbot: stdexten?
20:18.08Qwellmacro-stdexten in the sample config
20:18.11[TK]D-Fendernny_1: You might want to do some "lookup failure" safety checks, but thats otherwise sane.
20:18.28[TK]D-FenderKatty: "core show application chanisavail"
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20:18.40Qwell[TK]D-Fender: don't even need to do that, if you're already dialing the phone
20:18.53Kattyoooh
20:18.54Qwell(another assumption on my part, I guess)
20:18.55Kattyi see how to do this
20:19.03[TK]D-FenderQwell: depends on your definition of "busy".
20:19.19[TK]D-FenderQwell: ....
20:19.22[TK]D-Fender~assume
20:19.23jbotAbout assumptions : http://www.moviewavs.com/php/sounds/?id=gog&media=WAVS&type=Movies&movie=Long_Kiss_Goodnight&quote=assume.txt&file=assume.wav
20:19.51nny_1[TK]D-Fender:  seemed to work, need to try it with a number that is listed by 411 google etc
20:20.11Kattywhat about this Temp Greeting, thing?
20:20.16rob0My bum FXS was preventing the zaptel drivers from loading, so I had to remove it or not have zaptel.
20:20.26*** join/#asterisk shmaltz (n=chatzill@mail2.dmaven.com)
20:21.04[TK]D-Fenderrob0: foreign objects should not be plugged into your bum....
20:21.36[TK]D-FenderKatty: "I'm on vacation and don't want to have to re-record my normal greeting when I come back"
20:22.04Katty[TK]D-Fender: yes, but is it...
20:22.05Kattyu
20:22.15Kattyor...b
20:22.18Kattyor what is it lol
20:22.24Kattyor does it just /play/ if it's there and ignores all others
20:22.39LARefugeerob0: fxs? That would be a s100m module.
20:22.43[TK]D-FenderKatty: it overrides b/u
20:22.57Kattycheers
20:22.58Kattyi love it
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20:23.59DrkShadowdoes anyone know the url/name of that packte5 software? I swear the name was "Packet5".. but I can't find anything on it.
20:24.12*** join/#asterisk LARefugee (n=victorr@c-76-104-191-194.hsd1.wa.comcast.net)
20:24.26rob0[TK]D-Fender: No wonder it didn't work!
20:24.34*** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br)
20:25.07LARefugeedigum shouldn't list iaxtel numbers on their website. It makes the rest of us "hope".
20:25.46nny_1hmm agi script returns zero...
20:25.57nny_1<PROTECTED>
20:26.03*** join/#asterisk MikeJ (n=MikeJ@33.203.64.208.static.accentrainc.com)
20:26.16nny_1is there a way to debug the agi?
20:26.27[TK]D-Fendernny_1: "agi debug"
20:26.35nny_1ha
20:26.41nny_1yeah so damn intuitive it hurst ty
20:26.43nny_1hurts too
20:27.40[TK]D-Fenderok, checkout time here, heading home.  later all
20:29.26unpaidbillso im buying one of these handytone 286s
20:29.30*** join/#asterisk ikevin (n=kevin@2001:5c0:99b9:0:20f:b0ff:fe4a:4589)
20:29.36unpaidbilli hope to be impressed.
20:29.49LARefugeeis this chat archived?
20:31.17banzaikadon't think so
20:31.30banzaikasome clients do that by default
20:31.42Qwell~logs
20:31.43jbothmm... logs is apt/ibot/infobot/jbot/purl all log daily to http://ibot.rikers.org/<channelname>/ where channelname is html encoded ie: %23debian | lines that start with a space are not shown | some channels have stats at http://ibot.rikers.org/stats/<channelname>.html.gz
20:31.46Qwellit is
20:33.21banzaikagood to know
20:33.25banzaikathx
20:33.45unpaidbill~handbook
20:33.45jbot[handbook] http://www.digium.com/handbook-draft.pdf
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20:36.02Qwellunpaidbill: you will be largely impressed
20:36.03Qwell~gs
20:36.04jbotGrandSuck phones & gateways are cheap junk which should be avoided with extreme prejudice.
20:36.09draygonanyone feeling kind enough to help me with incoming call issue on asterisk?
20:36.33LARefugee~digium
20:36.34jbotdigium is probably the creator, primary developer, and maintainer of Asterisk.  They have a full-time team dedicated to open source Asterisk development which carries the majority of the load.  They also sell various hardware and software products, as well as provide support and development services.  http://www.digium.com/
20:37.09LARefugeejbot: I just got a GS bt200. I'm liking it.
20:37.10jbotYou just got a GS bt200. I'm liking it.?
20:37.10banzaikadraygon: post questions if people have answer you'll hear it
20:37.34*** join/#asterisk danalien (n=danalien@unaffiliated/danalien)
20:38.42LARefugee~contact
20:38.58LARefugee~digium support
20:39.33LARefugeeI lost that connect to pbx.digium.com. I'll hack at it. file?
20:40.35fileIAX2/guest@pbx.digium.com/s@default will call the main IVR
20:40.42LARefugeethanks!
20:44.07rob0and without callerID ! :)
20:44.15*** join/#asterisk Cresl1n (n=matt@216.207.245.1)
20:44.15*** mode/#asterisk [+o Cresl1n] by ChanServ
20:46.04LARefugeeon teh phone with digium now. thx.
20:50.01*** part/#asterisk lirakis_work (n=lirakis@65.200.191.241)
20:50.09rob0I'm trying to decide between buying a FXO+FXS ATA, or moving my existing TDM11B. I don't have a dedicated machine to run as server, so it has to be shared on a dual-core AMD64 which is also functioning as a KDE workstation.
20:50.55rob0I know the recommendation is no X11 on a zaptel+asterisk machine, is that still valid?
20:52.19*** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk)
20:52.24banzaikaX = bad on any server type machine.
20:52.27danpthe recommendation is probably to not have things you don't really need
20:53.12rob0haha
20:55.35*** join/#asterisk mort___ (n=mort@user-54446aa1.lns4-c10.dsl.pol.co.uk)
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20:58.20*** join/#asterisk bakermd (n=none@38.104.0.102)
20:58.33bakermdWhere should I ask questions regarding Polycom phones?
20:59.30b11d`here
20:59.49b11d`everyone here loves Polycom.. fire away :)
20:59.49rob0or at Polycom support
21:00.03b11d`or your preferred Polycom vendor ;)
21:01.10bakermdIP 501 - bootrom 4.0.0.0423 - SIP 2.1.2.0078 :: Manually assigned IP address... can ping it from the network - it will not register. I would like to use the web interface for the phone, but it will not come up - as if it is not running
21:02.43bakermdI can do SIP tracing on my Asterisk box to get the registration working, but why would it not allow me to connect to the HTTP access? (I need to have 2 lines connecting to 2 servers, so at some point I will have to get to the interface...)
21:02.48b11d`i had that problem once, had to "reest to default" the configuration in the setup menu..
21:03.07b11d`~b11d
21:03.08jbotb11d is a constant source of misinformation...
21:03.10b11d`:)
21:03.22bakermdlol - okay ;)  What is the more correct answer?
21:04.34fskrotzkiyou assigned the IP address but did you get the netmask correct?
21:05.04bakermdfskrotzki: Yes, simple class C.. 255.255.255.0
21:08.18*** join/#asterisk RoyK (n=roy@ti211310a080-7540.bb.online.no)
21:08.24*** join/#asterisk adjohn (n=adjohn@i220-221-4-188.s05.a013.ap.plala.or.jp)
21:08.33*** join/#asterisk [TK]D-Fender (i=Joe@64.235.218.194)
21:09.41bakermdSo I guess I am confused - is using "reset to default" sound advice for fixing the IP 501 issue? (Pardon my ignorance - I do not know if jbot is actually a bot or not...)
21:10.23[TK]D-Fenderjbot: are you a dog?
21:10.31[TK]D-Fender~areyouadog
21:10.32jbotBark! Bark!
21:10.37rob0jdog!
21:10.51bakermdgotcha ;)
21:12.53_ShrikE~botsnack
21:12.53jbot_ShrikE: :)
21:13.03bakermdb11d`: Reset local config or device setting?
21:14.06bakermdResetting local config.
21:14.38talntidThere was a problem connecting to mail.rtuinfo.com
21:25.25*** join/#asterisk vector (n=vector@host-178-246-220-24.midco.net)
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21:50.56Tako-sanAnyone have a recommendation for a multi-port ATA device?  In total we have 16 FXO ports that we want to connect to Asterisk via IP.
21:51.07Tako-san16 FXS ports rather
21:54.02[TK]D-FenderTako-san, 2 x Linksys SPA-8000 @ $230 ea
21:55.15Tako-san[TK]D-Fender: Really!  You would recommend Linksys.  Ok, if they will do the job.
21:55.26rob0I'm trying to decide between buying a FXO+FXS ATA, or moving my existing TDM11B. I don't have a dedicated machine to run as server, so it has to be shared on a dual-core AMD64 which is also functioning as a KDE workstation.
21:55.41Tako-san[TK]D-Fender: Thanks.  Will look into that product.
21:55.49[TK]D-Fenderrob0, pile it on in good health.
21:56.02rob0(I asked that during your drive home.)
21:56.20[TK]D-Fenderrob0, My server hosts mail, web, ftp, routing, *, my 120" HT setup, and used to make me coffee.
21:56.21rob0you think the TDM would be okay?
21:56.38*** join/#asterisk djs26 (n=djs@unaffiliated/djs26)
21:57.12[TK]D-Fenderrob0, should be fine
21:57.57rob0cool, thanks (it's worth a try anyway, and if I have trouble I can go with the ATA.)
21:58.41*** part/#asterisk Cresl1n (n=matt@216.207.245.1)
21:58.47*** join/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com)
22:00.43muiropastebin seems to be... down. Any alternatives?
22:01.02rob0~pastebin
22:01.03jbot[~pastebin] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste
22:01.05Tako-san[TK]D-Fender: Are there other ATA devices you would recommend?  Are there any shortcoming of the SPA-8000?  You hve used it in a production environment yourself or have heard (good) things about it?
22:01.25[TK]D-FenderTako-san, Multiple people here have used it and love it.
22:01.46[TK]D-FenderTako-san, good size, flexible wiring, inexpensive.
22:02.05Tako-san[TK]D-Fender: Ok.  The reason I am being a little cautious is when I hear the name "Linksys" I immediately think of consumer level routers/wireless access points.
22:02.59Tako-san[TK]D-Fender: But I am happy to hear the there are numerous people in this channel who are actively using the device and happy with it.  Cheers.
22:04.13muiroQuestion: can I not compare a variable to "*" in GotoIf() or if() as such: http://rafb.net/p/GA2gXP42.html
22:05.22[TK]D-Fendermuiro, Could be ok.  that of course depends on the state of variables & priorities
22:07.06muiro[TK]D-Fender: everytime I try to make the comparison, I get this error: http://rafb.net/p/1KARsK53.html
22:07.56[TK]D-Fendermuiro, And those single line warnings aren't doing any good.  look at the BIG PICTURE. I jsut told you that the STATE of your variables can cause that to fail.
22:08.09[TK]D-Fendermuiro, pastebin the ENTIRE deal at verbose 10
22:09.33muirosame error at verbose 10
22:10.31muirolooking up variable states on voip-info.org then I'll ask again with more info
22:11.51*** join/#asterisk RoyK (n=roy@ip-42-56-149-91.dialup.ice.no)
22:13.03*** join/#asterisk JayTee52 (n=jforde05@c-69-243-161-112.hsd1.in.comcast.net)
22:16.37muiro[TK]D-Fender: you wouldn't be able to point me in the direction of any documentation regarding variable sattes would you?
22:16.51muiro[TK]D-Fender: or anything that would be pertinent
22:16.52[TK]D-Fendermuiro, prove to me the variable isn't BLANK <-
22:17.19rob0noop++
22:17.25[TK]D-Fendermuiro, because the way * would evaluate that would leave nothing on the other side of the "="
22:17.37muiro[TK]D-Fender: I test for null directly before, but I'll throw a noop a little closer if it helps
22:18.03[TK]D-Fendermuiro, pastebin the pwhole bloody thing.  this 1 line at a time deal isn't getting you anywhere.
22:18.20muiro[TK]D-Fender: no need to be hostile
22:20.39JayTee52muiro, he wasn't being hostile.
22:21.14muiro"bloody", but I'm the one basically begging for assistance so I can't really complain
22:21.40JayTee52if he was actually being hostile he probably wouldn't be trying to help you
22:22.09[TK]D-Fendermuiro : small tip then.  When someone asks you for something to help you with a problem, that probably isn't a good time to start abstractly delcaring what is, and is not relevent.
22:22.25[TK]D-Fendermuiro, Because if you knew.... you would have a problem.
22:22.29[TK]D-Fenderwouldn't*
22:22.32muiroI didn't say it wasn't relevant. I'm currentl collecting the log
22:23.40[TK]D-Fendermuiro, See I also don't know that your null test is any good either.  A single error "message" doesn't prove much.  so jsut place another call and pastebin the entire thing along with the dialplan that generated it.
22:24.08muiroI am
22:24.14[TK]D-Fendergreat
22:25.48JayTee52wishes he had the BBC channel.
22:26.32muiro[TK]D-Fender: http://rafb.net/p/N8LCyv42.html
22:27.21muiro[TK]D-Fender: also earlier I null out MENU_CHOICE
22:27.59muiro[TK]D-Fender: I could put that in if you think it might be the problem
22:29.47plikJayTee52: which BBC Channel... depending where you are you can get some online at bbc.co.uk  or zattoo.com
22:29.53[TK]D-Fendermuiro, how about nooping it right before your test...
22:30.30JayTee52I'm in Indiana, I'd have to pay extra to Comcast for it.
22:30.33[TK]D-Fendermuiro, But this would be better regardless : exten => s,n,GotoIf($["${MENU_CHOICE}"="*"]?pds_main_menu,s,1)
22:31.02plikJayTee52: oh, those prolly won't work then - unless you can find a uk based proxy ;)
22:31.28JayTee52I wanna watch this supposedly "cheesy" scifi show called Hyperdrive
22:32.28plikJayTee52: uknova will probably have torrents of it shortly
22:32.55JayTee52plik, I'll keep my eye out for them, thanks!
22:33.04pliknp :)
22:33.29muiro[TK]D-Fender: can do, making te call
22:35.48muiro[TK]D-Fender: earlier I attempted to double quote the "*", but adding the double quotes around ${MENU_CHOICE} fixed her up. Thanks for the debug.
22:36.37*** join/#asterisk _polto_ (n=polto@elphelut.fttp.xmission.com)
22:36.44_polto_hello all
22:36.44muiro[TK]D-Fender: I wasn't sure using double quotes would amount to much, asterisk not really doing much typing, at least not on the dialplan side
22:36.51[TK]D-Fendermuiro, need it on both sides so that both have the "".  its a literal char in the comparison, not an actual type-cast
22:37.05muiro[TK]D-Fender: gotcha, thanks much
22:38.32_polto_can somebody help pls ? I am trying to configure a SIP trunk. My PBX is behind a GNU/Linux firewall with SIP helper. (it work for a SIP trunk provider). I am trying to connect a Sipura3000 located on the internet (real address, not NAT) as SIP trunnk.
22:39.06[TK]D-Fendermuiro, You're welcome
22:39.07_polto_[May  9 16:33:47] NOTICE[22459] chan_sip.c: Registration from 'elphelh <sip:lalala@192.168.1.16:5060>' failed for 'xxx.yyy.zzz.www' - No matching peer found
22:39.18_polto_sorry
22:39.30[TK]D-Fender_polto_, undo any "sip helper", and follow this guide :
22:39.32[TK]D-Fender~sipnat
22:39.32jbot[~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions
22:39.37_polto_this is the message I got in the Asterisk log.
22:40.01[TK]D-Fender_polto_, And its not ID-ing your device. looks like you didn't set it up right
22:40.27*** join/#asterisk jm|home (n=jm|home@zen.jamiem.com)
22:40.40_polto_[TK]D-Fender, on witch side? and why does it speak about the local address ?
22:41.01[TK]D-Fender_polto_, local address is because you didn't set up * to properly handle NAT.  Follow the guide
22:41.21[TK]D-Fender_polto_, and the other issue could be either side.  the problem is they don't both agree
22:42.46_polto_[TK]D-Fender, thanks!
22:42.50_polto_i am reading now.
22:48.59*** join/#asterisk fluff (n=dune@snowflake.fluffigt.net)
22:49.14_polto_do somebody know what can help ? sometimes my sipura3000 does not hangup the line properly and it's occupied. If I software reboot the Sipura the line is not released. I need to physically disconnect it and reconnect again.
22:49.35_polto_Is there a way to release the line on sipura3000 remotely ?
22:54.21[TK]D-Fender_polto_, log into it and tell it to reboot.
22:56.52*** join/#asterisk DarkRift (i=dark@bas10-montreal02-1177583659.dsl.bell.ca)
22:58.47_polto_[TK]D-Fender, I can log only with the web interface and if I apply changes and reboot, it does not hangup the line :(
22:59.19[TK]D-Fender<PROTECTED>
23:05.08_polto_[TK]D-Fender, I mean if I call the line is occupied and the only way is to reboot sipura3000 physically
23:06.04[TK]D-Fender_polto_, But does the SPA *say* thats its busy?
23:08.18_polto_[TK]D-Fender,  how to know it ?
23:08.31[TK]D-Fender_polto_, look on the status screen
23:09.54*** join/#asterisk mitcheloc (n=mitchel@216.207.245.1)
23:11.44_polto_[TK]D-Fender, I suppose it's that : "Hook State:On"
23:12.16*** part/#asterisk muiro (n=muiro@cpe-76-189-84-108.neo.res.rr.com)
23:12.49_polto_sorry, the PSTN line status is Hook State:Off.
23:13.35_polto_[TK]D-Fender, PSTN State:PSTN Caller Accepted
23:16.03*** join/#asterisk wordzilla (n=me@122.109.126.193)
23:18.26*** join/#asterisk workaphobia (n=workapho@ool-457fa98d.dyn.optonline.net)
23:23.01*** join/#asterisk Simon-- (n=sim@staff-nat.netnation.com)
23:23.29*** join/#asterisk angom (n=angom@201.170.65.143)
23:23.43Simon--anybody know which variables other than cid name, cid num are passed along an iax channel without patching in generic variable passing?
23:23.57Simon--I don't feel like maintaining a patch just to pass a few other things..
23:30.29*** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com)
23:36.37*** join/#asterisk seanbright (n=sean@c-69-251-175-43.hsd1.md.comcast.net)
23:46.48*** join/#asterisk colinm_ (n=colinm@VDSL-130-13-116-41.PHNX.QWEST.NET)
23:48.54_polto_[TK]D-Fender, I think the issue with my trunk is just as stupid as port number. Sipura is connected is extension on port 5060, but use the port 5061 for the trunk. I never seen it before.. On my Asterisk box he message I see is : chan_sip.c: Registration from 'elphelh <sip:lalala@192.168.1.16:5060>' failed for 'xxx.yyy.zzz.www' - No matching peer found
23:49.01_polto_port 5060
23:49.13_polto_may it be the reason ?
23:50.03[TK]D-Fender_polto_, could be.  each peer needs its port set on those.
23:50.17_polto_oh..
23:54.52*** join/#asterisk mindCrime (n=chatzill@cpe-065-190-188-124.nc.res.rr.com)

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