irclog2html for #asterisk on 20060423

00:08.44dlynesdamn....dead in here today
00:10.38Drukenyep
00:11.24dlynesprobably everyone's out enjoying the nice weather like i was :)
00:11.34dlynesgood day for a drive and rollerblading all that stuff
00:11.51dlynesIf it was a little warmer outside, I'd go kayaking instead
00:12.51*** part/#asterisk kamileon (n=kamileon@68.62.190.253)
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00:15.36Drukenraining here
00:17.24dlynesi wonder why if amp/freepbx/... peeps are told to go to #freepbx, why aren't centos users told to go to #centos?
00:19.07Drukenbecause centos is a distro, wheres amp is an asterisk related "package" with alot of crap that only the amp/freepbx knows.... we support the basic asterisk program
00:19.48dlynesI thought centos wasn't just a distro?  I thought it was a distro with a bunch of asterisk cruft with it such as AMP, Flash operator panel, ..?
00:20.47Drukendunno, i'm a slackman
00:20.48dlynesOr is that just when it's part of Asterisk@Home?
00:20.59dlynesYeah....same here
00:21.13dlynesi'm just itching to get my hands on Slackware 11.0
00:28.19Delmardlynes, that problem yesterday went away after i did a full package update, and installed the latest kernel and headers... the kernel and headers were installed correctly according to dpkg but who knows.. musta got messed up i reckon
00:28.38Delmardlynes, was u i was talkin to wasnt it? think it was
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00:35.23*** part/#asterisk _-_ (n=nabudoco@dpc674729057.direcpc.com)
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00:51.13ariel_hello everyone
01:00.08ariel_Seems very slow here tonight
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01:24.20Hmmhesaysa bit
01:24.22Hmmhesaysi'm going to the bar
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01:41.39Drukeni'm gonna open a bar, i'll call it the drunken monkey... think anyone would go?? hehe
01:47.12*** part/#asterisk mitcheloc (n=mitchelo@ip67-153-163-202.z163-153-67.customer.algx.net)
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01:49.36ariel_Druken, are you sure you don't want to call it spank the monkey.
01:50.08Drukenpositive :)
01:50.38Drukennot unless it's a ladies only club....
01:50.58Drukencourse, then i'd never leave :)
01:54.58*** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com)
01:56.40brodiemI'm trying to configure a TDM400P w/ FXS modules. I'm wondering what it is I need to do so that I get a dial tone when a telephone is plugged into one of the ports. Basically when I pick up the phone it does whatever is configured as an incoming call to the context specified for that channel in zapata.conf
01:57.36Qwellbrodiem: immediate=no
01:59.30brodiemahh thanks
02:00.31Drukenyou can use a fxs as a hotline?
02:00.41Drukensweet, i didn't know that
02:00.43QwellDruken: batphone?  sure
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02:00.51brodiemDruken I guess I just found out that you can :)
02:01.01brodiemimmediate=yes lol
02:01.01Qwells,1,Dial(SIP/batman)
02:01.09Drukencourse, fxs sucks ass
02:04.33brodiembest thing for fax besides a seperate pots though..
02:06.43Drukenagreed, however, my fxs ports die all the time
02:06.58Drukenfucken fax very rarely has a dialtone
02:07.40tzangerDruken: on what
02:07.51Drukenon my tdm card
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02:08.19Drukenfxs port usually lasts about 10 days... then i have to reboot to fix it
02:09.22coppicesome people seem to get that, and some don't. nobody seems to investigate the cause, though
02:09.44Drukenpersonally, i'd never buy a tdm card ever again
02:10.02Drukeni have a tdm and 2 fake x100p's in the same machine
02:10.12Drukenand the fake cards blow the shit out of the tdm
02:10.43Drukenthe fxo on the tdm, i get echo on incoming calls, and feedback, and static, and it's just a peice of crap
02:11.04Drukenyet with the exact same settings, the fake cards work like a dream
02:11.10coppicedepends which clone you get. some have poor analogue front ends. the good ones are pretty good, though
02:12.10Drukeni guess
02:13.40tzangerDruken: what rev of TDM card do you have?
02:13.52Drukenno idea. purchased it about a year ago
02:13.56tzangerdmesg should tell you
02:14.04tzangerdoes it have RJ45 jacks or RJ11?
02:15.00Drukeni belive it has the rj45 jacks
02:15.21Drukenit's a pain in the ass removing the rj11 clips from the card
02:15.52tzangerthat's an older card, but generally it's not the card that has the issues, it's the FXO modules. You can try putting a 0.22uF cap across the reset and ground pins on ONE of the modules (they're all in parallel),I can dig up the pinouts if you want
02:15.52Drukenany other way to find the rev number aside from dmesg?
02:15.56tzangerit was fixed in a later version
02:16.03tzangerDruken: yeah, pull the card and read the board :-)
02:16.23Drukennot feasable at the moment in time...
02:16.38tzangerDruken: actually "zap show status" should show it on a recent version of *
02:16.53tzanger*CLI> zap show status
02:16.53tzangerDescription                              Alarms     IRQ        bpviol     CRC4
02:16.54tzangerWildcard TDM400P REV E/F Board 1         OK         0          0          0
02:16.59tzanger*CLI> show uptime
02:16.59tzangerSystem uptime: 10 weeks, 5 days, 34 minutes, 50 seconds
02:17.04Drukenhehe
02:17.12Drukencommand not found
02:17.18tzangerI have *zero* trouble with my TDM400s
02:17.23tzangerbut then again, NONE have FXO
02:17.34Drukenrev h
02:17.51Drukengood ol zttool told me :)
02:17.56drrayhey Druken
02:17.57tzangerok it's a newer carrier, I wonder if you have old FXO modules.  they were extraordinarily sensitive to noise on the RST/ line
02:18.12Drukenhey drray
02:18.12tzangerwhich is what the cap is for (not a great way to do it but works)
02:18.22drraydid you ever get your payphone going?
02:18.42Drukennope... the board you sent me was for a newer phone...
02:19.00Drukenmines an old western electric type
02:19.23drraydamn, that's too bad
02:19.32Drukentzanger: what is that cap your talking about?
02:19.54Drukendrray: yeah, i was a bit disapointed, but oh well, storey of my life lately
02:21.10tzangerDruken: 0.22uF cap across the RST/ and common pins of ONE of the modules (any one)
02:21.19tzangerI think it's pins 2 and 20 but I'll have to look
02:21.36Drukenyou lost me...
02:21.40tzangerDruken: :-)
02:21.51coppiceTDM400s don't really work for FAX. I think its a driver problem, but for most people they keep frame slipping.
02:21.52Drukenhehe i can put the cards together, but my understanding of them stop there
02:21.54drraytzanger - if you could look and tell me that pinout
02:22.35Drukencoppice: when the fxs works, the fax works fine...
02:23.02Drukeni'd prefer to have a t100p and use my channelbank, but i'm poor
02:23.11coppiceseems OK for some people, but not for most. maybe its depends on PCI settings or something.
02:23.46drrayI had a tdm400 that worked for 6 months for me, then I lend the card out and get Drukens 10 day reboot lockup
02:24.09Drukenisn't that a fun "feature" ?
02:24.23coppiceI have a little test program that looks for slipping. It seems to show slipping for most people who tried it. Is your faxing really solid, or are you relying on ECM to correct until it works. i.e. if faxing far slower than it should be
02:24.31Drukeni gave up after a while, just vowed to never buy one again
02:25.04coppicethe snag with most reports of rock solid behaviour is the reporter has actually never bothered to look and notice all the problems
02:25.07drrayI've since moved on to a govarion tor card and a adit 600
02:25.12Drukencoppice: no idea, it's an older fax machine.... but generally works good
02:25.48drrayg2 or g3?
02:26.52coppiceif you have a g2 fax, donate it to the smithsonian :-)
02:27.02tzangercoppice: I tried that, I get no slipping ... that's the one that listens to the line, right?
02:27.05tzangermeasures delay?
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02:27.47coppiceyeah. I can run it for 10-15 minutes sometimes without a slip. then it will start slipping again
02:27.52tzangerhmm
02:27.55tzangerI've never run it that long
02:27.58tzangerI'll have to give it another shot
02:28.31coppiceits doing autocorrelation on the card's own echo to detect the loop delay
02:28.53tzangeryeah it's spitting out white noise and listening, right?  I think... it's been a while
02:29.32Drukenuhg.... transfering data takes too damn long...
02:33.14coppiceright. i looks for the peak in the autocorrelation of an AWGN signal it pumps out
02:36.52Drukenman... i am so bored it's not even funny
02:38.34xachenSo yeahhmm
02:38.37xachenwoops
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02:45.04dlynesDelmar: which problem was that again?
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02:55.32coppicelife. life is always the biggest problem
02:56.51Drukenno... life isn't the problem... it's generally the women in your life.. hehe
02:57.20coppicearen't the women much less of a problem if either you or they are not alive?
02:58.05Drukenvery true
02:58.30Drukenhowever, i am not planning on being not alive anytime soon....
02:58.47coppicei try to take my mind off beautiful women, but I find it hard :-)
02:59.55Drukenhehe beautiful women are usually good, cause being not a beautiful man, i would have no chance in smacking it anyways...
03:00.41Sedoroxhehe.. I'm playing dominoes online with a beautiful women right now :p
03:00.58Drukeni was playing pool, but it got boring
03:01.01Drukenhehe
03:01.02dlynescoppice: How can you possibly take your mind off beautiful women, when you live in hong kong?
03:01.17Sedoroxlol
03:01.18QwellSedorox: It's a guy
03:01.21Drukenmaybe he doesn't like orientals?
03:01.26coppicedlynes: duh! why do you think I live here?
03:01.31dlyneslol
03:01.32SedoroxQwell: considering I know them in life... naaa :p
03:01.44QwellSedorox: are you SURE? :p
03:01.46dlynesHong Kong's got some of the most beautiful chinese girls around
03:01.52dlynesSame with Vancouver (where I live)
03:02.13SedoroxQwell: hehe, yes... :p we've spent a lot of time together (in fact.. thinking of asking her out soon, but anyway) :p
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03:06.46coppicedlynes: try singapore. similar girls to HK, but they dress sexier :-)
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03:07.50QwellTry CA
03:08.01QwellCan't go but 10ft without running into one :p
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03:08.28coppicethey don't look so good though
03:09.00QwellI'm just talking in general.  Can't go 10 ft without seeing something worth looking at :p
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03:10.05coppiceamerican chinese don't look like other chinese
03:10.07coppiceamerican europeans don't look like other europeans
03:10.08coppiceamerican blacks don't look like other blacks
03:10.10coppiceis it just the diet?
03:10.26dahunter3I'm looking for a function to strip all characters except for numbers, does one exist or should I roll my own?
03:10.41Qwelldahunter3: REGEX, I believe
03:11.36Drukencoppice: fat looks like fat :)
03:12.05coppicefat is certainly a part of it, but only a part.
03:12.35coppiceQwell: due to our higher population density, I don't need to travel as far as you :-)
03:12.47dlynescoppice: Yeah, but most of the Chinese here are fresh from China, Hong Kong, or Taiwan
03:12.48Nugget"A lot of people, when they have a problem, think 'I know, I'll use regular expressions!' Now they have two problems." -- jwz
03:12.58dlynescoppice: Most of them can barely speak a word of English
03:13.39QwellNugget: :p
03:13.48xachenAnybody know of a good way to merge Monitor sessions into one?
03:13.49dlynescoppice: But the Chinese that have been here a while longer are usually a lot fatter, and don't seem to care what they look like
03:14.12dlynesxachen: sox...it's in the wiki for Monitor on how to do it
03:14.26coppicedlynes: I know a few american chinese who went there is their teens, and still look very american a few years later
03:14.35xachenI'm doing call audits on our support staff and need to record them ;)
03:14.50dlynescoppice: Yeah..it all depends on how fast they blend in
03:14.59dlynescoppice: And how young they are when they come here, too
03:15.23dlynesxachen: You could always use MixMonitor, too...apparently that nasty core dump/segfault problem has been fixed now
03:16.08*** join/#asterisk opus_ (n=opus@dahphish.org)
03:16.18coppiceof course people always tend to blend in. when I first came to asia I had a mass of blond hair, and really stood out. now i have less of that :-)
03:16.18opus_hello
03:16.19dlynescoppice: The one girl I was engaged to, after she'd been here for three years, I was more Chinese than she was Canadian
03:16.38dlynescoppice: iow, she hadn't Canadianized at all
03:17.01coppicein vancouver? of course not
03:17.10dlynescoppice: Heh
03:17.25dlynescoppice: Yeah...you can live here for years with only knowing either Cantonese or Mandarin
03:17.28coppicesame would happen in toronto
03:17.46dlynescoppice: You don't need to know English to survive in Vancouver
03:18.12coppicesome mainlander moving to toronto find they don't need to learn english. they need cantonese
03:18.23dlynescoppice: Yeah...easier to find a job
03:18.51dlynescoppice: Another Chinese person will give you a job much faster than a Canadian will, because the Canadian doesn't know the value of your education
03:19.19dlynescoppice: Nobody here has ever heard of Tsinghua University, or Nanjing University, ...
03:20.03dlynesWell, that and they wouldn't know the difference between Tsinghua and Beijing universities
03:27.53coppicei was in nanjing this week
03:28.52dlynesheh...you're making me jealous
03:28.59dlynesI want to go back to china again
03:29.23dlynesJust no desire to live there...the pollution was way too bad
03:29.27coppicei'm not sure of the value of a chinese education. I've met some real dummies who graduated from tsinghua
03:29.44coppicedlynes: where are you from?
03:29.59dlynescoppice: I don't know....most of the guys I know that graduated from Tsinghua are quite smart, but they're way too arrogant
03:30.06dlynescoppice: Ontario
03:30.26coppicei thought you said you want to go back to china?
03:30.32dlynesYeah, I do
03:30.35dlynesFor another visit
03:30.38dlynes=)
03:30.47dlynesI've been there before, and loved it
03:31.00dlynesJust don't think I could handle it for long stretches because the pollution is so bad
03:31.07coppiceoh. most of the cities are much cleaner than a few years ago, except beijing
03:31.28dlynesBeijing has a mandate to clean it up by 2008(?), too
03:31.38coppicebeijing is suffering huge dust problems, due to the drier climate to the west
03:31.43dlynesIncluding getting rid of all the coal fired furnaces
03:31.56coppiceyou might have seen that in the news this week
03:32.01dlynesWhen I was there, the coal dust on the balcony in the morning was absolutely horrible
03:32.14dlynesIt was usually about 1/8" thick
03:32.17coppiceyou don't get much of that now.
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03:32.25dlynesYou could wipe it off, and the next morning it would be back
03:32.39dlynesAll the poorer tenements used coal
03:33.07coppicemany cities a few years ago were thick with dust from construction. now they use nets to control that
03:33.49coppicewell, beijing has a big problem with the olympics, because there is no way they are going to control these dust storms
03:33.51dlynesHangzhou apparently still has a major pollution problem, though
03:34.07dlynesThat;'s what a couple of my friends tell me that live there
03:34.19coppicehangzhou is a beautiful place.
03:34.30coppicethe air is really clear there.
03:34.32dlynesYeah...I've seen pictures of West Lake
03:34.40dlynesIt's one of the most beautiful lakes I've seen
03:35.17coppicewest lake was a horrible mess a couple of years ago. they were constructing hotels and conference centres and masses of touristy things there. its settling down now
03:35.18dlynesWell, besides
03:35.35dlynesHangzhou is famous for many beautiful things :)
03:36.09coppicethey are trying to turn hangzhou into a very academically oriented area
03:36.39dlynesMaybe that'll make it more voip friendly then
03:36.48dlynesGovernment-wise
03:36.52coppicevoip is outlawed
03:37.08dlynesThat's my point :)
03:37.23dlynesBut it's not totally outlawed
03:37.27dlynesIt's still allowed
03:37.37dlynesJust not for outbound calls
03:37.42coppiceits filtered in most situations
03:38.13dlynesSo it's not allowed for domestic traffic, and international inbound traffic?
03:38.32dlynesSomeone I know at Shanghai Bell Alcatel was telling me it was
03:38.50coppicethey don't really need to ban it. they've just filtered every protocol you can think of :-)
03:39.35dlynesMake it go through on a different port then :)
03:39.43dlynesSkype works just fine in China
03:40.14dlynesIt actually sounds quite clear coming from China, too
03:41.21coppicebroadband in china is generally excellent, so the connections should be pretty clean, unless the international part gets congested
03:42.04coppicethe free broadband in my nanjing hotel room this week let me download something at about 1MB/second
03:43.09coppicebut they'd filtered UDP to the point where VPN wouldn't work, and I needed to use dialup :-(
03:43.49*** join/#asterisk rogercharlie (n=adrian@c-69-181-20-122.hsd1.ca.comcast.net)
03:44.46rogercharlieanyway to change group count with a sip transfer?
03:46.08rogercharlieupdates the correct channel with an asterisk transfer
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03:47.38xbmodder_lappyhow does E911 work?
03:51.02dlynesxbmodder_lappy: If I remember correctly, your PSTN/PRI upstream service provider provides that information on the ring delivery to the 911 call center
03:51.46xbmodder_lappydlynes, Yeah, I understand that part of it, but I would like to know how they send it to the 911 call center...
03:51.47dlynesxbmodder_lappy: Of course, when you get that line/pri, you have to provide your SP with all that information for every new subscriber that needs E911 capability
03:52.13dlynesxbmodder_lappy: It's sent on the wire similarly to the caller id info
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03:53.58dlynesman....i need to do something about this beer...been sitting here since Christmas, and I've only drank two of them :(
03:57.22xbmodder_lappyi can help
03:57.34rogercharlieemail me the beer
03:59.02coppiceI wonder if any 911 call centres have been outsourced to bangalore :-)
03:59.34Qwellcoppice: no doubt
04:00.26coppiceit would be one of those homeland insecurity things :-)
04:01.15xbmodder_lappylol
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04:16.11dlynesemail beer...hrm....that would be....kinda difficult....mmmmkay?
04:18.01coppiceBoIP
04:18.23coppiceor the more generic AoIP
04:19.47xbmodder_lappya?
04:19.52xbmodder_lappyMoIP
04:19.57coppicealcohol
04:20.03xbmodder_lappymatter over IP
04:20.28coppicerather than usual immaterial over IP?
04:25.33kamileonanyone in here interested in a used tdm40b digium card?
04:28.13xbmodder_lappyhow much?
04:28.33kamileon250
04:28.51kamileonall check out 100.0%
04:29.32xbmodder_lappywhy are you selling it?
04:33.41rogercharliehe beat up a geek for it
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04:48.17Qwellkamileon: Sell it to the guy who posts every day on asterisk-biz
04:54.09kamileonim  not familiar with that
04:55.02dlyneskamileon: it's a mailing list
04:55.03kamileonxbmodder_lappy: i need to purchase other toys to p;lay with
04:55.03kamileonand someone is posting trying to buy gear?
04:55.03kamileonsign up at asterisk.org or digium.com?
04:55.40dlyneskamileon: asterisk.org
04:55.50*** join/#asterisk shido6 (n=shido6@d38-45-81.commercial1.cgocable.net)
04:56.02dlyneskamileon: people post on asterisk-biz regularly selling routes, buying routes, selling hardware, buying hardware, ...
04:57.40rogercharlieis OUTBOUND_GROUP able to be updated with a SIP transfer?
04:58.12rogercharlieit no workey even with __OUTBOUND_GROUP
05:00.09kamileonoh cool
05:00.48dlyneskamileon: Yeah...you should sign up on there...the crew on voipsupply regularly blows out specials on there
05:01.24Qwellkamileon: lists.digium.com
05:02.05Qwellkamileon: Just send all messages with "Rehan", "didx", or "used digium hardware" to the bit bucket :p
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05:02.20dlyneslol
05:02.29Qwellwell...the first two, keep those if you like to laugh
05:02.45dlynesor like to get annoyed :)
05:02.47QwellYou'll get some classic quotes, such as hungry dids, or tool free dids
05:03.01Qwellhaha, I just got tool free, also
05:03.18kamileonlol
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05:22.30drfoomod2for a ta750, is this all that is needed in the zaptel.conf file?
05:22.30drfoomod2span=1,0,0,esf,b8zs
05:22.30drfoomod2fxoks=1-4
05:22.30drfoomod2fxsks=5-8
05:28.03*** join/#asterisk m_a_g_o (i=maxgluck@201.243.97.246)
05:29.34m_a_g_ogood evening guys... I'm installing DBD::mysql in order to run AGI scripts and I'm getting: Can't locate loadable object for module DBD::mysql in @INC
05:29.44m_a_g_oany idea where I can modify @INC?
05:30.31dlynesm_a_g_o: it's probably looking for a mysql client library...do you have the mysql client libraries installed?
05:32.23m_a_g_oI downloaded and compiled DBD-mysql-3.0002 with the option --mysql_config= and the path...
05:32.43m_a_g_obut I can see there are perl versions installed from 5.8.5 to 5.8,7
05:32.44coppiceanyone used the rhino T1 card? it seems to have been around for a while, but I never heard from anyone using it
05:32.52dlynesm_a_g_o: but do you have the mysql client libraries installed?  i.e. libmysql.so
05:33.38dlynesm_a_g_o: erm libmysqlclient.so
05:33.42m_a_g_ofrom asterisk-addons?
05:33.50dlynesm_a_g_o: From your linux distribution
05:34.24dlynesm_a_g_o: It would help to know which distribution you're using; then someone can let you know what package you need to install
05:34.36m_a_g_o'/usr/lib/libmysqlclient.so.10
05:34.47dlynesm_a_g_o: there ya go...is it there?
05:34.48m_a_g_oFC3
05:35.11*** join/#asterisk tzafrir_laptop (n=tzafrir@80.178.4.164.adsl.012.net.il)
05:35.27m_a_g_othere and in /usr/lib/mysql/libmysqlclient.so.10
05:35.40dlynesm_a_g_o: Have you tried asking on #perl?
05:36.09dlynesm_a_g_o: I just did perl -MCPAN -e shell, and the install DBD::Pg for Postgres and it works just fine
05:36.34dlynesI'm using perl 5.8.7
05:40.15m_a_g_onothing... tried compiling with --mysql_config=/usr/local/mysql/bin/mysql_config  --cflags=-I/usr/local/mysql/include/mysql/mysql.h, also from CPAN and got some errors...
05:40.47dlynesm_a_g_o: like i said...did you try asking on #perl?
05:41.47m_a_g_oasking right now
05:43.22drfoomod2should a autogenerated zaptel.com have at least one span= in it?
05:43.26drfoomod2mine does not
05:44.00dlynesyou mean zaptel.conf?
05:44.12drfoomod2yes
05:44.25dlynesonly thing it usually has uncommented in it is loadzone and defaultzone
05:44.32drfoomod2right
05:44.53drfoomod2i added a span=1,1,0,esf,b8zs
05:44.59dlynesthe span= line will be dependant upon what you've got it hooked up to
05:45.01drfoomod2and a fxsks=1-4
05:45.08drfoomod2and fxoks=5-8
05:45.16drfoomod2and i get an error when i try ztcfg
05:46.03dlynesYou probably have your fxsks/fxoks backwards then
05:46.20dlynesYou've got a tdm04b and a tdm40b installed?
05:46.24drfoomod2i get invalid argument
05:46.52drfoomod2te110p
05:47.00dlynesand the span= afaik is for the te400p/te110p/te410p cards
05:47.18dlynesOk, so why are you trying to define fxsks/fxoks?  Isn't the te110p a t1 card?
05:47.24drfoomod2it is
05:47.35drfoomod2and it;s connect to a ta750 channel bank
05:47.37dlynesYou don't define fxoks/fxsks on a t1 card
05:47.44drfoomod2oh?
05:47.49dlynesno
05:48.14drfoomod2drop those lines?
05:48.28dlynesspan=, bchan, dchan, loadzone, and defaultzone for a t1 card
05:48.34dlynesyes
05:48.34drfoomod2i did and i still get invalid argument
05:48.46drfoomod2may i flood?
05:48.48dlynesbecause you haven't specified bchan, dchan more than likely
05:49.15dlynesbchan is 1-23 and dchan is 24 for most north american t1's
05:49.29drfoomod2this is not a PRI
05:49.34coppicedlynes: he's using a channel bank, not ISDN. he doesn't have any B channels or D channels
05:49.54dlynescoppice: ah...thought you had to specify all that stuff for channel banks, too
05:50.02drfoomod2coppice: tx
05:50.23dlynesdrfoomod2: but regardless, the fxoks/fxsks aren't appropriate for a t1 card
05:50.26drfoomod2the ztcfg -v shows span 1, 8 channels configured, and then barfs out a invalid argument
05:50.47drfoomod2dlynes: yes they are
05:50.50kamileoncat tums > asterisk
05:51.21drfoomod2ns
05:51.46drfoomod2and what is ZTDUMMY/1?
05:52.05dlynesdrfoomod2: a timing interface that relies on your rtc kernel driver
05:53.43drfoomod2reboot
05:53.47drfoomod2*
05:56.00opus_anyone here use asterisk with large amount of simutaneous calls?
05:56.07opus_like, more then 40 at once?
05:56.21Qwellopus_: yes, many people, I'm sure
05:56.45opus_Qwell do you?
05:57.00QwellTo save time, let's say "sure"
05:57.18coppiceopus_ 40 would be a rather small number of calls
05:57.48opus_well, how do people stop asterisk from locking up? for example if it was in pure SIP mode
05:57.53opus_eventually I get a locked up *
05:58.12drfoomod2how can i tell what * thinks i have for a t1 card?
05:58.13Qwellopus_: What type of hardware, and how much transcoding and such?
05:58.31coppiceopus_: do you cross your fingers when you start asterisk?
05:58.35opus_g711, vmware, and sipp
05:58.39drfoomod2oh boy
05:58.43drfoomod2i have to go to sleep
05:58.50Qwellopus_: That's why.
05:58.57drfoomod2i just remembered i tool the digium card out of this box
05:58.59Qwellasterisk requires a realtime environment
05:59.02Qwellvmware simply cannot provide that
05:59.03opus_vmware. yeah yeah
05:59.04dlynesdrfoomod2: dmesg
05:59.09drfoomod2and put an old ethernet card back in its place
05:59.17coppicedrfoomod2: what does /proc/zaptel/<whatever> say you have?
05:59.25opus_Qwell, ok. so if I run it in a realtime environment it will be OK
05:59.27opus_?
05:59.41drfoomod2coppice: it aint; gunna say shit
05:59.43opus_nothing in my dial plan requires realtime stuff btw.
05:59.47Qwellopus_: Yes.  Don't run it in any type of vm
05:59.48opus_answer, play wave file, hangup.
05:59.52Qwellrealtime != real time
05:59.58opus_yeha
06:00.02QwellI meant real time
06:00.04opus_i mean zaptel hardware
06:00.06drfoomod2coppice: the card is in another machine on the other side of the room
06:00.13drfoomod2it';s later
06:00.14tainted-bengay != ben gay
06:00.15drfoomod2g'nite
06:00.37Qwelltainted-: helping your uncle jack..nevermind
06:00.49tainted-Qwell i got my box up to ~350 calls
06:00.49opus_i'm coming to the belief that anyone who saids they have huge amounts of simutaneous calls setup with asterisk is lying
06:01.00Qwelltainted-: I got my box up to ~2500 calls :p
06:01.03opus_tainted: with RTP audio?
06:01.06tainted-did u ever get sipp to push media through?
06:01.07rogercharlieanway to update destination of OUTBOUND_GROUP with a SIP transfer?
06:01.08Qwelland that's with rtp
06:01.11Qwelltainted-: yeah
06:01.14tainted-what no way
06:01.20Qwellyep
06:01.27Qwell2500 without any retransmits
06:01.29opus_Qwell: you have a screen shot with 'show channels'
06:01.31opus_?
06:01.35tainted-what kind of sipp scenario
06:01.40coppiceits amazing how many claim they pass RTP through when they don't
06:01.50Qwelltainted-: -sf uac_pcap.xml
06:01.53opus_coppice: exactly
06:01.53tainted-opus_ cli is pretty useless at that point
06:01.57Qwellcoppice: I assure you, I was
06:02.16opus_Qwell were you doing RTP 100% of the time or just once for a few seconds..
06:02.17tainted-i gave up on sipp and started loading the box up with call files
06:02.32Qwellopus_: 100% of the time
06:02.36coppiceIn the early days of T.38 investigation I was assured by lots of people T.38 was passing through their * box. of course, it wasn't :-)
06:02.56opus_169 active channels
06:02.56opus_86 active calls
06:03.14opus_that will run OK for about half an hour then it will bomb .
06:03.21coppice2500*50 packets a second*2 directions == a lot of rescheduling of threads
06:03.22tainted-really?
06:03.34tainted-hmm
06:03.52Qwellcoppice: two words.  ultrasparc T1
06:03.54tainted-is it accurate when i use call files and the local/ channel
06:04.13opus_tainted : no :(
06:04.17tainted-i just drop it in a looping background context
06:04.31opus_tainted: because its all inmemory
06:04.32tainted-how do i accurately load test it then
06:04.38opus_if you had two servers, it would be valid
06:04.39coppiceQwell: so its pure IP? no PSTN? why do you want the audio to pass through?
06:04.52tainted-opc0de i have two servers
06:04.52opus_then the test would go through the network statck and be more legit
06:04.57tainted-opus_
06:04.59Qwellwhy do I want it to pass through?
06:05.05*** join/#asterisk Shaun222 (n=ndci@ip68-5-63-223.oc.oc.cox.net)
06:05.21opus_Qwell: so.. you have a server that does 2500 simutaneous calls at once?
06:05.28Qwellopus_: 2500 channels, yes
06:05.42Qwellwith rtp flowing both directions
06:05.55tainted-opus_ so what if i used call file and connected two servers to each other
06:05.59opus_is there anyway you could give me a screen shot..  or something?
06:06.05Qwellopus_: tomorrow
06:06.07opus_tainted: I would say that would be valid
06:06.23tainted-why would the network stack involvement be more legit
06:06.23dlynesQwell: I'm guessing you're running modified asterisk code, if it's running on a SPARC?
06:06.23opus_Qwell: cool
06:06.37Qwelldlynes: only slightly modified, and really only the Makefile
06:06.54dlynesQwell: ah...but no hardware, right?  pure software?
06:07.07Qwelldlynes: right
06:07.27dlynesQwell: And you're using the Sun compiler, or the GNU compiler?
06:07.34Qwellgcc
06:07.39dlynesah
06:07.46Qwellgmake, ginstall, etc
06:07.52dlynesSun produces more optimized code though, doesn't it?
06:08.04Qwellyeah, but it won't compile asterisk.  it would take a lot more work
06:08.08dlynesah
06:08.09opus_one thing about the SIP stack, I don't think its fast enough to create/destory so many call stacks, like it needs to be refactored/nmore multithreaded
06:08.23Qwellopus_: I was getting 100+ cps
06:08.31dlynesQwell: So you're not taking advantage of Solaris threads then?
06:08.39Qwelldlynes: oh, I most certainly am :p
06:08.41opus_Qwell: i can't get over *4* cps
06:08.50Qwellopus_: dude, my amd64 can get 25
06:09.09Qwellstop trying to run it in vmware
06:09.25opus_636 active channels
06:09.25opus_342 active calls
06:09.46opus_Now the system is showing about 1 page of warning: per second
06:09.47QwellThat's without rtp, I'm sure
06:09.51*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
06:09.55dlynesQwell: How do you think an Ultra2 Creator dual cpu would handle it, then?
06:09.57opus_no this is with rtp
06:10.01Qwelldlynes: dunno
06:10.39tainted-<Qwell> stop trying to run it in vmware <--- lol
06:10.43opus_<PROTECTED>
06:10.50Qwell34k/s?  rofl
06:10.56tainted-opus_ stop trying to connected to your access farm dude
06:11.08Qwellopus_: That is NOT with media
06:11.08tainted-s/connected/connect it
06:11.11opus_wtf
06:11.21QwellThat's like...not even half a ulaw channel
06:11.47opus_sipp is buggy
06:11.56Shaun222i need the number to a international pizza hut or somthing to test int calls, lol
06:12.14QwellShaun222: bkw__ called Hilton hotels around the world
06:12.17QwellI think it was Hilton
06:12.17opus_Well anyway asterisk crashed.. just after 350 calls
06:12.46Shaun222Qwell: not a bad idea..
06:12.49Shaun222gotta find the number...
06:14.02tainted-Qwell where'd u get uac_pcap.xml
06:14.18Qwelltainted-: I think that was the filename.  I got it from the sipp site...it's in the faq
06:14.22tainted-u created it?
06:14.23tainted-oh
06:14.30opus_tainted: sipp automatically generates it when you run ./sipp -sd uac_pcap
06:14.34Qwellno
06:14.38opus_tainted: rtfm :)
06:14.39tainted-really?
06:14.39Qwellhttp://sipp.sourceforge.net/doc1.1/reference.html#PCAP+Play
06:14.41tainted-k
06:14.59Qwellhttp://sipp.sourceforge.net/doc1.1/reference.html#uac_with_media
06:15.02Qwellthose two
06:15.02tainted-i need to get a better handle on sip
06:15.12tainted-opus_ stop loading testing against your desk phone
06:15.23tainted-lol
06:15.43opus_i'm stress testing it
06:15.52tainted-i get seg fault
06:16.00tainted-when i do the -sd uac_pcap
06:16.01opus_welcome to the club
06:16.14tainted-hmm
06:16.31opus_tainted: hint, its also in the .tar.gz file from the download site
06:16.49tainted-did u guys 'make pcapplay'?
06:16.58tainted-or just 'make'
06:17.02opus_yeah with oss
06:17.21Qwellmake pcapplay, yeah
06:17.29Qwellwhich unfortunately didn't work on solaris...
06:17.38Qwellso I had to generate my calls from my amd64, and send them over the wire
06:18.26opus_qwell, what timing device is used in solaris? h is that a problem you are going to need to tackle?
06:18.26Qwelliirc, it was about 200mbit/s?
06:18.46Qwell176mbit/s, I guess
06:18.54Qwellwait, wrong number
06:18.59distortionqwell you arent human
06:19.01Qwellyeah, just at 200
06:19.03tainted-opus_ i'm positive it wasn't in tarball
06:19.08Qwelldistortion: yes, we've established this
06:19.12tainted-but gonna try the one in faq
06:19.17*** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com)
06:19.20Qwelltainted-: it isn't. :p
06:19.22distortionjust making sure..
06:19.29Qwellthe .pcap file is there though
06:19.33websaegood evening all, how fair you?
06:19.50websaeanyone have any good provisioning concepts?
06:20.06distortionwebsae you nerd
06:20.12dlynesopus_: not if you dont' need music on hold or conferencing
06:20.13Shaun222any reason i couldnt use dial() to ring/find a agent with out using queue() ?
06:20.25websaedistortion: how are you doing?
06:21.02distortiongood, had a big day, almost 750k mou today mmm
06:22.25rogercharlieanyway to change OUTBOUND_GROUP to the SIP transfered username?
06:25.55*** join/#asterisk gmaruzz (n=maruzz@217-133-80-112.b2b.tiscali.it)
06:26.16*** join/#asterisk Abydos313 (i=abydos31@adsl-71-129-60-235.dsl.irvnca.pacbell.net)
06:26.48tainted-Qwell do u put actual values in the xml [keyword] or leave them?
06:26.59opus_tainted, you have to create a csv file
06:27.03Qwelltainted-: huh?
06:27.05tainted-INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
06:27.15Qwelltainted-: no, it fills them in.  Leave it alone :p
06:27.16tainted-should i leave that alone or put in actual values
06:27.24Qwell./sipp someargs -sf uax_pcap.xml
06:27.26tainted-<PROTECTED>
06:27.40Qwelltainted-: That's with `make pcapplay`?
06:27.47tainted-yea
06:27.51Qwellfun
06:28.02tainted-http://pastebin.ca/50663
06:28.10tainted-i just took that out of the faq uac section
06:28.32Qwelllooks different than the one I had
06:29.04opus_tainted you need to root when you run it, because on some distros the pthread method it uses for scheduling is blocked for security reasons
06:29.06tainted-looks like a generic invite, response 200, ack, then bye
06:29.15tainted-yea i'm root
06:29.29opus_ok, then, its broken for sure:)
06:29.42Qwellahh, heh
06:29.45Qwellhttp://sipp.sourceforge.net/doc1.1/uac_pcap.xml.html
06:29.49QwellGet that, and save it
06:30.05*** join/#asterisk {tasker-} (n=ghes@modemcable252.110-83-70.mc.videotron.ca)
06:30.06tainted-oh look at that
06:30.21QwellYou need the uac_pcap one, not uac.xml :p
06:30.28Qwellhttp://sipp.sourceforge.net/doc1.1/uac.xml.html  Thats what you got
06:30.52{tasker-}anyone have success getting chan_h323.to load in svn trunk?
06:30.55tainted-hmm.. but the xml should still be valid
06:30.56tainted-weird
06:31.01Qwelltainted-: dunno
06:31.49*** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk)
06:32.00{tasker-}all I get is this when I load asterisk -> loader.c:731 __load_resource: missing mod_data for chan_h323.so
06:32.13{tasker-}trunk
06:32.44Qwellis chan_h232 in the asterisk repository?
06:32.58QwellIt's in -addons, isn't it?
06:33.06{tasker-}no
06:33.09distortionno, its in channels/h323
06:33.20distortionooh323 is in addons
06:33.20{tasker-}yup
06:33.24Qwellwell, wait for the loader changes to calm down
06:33.30opus_tasker, grep for mod_data in the asterisk sources and make sure whatever module proves mod_data loaded before chan_h323.so in modules.conf? Or maybe its because in SVN trunk right now loader.c is being drammatically changed
06:33.32QwellThings that aren't in the base will not work
06:33.40opus_tasker: what Qwell said
06:33.46QwellIf you run svn trunk, you MUST subscribe to the -dev list
06:33.57Qwellnot doing so is pretty stupid...
06:34.19{tasker-}crap
06:35.00{tasker-}the 1.2.7.1 version locks up after 15 minutes
06:35.04{tasker-}forcing a restart
06:35.07opus_tasker you can probably create a mod_data structure .. or splice in the new module requirements into the top of chan_h323
06:35.11{tasker-}and ooh323 is another useless piece of garbage
06:35.19{tasker-}under heavy load it stops answering calls after 1 hr
06:35.29{tasker-}issuing a debug shows nothing
06:35.43{tasker-}i can telnet into port 1720 of chan_ooh323.so and it does nothing
06:35.49{tasker-}when it's dead
06:36.11{tasker-}opus_: i suppose i can try that
06:36.24opus_tasker: crash? really, I had some code that no longer works in 1.2.7.1 , it was .. lemme look
06:36.45{tasker-}chan_h323.so has a bad deadlock problem
06:36.58{tasker-}chan_ooh323 has another problem that I can't seem to trace
06:37.03{tasker-}no locks, it just stops
06:37.22{tasker-}they keep closing the channel deadlock issues on bug tracker
06:37.35{tasker-}without resolving anything
06:38.01Qwell{tasker-}: When bug marshals close bugs, they give a reason.  What is the reason they give?
06:38.09{tasker-}and I'm getting a headache trying to trace through all the mutex locks & unlocks in JerJer's chan_h323.c
06:38.19{tasker-}fixed
06:38.22distortiontasker- i know your pain
06:38.38distortionI've had to resort to a load balancer and 3 servers to get ooh323 to work
06:38.39{tasker-}Qwell: they always report it fixed
06:38.40Qwellpfft, nobody knows the pain of dealing with JerJer's code more than I do :P
06:38.47{tasker-}lol
06:38.59distortionone server goes down and then is taken out of hunt to be reset
06:39.02Qwell(in 6 months, somebody will say that about my code)
06:39.07opus_distortion HAHA that was going to be my next step :)
06:39.18distortioneven a restart of asterisk doesnt fix it, i have to reboot the faking server
06:39.22{tasker-}distortion: what do you use to load balance
06:39.23Qwellh323 sucks anyhow
06:39.32distortionh323 is unavoidable
06:39.37{tasker-}i agree
06:39.43Qwellno it's not :p
06:39.48{tasker-}the majority of the world's carrier minutes still run through h323
06:40.02QwellAre you trying to run a carrier?
06:40.08{tasker-}not trying
06:40.12distortionI have a lame proprietary app that does the load balancing now, the very app i was hoping asterisk(h323) would help me get rid of
06:40.15{tasker-}we have over 250 of them on our network
06:40.35{tasker-}distortion: sigh :(
06:40.36Qwellh323 was irrelevent by the time the spec was finished
06:40.48distortionI have about 45 active h323 carriers in our network that "cant" do sip
06:40.58{tasker-}Qwell: whatever your opinion, the carriers adopted it first and it's very widespread
06:41.04{tasker-}not in the retail space
06:41.08{tasker-}SIP dominates there
06:41.10Qwellit was designed by telecom guys...so they used telecom issues
06:41.15distortionyeah, retail == sip
06:41.16{tasker-}but the retail space is nothing compared to carrier space
06:41.36*** join/#asterisk hads|home (n=hads@203.109.245.87)
06:41.37{tasker-}H.323 was designed to mimick SS7 in some respects with very poor planning
06:41.55distortionwell, i disagree, it has q931 which is quite nice
06:42.00*** join/#asterisk [Airwolf] (n=airwolf@83.98.235.219)
06:42.01{tasker-}yes
06:42.09distortionalthough h225 sucks balls
06:42.21{tasker-}and that's one of the driving factors keeping H.323 alive with carriers
06:42.28{tasker-}q931
06:42.35QwellWhat is q931?
06:43.06distortionits isdn signaling in voip (in lamens terms)
06:43.11Qwellwhy?
06:43.33opus_its like SS7 for pbx systems that runs on the D channel from what I understand (?)
06:43.40{tasker-}it can help in diagnosing an endpoint's TDM problems
06:43.42{tasker-}for example
06:43.47{tasker-}SIP -> 404 not found
06:43.50{tasker-}can mean a dozen things
06:43.57{tasker-}q931 is very specific
06:44.27{tasker-}you get a cause code reflecting the actual error (to some degree, although some networks have gotten very loosey-goosey about that)
06:44.29Qwellbesides that the requested resource couldn't be found, what else does 404 mean?
06:44.47distortionnot only specific, but more importantly very relavent to TDM (isdn/ss7) messages
06:44.47*** join/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org)
06:44.48{tasker-}hang on, i had a translation list handy
06:45.01{tasker-}some SIP errors translate into 3-4 q931 errors
06:45.27dlynesQwell:  I've got one sip error I get back from my upstream that can mean ten different things
06:45.37Qwelldlynes: which?
06:45.40{tasker-}oh
06:45.52dlynesQwell: They don't feel like returning specific errors, because most of the switching eq doesn't understand it
06:45.53{tasker-}i know that pain all too well
06:46.17QwellThey probably aren't following the SIP spec either :p
06:46.23opus_tasker are you running h323 and q931?
06:46.38distortionhttp://pastebin.com/676427
06:46.52distortionthats the sip -> q931 mapping from my lucent gateway
06:47.04dlynesQwell: Can't remember offhand...don't have any of those errors in my recent logs
06:47.22{tasker-}yes
06:47.27{tasker-}opus_: yes
06:48.06{tasker-}good lord
06:48.14{tasker-}i forgot how stupid error 503 was
06:48.27distortiongrr 503
06:48.34distortion503 can mean anything
06:48.51{tasker-}"no circuit available" is a far cry from "network out of order"
06:48.58*** join/#asterisk cian (n=cian@g5.cian.ws)
06:49.12{tasker-}does anyone remember the SIP -bis extension?
06:49.19dlynesYeah...no circuit available...that's the stupid error I get once in a while
06:49.26dlynesIt can mean many different things
06:49.29{tasker-}it was meant to integrate some ISDN idioms into SIP
06:49.47{tasker-}i.e. q931 cause codes
06:49.57dlynesIt's error 403 or something is what my provider has it mapped into
06:49.58{tasker-}somewhere along the lines it was dropped
06:50.01*** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net)
06:50.11distortionwell i think the new thing is sigtran which i know little about but am very interested in
06:50.16dlynesAlmost everything they have mapped into 403
06:50.23{tasker-}ugh
06:50.51tainted-Qwell can u pastebin your uac_pcap.xml file? the one u pointed out generates 'No 'scenario' section in xml scenario file.' even though it validates, and ./sipp -sd uac_pcap segfaults
06:50.53firestrmanyone here good at wired iax problems?
06:50.54dlynesSo whenever I have a problem with a route, I get on the msn blower to them and bitch
06:51.12tainted-firestrm what's the problem
06:51.39distortiontasker- you use * in your wholesale environment?
06:51.49firestrmim using sixtel (ya dont laugh) for long distance termination, i can make calls to cellular numbers, but not landlines..
06:51.56tainted-the thing about asterisk is -- even when u get a crazy sip response, u can't even access it from AGI
06:52.15Qwellfirestrm: That's on their side, no doubt
06:52.29tainted-firestrm sixtel is run by a joker
06:52.39{tasker-}no, just for some anciliary services
06:52.39firestrmthats what im afrade of... :( that means it will NEVER get fixed..
06:52.40dlynes{tasker-}: another thing that really sucks about asterisk for connecting to upstreams is the lack of a g723 codec transcoder
06:52.57dlynes{tasker-}: a lot of the same guys that do h323 also seem to do g723
06:53.01{tasker-}dlynes: allow me to fix that issue for you right now
06:53.02{tasker-}sec
06:53.18dlynes{tasker-}: ?
06:53.22tainted-dlynes yea i agree on that!! especially for intl termination
06:53.38firestrmtainted-, ya i know.. joker is one of the nicer words i would use for him, but he is one of the few that offer 250 and canadian 1-800 did's
06:53.56dlynestainted-: 75% of my available north american termination uses g723
06:54.04tainted-firestrm that's not termination though, that's origination
06:54.09dlynestainted-: I have to ask my terminator to force g729
06:54.15tainted-firestrm why don't u terminate through someone else
06:54.40firestrmive tried, cant find anyone who gives as good quality connection as sixtel
06:54.51{tasker-}http://kvin.lv/pub/Linux/Asterisk/
06:54.53dlynesfirestrm: Group Telecom offers 1-800 dids
06:54.54{tasker-}have fun
06:55.01{tasker-}g.723.1 and g.729a
06:55.32firestrmdlynes, any good? as in , choppy audio, and delays like your on the moon?
06:55.49Qwell{tasker-}: Are you in the US?
06:55.55{tasker-}canada
06:56.14distortiontasker, have good routes to canada? :)
06:56.16Qwellmmhmm
06:56.20distortion<-- has minutes
06:56.36{tasker-}what's your volume?
06:56.46{tasker-}:)
06:56.48distortion950k/mo
06:56.53distortion~4t1's
06:57.01{tasker-}small volume
06:57.26Qwell{tasker-}: That violates both Intel copyright, and the codec patents...
06:57.26dlynesfirestrm: They offer them on their pri's and landlines...so you'd have to find someone that can voip them for you
06:57.34{tasker-}Qwell: it violates nothing
06:57.45dlynesfirestrm: try www.calltermination.com...there's probably someone on there that can do it for you
06:57.52{tasker-}Qwell: Intel provides the code base free on their website to developers
06:57.57{tasker-}Qwell: and for internal use
06:57.58QwellWRONG
06:58.01{tasker-}nope
06:58.05{tasker-}go read their site
06:58.08{tasker-}I did
06:58.15{tasker-}already consulted a lawyer about it, too :)
06:58.17QwellThey provide it for RESEARCH ONLY, or you can get a license for non-commercial use
06:58.18opus_Qwell: I think taske ris right
06:58.39opus_from reading their site i came to the same conclusion
06:58.57QwellYou *MUST* get a license if you intend to USE the code, and you STILL have to pay the patent royalties
06:59.19Qwellthe README file on the same damn link says the same
06:59.24{tasker-}sure
06:59.34distortiontasker: 1mm/mo customer may be small but they are the best kind ;)
06:59.34{tasker-}if you plan to use it commercially
06:59.53{tasker-}distortion: i agree
06:59.56Qwell{tasker-}: Go read the license at intel.com
07:00.00{tasker-}don't have to
07:00.09{tasker-}i already did that whole exercise
07:00.13distortionbut my canada rates are dog cheap, i doubt you could touch em
07:00.19{tasker-}ITU's patent on g.729 is almost up
07:00.24tainted-woah are those 723/729 binaries?
07:00.32{tasker-}distortion: try me
07:00.39{tasker-}tainted: yup
07:00.47tainted-woah
07:01.08dlynesdistortion: I'm currently getting $0.08/mi for Canada
07:01.23Qwellhttp://www.intel.com/cd/software/products/asmo-na/eng/perflib/ipp/219689.htm
07:01.32{tasker-}Qwell: dude, it's cool
07:01.42{tasker-}Qwell: I already did that whole exercise a year ago with the lawyers
07:01.49Qwellwell, your lawyers were wrong.
07:01.53tainted-{tasker-} what kind of volume do you push?
07:01.54QwellRTFEULA
07:01.56{tasker-}Qwell: I'm not violating anything the way we're set up
07:02.04{tasker-}Qwell: take a chill pill, dude
07:02.04Qwell{tasker-}: Do you have a license to use the code?
07:02.19tainted-Qwell well if he's doing pass-through it's legal eagle
07:02.29Qwelltainted-: and he doesn't need binaries
07:02.30{tasker-}tainted: exactly :)
07:02.49Qwellasterisk supports g723 and g729 passthrough
07:03.00dlynesQwell: Just not very well
07:03.03{tasker-}tainted: except in the case where we play back wav files, but encoding prompts in g.729 and g.723 fixes that, too
07:03.06tainted-maybe just for the stray ulaw channel?
07:03.09Qwelldlynes: What "well"?
07:03.14dlynesQwell: because the codec autonegotiation blows in asterisk
07:03.20Qwelldlynes: It's passthrough.  It's 100% transparent
07:03.38tainted-{tasker-} what kind of volume do you do?
07:03.41{tasker-}dynes: agreed
07:03.46dlynesQwell: Yeah...the passthrough works just fine, but the codec autonegotiation to make it all come together blows
07:03.49{tasker-}to Canada
07:03.50{tasker-}?
07:03.54{tasker-}or in total?
07:03.55tainted-anywhere
07:03.57tainted-total
07:04.05{tasker-}about 4 million / day
07:04.08dlynes{tasker-}: pardon?
07:04.12distortionnice.
07:04.14tainted-woah
07:04.19tainted-that's small volume
07:04.32distortionwe did about 750k today.. 500 on asterisk :DDD
07:04.32tainted-i do that between my polycom and next door
07:04.42tainted-distortion over how many boxes
07:04.47distortion6.
07:04.53tainted-damn some heavy hitters in here tonight
07:04.59opus_distortion, do they dead lock and lock up?
07:05.14opus_how many simutanous calls can your asterisk box get?
07:05.19distortionh323 boxes, yes- sip boxes on 1.2.6 == rock solid
07:05.26*** join/#asterisk konfuzed (n=Konf@H211.C18.B96.tor.eicat.ca)
07:05.32tainted-distortion are u pure voip?
07:05.33{tasker-}lol
07:05.59distortionno, we have cisco/lucent gateways that pipe minutes into *
07:06.13kamileonwhere can i get a 1800 did ?
07:06.35opus_distortion, really.. do you do anything in the dial plan or do you just route the calls around?
07:06.50distortionin response to taskers 'lol' asterisk has its issues but sip only the service is stable
07:07.01dlynesdood...rehan walla wallah agar has like tonnes of 1-800 dids
07:07.12tainted-lol
07:07.14{tasker-}distortion: i agree
07:07.18kamileonwtf
07:07.19tainted-rehan wallawalla?
07:07.35distortionwe use asterisk for least cost routing through mysql. not much dial plan work
07:07.36dlynestainted-: hehe...that clown in asterisk-biz :)
07:07.49dlynestainted-: the one that's reselling other peeps dids
07:07.50{tasker-}distortion: my next try is to use a SIP/H.323 gateway between asterisk and the rest of the H.323 world
07:07.56{tasker-}has anyone looked at YATE?
07:07.57opus_distortion whats the most one of the asterisk box can take?
07:08.02distortionyate is good
07:08.12tainted-distortion how are u balancing load
07:08.17opus_distortion in numjber of simutaneous calls
07:08.30tainted-opus_ send me your uac_pcap.xml
07:08.31kamileonwhy when i switch to g729 over ulaw people tell me my calls sound worse
07:08.33websaedistortion: i told tasker you would be talking to him about our rates
07:08.35distortiontoday, about 800 between 6 servers
07:08.40tainted-opus_ my sipp segfaults on -sd
07:09.04opus_distortion: 200 calls on each server?
07:09.06distortionload balancing on sip side by ser
07:09.16tainted-thought so
07:09.21*** part/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net)
07:09.29{tasker-}i'm planning to use YATE as a proxy between asterisk and the h.323 providers
07:09.38{tasker-}if i could get the damn thing to compile
07:09.39tainted-i've yet to see a non ser involved * load balance solution
07:09.50opus_tainted whats your email
07:10.00distortionultramonkey anyone?
07:10.12distortioni purchased a f5 load balancer
07:10.17distortionfaking thing worked amazing
07:10.28distortionbut then it would drop sip sessions after 10 minutes
07:10.29opus_distortion when you load balance to the box do you get 200 calls per server? is your load average on the server pretty low?
07:10.35distortion*sigh*
07:11.33{tasker-}sucks
07:11.38distortionso far i've seen about 320 calls max on a single server (g729 passthrough), cpu was at about 20% on a dual zeon 2.8 dell
07:11.53{tasker-}on asterisk?
07:12.32{tasker-}i've brought it up as high as 1600 channels on passthrough, dual xeon 3.2 800 fsb
07:12.51distortiontahts fkn pimp
07:12.53*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
07:13.03{tasker-}it's not ideal, though
07:13.05opus_distortion: ah ha. what happeneds if you go over that amount?
07:13.15{tasker-}max 1200 channels to keep it cruising
07:13.34opus_distorition, or.. how often does your asterisk box go down?
07:13.51{tasker-}h.323 -> every hour
07:13.54{tasker-}SIP - never
07:14.00*** part/#asterisk rogercharlie (n=adrian@c-69-181-20-122.hsd1.ca.comcast.net)
07:14.01distortionagreed
07:14.03distortionsip never
07:14.20*** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com)
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07:14.33opus_crap.. you guys have it easy, all you are doing is routing calls:)
07:14.40opus_(you could do that in SER!) :)
07:14.43distortion...
07:15.22{tasker-}SER can proxy media or pass it to the other end, right?
07:15.29distortionyes.
07:15.33{tasker-}allow two endpoints to connect media channels
07:15.38{tasker-}just like GnuGK
07:15.53distortionits much better than * on the media direct connect, they call it "record route"
07:16.02{tasker-}on both GnuGK and SER, we've never been able to hit as of yet
07:16.06opus_coppice, are you still around
07:16.11distortionasterisk fucks itself on billing if it doesnt proxy media
07:16.11{tasker-}to hit a limit as of yet
07:16.15{tasker-}i missed a word :(
07:16.43{tasker-}GnuGK and SER can easily take several DS3s
07:16.59{tasker-}the problem is connecting SER and GnuGK clients together :(
07:17.05{tasker-}but GnuGK also never goes down
07:17.19opus_GnuGK ?? never heard of it
07:17.27{tasker-}gnu gatekeeper
07:17.30{tasker-}http://gnugk.org
07:17.33{tasker-}open gatekeeper
07:17.38{tasker-}built on openh323
07:17.39distortioni dislike gnugk
07:17.46distortionbut it works
07:17.47{tasker-}it's flawless
07:17.57distortionits h323 implementation is very specific
07:18.02distortionand euro based
07:18.02{tasker-}stock it's missing stuff
07:18.09{tasker-}we added some code mods
07:18.14distortioni dont like euro h323
07:18.18{tasker-}lol
07:18.25{tasker-}i've never heard of euro h.323
07:18.26distortionits impossible to interop with
07:18.34{tasker-}we have no issues
07:18.57{tasker-}we have carriers going through it from europe, usa, canada, latin america, asia and north africa
07:18.58distortionconnect with something that doesnt re-write h225 and you'll have fun
07:19.00{tasker-}no interop issues
07:19.25{tasker-}rewrite h225?
07:19.55distortionok, pop quiz: q931 dialed number: 19492744000 h225 dialed digits: 200519492744000 what will gnugk route off of?
07:20.20opus_the one that i am not familar the most with of course:)
07:20.26{tasker-}h225
07:20.39{tasker-}i've seen that issue
07:21.08{tasker-}iv'e seen mixed prefixes on q931 and h225
07:21.28distortionhoho
07:21.31{tasker-}but part of the mods we added manage all of that in the routing algorithm
07:21.50{tasker-}we don't use the stock static routing garbage
07:21.51{tasker-}lol
07:22.04distortionwell, i dont hate you then
07:22.05{tasker-}those guys aren;t carrier guys, though they've put together a nice tool
07:22.14{tasker-}give me a break, man
07:22.28{tasker-}even SER needed a few sprinkles for us
07:22.33opus_is there anyway to do intelligent routing in SER? like , these two calls are related: send them to the same server
07:22.39{tasker-}our routing matrix is fairly advanced
07:22.48distortionI've interconnected with some fools out of San Fran that claim to have written gnugk and they ouldnt figure out the q931/h225 routing issues
07:22.56distortioni think they scarred me
07:23.06{tasker-}you can input a million codes and routing decision still only takes an average of 3ms - 8 ms
07:23.30{tasker-}ouch
07:23.31{tasker-}lol
07:26.51opus_so, is there anyway to load balance between servers with SER with some degree of intelligence(like my example)
07:27.17*** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-242.claranet.co.uk)
07:27.53{tasker-}open_: yes
07:28.03websaetainted- are you alive?
07:28.10{tasker-}open_: but you'd have to build the intelligence through scripting
07:28.55{tasker-}damn
07:29.04{tasker-}i meant opus_
07:29.34opus_is there a way to do it after asterisk has already picked up the line?
07:30.09opus_can the scripting do database look ups or would i have to hack that in
07:30.53{tasker-}i don't recall off-hand
07:33.21opus_SER doesn't require any crazy timing device, i could safely run it in vmware right?
07:33.44*** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at)
07:33.44tainted-opus_ yes, i have
07:34.41opus_do I want to get SER or OpenSer?
07:34.48opus_probably openser right?
07:34.54tainted-yea
07:35.08tainted-more activity
07:36.09{tasker-}ser is pretty much dead
07:36.16opus_?
07:36.21opus_what happened
07:36.28{tasker-}it's moved to openser
07:36.34{tasker-}openser is at release 1.1.x
07:38.00{tasker-}the openser repository has the SER revisions up to OpenSER
07:38.13tainted-no
07:38.22tainted-ser != dead
07:38.34{tasker-}for all intents and purposes, it is
07:38.46{tasker-}openser continues from where it left off on revisions
07:40.39*** join/#asterisk {tasker-} (n=ghes@modemcable252.110-83-70.mc.videotron.ca)
07:40.57{tasker-}my wireless router is acting up
07:41.02{tasker-}stupid linksys
07:41.12*** join/#asterisk smurf (n=smurf@debian/developer/smurf)
07:50.41*** part/#asterisk {tasker-} (n=ghes@modemcable252.110-83-70.mc.videotron.ca)
07:50.52tainted-master of minutes enslaved by wifi
07:51.17distortionhe has no minutes :)
07:51.54tainted-u took them all already?
07:51.55tainted-haha
07:52.09distortionnot yet, i think im going to send all mine to him
07:52.29distortionhe's trying to convince me that i should ditch asterisk
07:52.34distortionim starting to agree
07:52.37distortion:(
07:52.37tainted-and go with
07:52.46distortionser/openser
07:52.55tainted-i'm heading to freeswitch as soon as it's stable for production environment
07:53.08tainted-that is a slippery slope
07:53.17tainted-ser is amazing
07:53.19tainted-BUT
07:53.34tainted-doing billing w/ ser is a serious commitment
07:53.50distortionwe havent invested enough time into seeing if it will work for our routing/billing needs
07:54.02distortioninto "ser"
07:54.12tainted-what do u use for LCR
07:54.38distortioni use both "rate-engine" and a modified "lcdial"
07:55.10distortioni love the concepts, they are pretty close
07:55.34distortionrate-engine is good for routes that have congestion because it returns multiple route lists
07:55.59distortionlcdial is cool because its logic is based on sql which is customizable
07:57.23tainted-i see
07:57.51tainted-i'd recommend something db-driven if ur headed into SER land
07:58.22distortioni would love it, im much more db wise than programming wise
07:59.15tainted-i built a lcr engine from scratch
07:59.24websaetainted- you did?
07:59.49tainted-yea
07:59.52distortioni built one as well- ms sql based- very nice. but the switch that uses it sucks
08:00.09tainted-and a web-based billing solution to match
08:00.22websaetainted- can I buy that from  you?
08:00.25tainted-distortion how is * talking to mssql in your setup right now
08:00.29tainted-websae i don't sell it
08:00.33distortionand as i told twisted- i would shoot myself if i used tds w/asterisk
08:01.15tainted-freetds is based on old sybase tsql lol
08:01.21distortiontainted- i dont, i use it for our old h323 platform which im trying to replace w/asterisk (as soon as asterisk's h323 becomes worthy)
08:01.26tainted-but works w/ mssql 7.0/2000
08:01.52tainted-*'s h323 is shit
08:02.11distortionwell, it could work, yes- but would require an application to use it and im not a programmer
08:02.15tainted-how do they talk then? gw & mssql
08:02.38distortionno, h323 session boarder controller runs on MS server 2003 (gasp)
08:02.42tainted-my solution is fastagi based and scales beautifully
08:03.11tainted-i can't wait to do clustering w/ freeswitch though
08:03.24tainted-they intend on release in win32 platform
08:04.01distortionman, i cant wait until this niche overpowers the wholesale switch makers (nextone, sansay ect)
08:04.13distortionfrakin nextone wants 100k for 1000 ports
08:04.24tainted-it will be very very soon
08:04.29distortionand people buy them left/right
08:04.35tainted-there is so much activity right now
08:04.37tainted-it's great
08:04.38distortioni have like 15-20 nextones in my datacenter
08:04.45tainted-damn
08:05.14tainted-i guess i'm fortunate to run sip end-to-end
08:05.15distortionsome jackass has 3 of them fully loaded, (which is like $800k)
08:05.23distortionyes you are
08:05.27tainted-not really into playing the penny pinching wholesale game
08:05.37distortion:'(
08:05.51distortion<-- is :(
08:06.26tainted-why?
08:06.31tainted-did u start out in that market?
08:06.41distortionyeah, i think its all i know
08:06.46distortionsad
08:06.52tainted-only folks making money in that is intl gray routes nowadays
08:07.11distortionwell, that's my co's #1 initiative
08:07.12tainted-grey too lol
08:07.17distortionwhich is sad
08:07.23distortioncause i hate grey routes
08:07.24tainted-what do u mean
08:07.29tainted-yea
08:07.40opus_is there anyway SER can stop an already SIP session, like "oh shit, that asterisk box is blown !"
08:07.41distortionthey make lots of $$ but its fraking way too much maintenance
08:07.51opus_and reroute it to another server?
08:08.31distortionexisting session?
08:08.36tainted-distortion what co?
08:08.43opus_yes
08:08.45tainted-do u work for an itsp?
08:09.21distortionyeah, comsolo is the international side- zues is the domestic.
08:09.59distortionopus: i think you need to run sip(tcp) to do that which means it cant be done w/asterisk yet
08:10.54distortionbut it will work for "next call" which is how we use it
08:11.44opus_is there any way of doing a transparent proxy so that all my rtp data comes from one IP?
08:11.53tainted-distortion i don't see how pure routing is a long term viability
08:12.51tainted-distortion i guess right now the protocols are still settling so there's a niche for transcoders..
08:12.52distortionopus: use ser and a "virtual ip" solution like ultramonkey- there are several out there
08:13.11opus_ok.. hm
08:13.20distortiontainted- pure routing?
08:13.35tainted-opus_ i'm telling u man.. voip slashdot is the next big thing lol
08:13.42tainted-distortion origination/termination
08:14.02distortiontainted- with this industry its less becomming about what you know and how much money you have, and more about who you know
08:14.33distortiontainted- $$ was always the big barrier to entry and with these new open techs, that barrier is fading.
08:14.49*** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au)
08:15.13tainted-yea no kidding!
08:15.21*** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it)
08:15.32distortiontainted- so i agree, those who are closer to the end user will prevail, but there will always be room for arbitrage since the big guys dont want to deal with the small-mid customers
08:15.41tainted-$1/DID, 0.01/US org/term
08:16.27distortion.01 w/ani? :)
08:16.51tainted-dunno.. just numbers i see on whatever peering fabrics/exch etc
08:17.12distortionthere are always catches
08:17.37tainted-i figured
08:17.39distortioni came accross this lame company offering .008 flat us
08:17.43tainted-but it's headed that direction
08:18.05tainted-there's calltermination.com, voipmatch.com, ipxc.com etc etc
08:18.06distortioni signed up, sent minutes and they charged me .03/min cause i was sending "off-net"
08:18.08tainted-it's the trend
08:18.27tainted-thevpf.com
08:18.35distortionim a member :)
08:18.47distortioni went to the conference in miami
08:18.48websaeit's all about the quality......the support........and the follow through
08:19.21distortioni actually think that the vpf (or the like) is the future for voip
08:19.34distortionfrak the internet
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08:20.47distortionbrb
08:22.19*** join/#asterisk Kernel_Core (n=I@193.251.135.118)
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08:30.52*** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de)
08:31.06opus_cool
08:37.11*** join/#asterisk lorinc (n=ang@caracas-2717.adsl.interware.hu)
09:02.16*** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net)
09:06.34*** join/#asterisk ketil (n=chatzill@217-131-74.5001.adsl.tele2.no)
09:07.31*** join/#asterisk websae (n=websae@CPE-24-167-206-22.wi.res.rr.com)
09:07.47websaeso where's everyone from here?
09:07.51websaeUnited States here
09:10.55OliverXG. W. Bush
09:11.03websaehahaha
09:11.04websaeyes
09:11.11websaewhere are you from OliverX?
09:11.40*** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net)
09:20.27*** join/#asterisk Vyeperman (n=Vye@ip68-6-130-59.sd.sd.cox.net)
09:22.20[Airwolf]'morning
09:22.48*** join/#asterisk pengyong (n=lala@218.93.119.110)
09:22.53[Airwolf]I want to give users a SIP accounts and IAX2 account. But woth with the same extension.
09:24.14[Airwolf]But I want to connect to the account where the user is logged on.
09:24.43[Airwolf]Now I solve this like this: exten => s,1,Dial(SIP/user&IAX2/user,20)
09:25.22[Airwolf]It works, but i'm not really satisfied with it, because it will generate an error on the account where the user isn't logged in.
09:25.40[Airwolf]And I was wondering if someone knows another way of handeling this situation.
09:28.40*** join/#asterisk HolyGod (i=nobody@got.securebinary.com)
09:34.27*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
09:34.44Shaun222[Airwolf]: http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ChanIsAvail might do the trick, not sure...
09:34.54Shaun222seams like you could check the channel first and then send the call...
09:39.58*** join/#asterisk blop (i=blop@openbeer.be)
09:40.55*** part/#asterisk littleball (n=littleba@cm55.epsilon171.maxonline.com.sg)
09:41.43*** join/#asterisk CrashHD (n=timf@c-67-182-168-37.hsd1.ca.comcast.net)
09:44.19*** join/#asterisk CrashHD (i=CrashHD@c-67-182-168-37.hsd1.ca.comcast.net)
09:45.39[Airwolf]Shaun222, that was what I was looking for.
09:45.43[Airwolf]Thanks
09:52.26*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
09:53.00inthi
09:53.04inti got a new warning when reloading, Apr 23 11:51:45 WARNING[8280]: chan_zap.c:11050 setup_zap: Ignoring signalling => any idea where it does come from,
09:53.07int?
09:56.40blopk its nothing :)
10:04.29diLLecint: if you reload asterisk those parameters are ignroed
10:04.32diLLecignored
10:05.19diLLecdo a fully restart and they will be applied
10:11.16blopi also get :
10:11.16blopApr 23 12:04:40 NOTICE[8352]: chan_zap.c:6184 ss_thread: Got event 18 (Ring Begin)...
10:11.16blopApr 23 12:04:42 NOTICE[8352]: chan_zap.c:6184 ss_thread: Got event 2 (Ring/Answered)...
10:11.32blopwhich i did not before, and the caller id isnt working anymore on the fxo :(
10:11.40blopcant figure out why
10:23.38*** join/#asterisk Mavvie (n=edwin@252-131-222-203.rev.techex.net.au)
10:25.44*** join/#asterisk ToTo (n=ToTo@host12-137.pool879.interbusiness.it)
10:29.06*** join/#asterisk RoyK (n=roy@cD9088681.inet.catch.no)
10:57.37*** join/#asterisk BearMan (i=karsten@freenode/staff/sourcemage.wizard.BearPerson)
10:58.45*** join/#asterisk RoyK (n=roy@cD9088681.inet.catch.no)
11:07.40*** join/#asterisk tengulre (n=tengurle@222.90.16.134)
11:15.54*** join/#asterisk joat (n=joat@ip70-160-147-169.hr.hr.cox.net)
11:16.36*** join/#asterisk Druken (n=Druken@CPE00121716da99-CM00159a090acc.cpe.net.cable.rogers.com)
11:31.16*** join/#asterisk RoyK (n=roy@80.239.107.70)
11:34.21*** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr)
11:34.30jhiver~seen tzanger
11:34.35jbottzanger is currently on #asterisk (9d 12h 25m 5s). Has said a total of 492 messages. Is idling for 9h 5m 42s, last said: 'yeah it's spitting out white noise and listening, right?  I think... it's been a while'.
11:36.49tzafrir_laptopsomeone asked me if "asterisk supports Radius". I can't find almost any trace for direct support. I figure that there are some workarounds for billing, right?
11:41.43*** part/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org)
11:45.00tzafrir_laptopanybody?
11:46.28RoyKsp,ebpdy
11:46.31RoyKsomebody
11:47.34RoyKtzafrir_laptop: ask jerjer. he loves radius ;)
11:51.35Drukenradius.... pfft
11:54.59RoyKtzafrir_laptop: are you in .il?
11:55.07tzafrir_laptopRoyK, yes
11:55.35tzafrir_laptop(and not in il.us)
11:55.43RoyKdidn't think you were israeli.....
11:57.41*** join/#asterisk saftsack (n=saftsack@p54A7C470.dip.t-dialin.net)
12:01.09coppicesome people are just diametrically opposed to radius
12:11.13loonacyradius is for squares.
12:23.25*** join/#asterisk pengyong (n=lala@222.185.18.214)
12:31.56*** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin)
12:32.04PakiPenguinhello everyone
12:39.23*** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net)
12:45.12*** join/#asterisk Ateboy (n=ugob@modemcable002.152-81-70.mc.videotron.ca)
12:46.14Ateboyshort question about options to Dial()... how do you separate multiple options?  I want to use m and A()...
12:50.06AteboyI've read 'show application dial' and I'm at page 244 of the book, but I can't see examples of more than one option
12:58.02RoyKaM
12:58.13RoyKno separation or anything
13:01.36Drukenuhg...
13:02.13Drukeni remember bitching about dial-up, but now i'm bitching about 800k up.... pretty sad...
13:02.41Ateboyso it would be a(file)m... ok thanks RoyK
13:03.12AteboyI'll try that
13:06.22*** join/#asterisk Flauto (n=zhao@adsl-75-3-189-92.dsl.chcgil.sbcglobal.net)
13:07.09Flauto-- Got SIP response 500 "Server Internal Error" back from 209.47.41.48
13:07.22Flautoi got a lot of this kind of stuff
13:07.30Flautois there anything i can do
13:07.38Flautoeverything seems working
13:12.35Ateboyare you using vbuzzer.com?
13:15.11AteboyI'm trying to use the 'w' option now... doesn't seem to be working
13:15.52AteboyI've uncommented the line 'automon => *1' in features.conf, but whey I do *1, nothing happens (* CLI shows nothing)
13:19.42Ateboyhmmm, sunday morning doesn't seem like the best time to ask questions ;)
13:23.06Flautoateboy, i am using vbuzzer
13:23.22Flautoi don't know why
13:23.40Flautoit is giving me so many these
13:26.21Ateboyflauto: maybe you should ask them...
13:27.32Ateboyflauto: you have connected your * box to vbuzzer?
13:28.03*** join/#asterisk Skymarshal (n=Skymarsc@p54AF3DC8.dip0.t-ipconnect.de)
13:28.13Flautoyes
13:28.16Flautoit is connected
13:28.18Flautoand it is working
13:30.03Ateboysound like a very good way to get cheap long-distance :)
13:30.23Ateboywhat protocol do they support?
13:30.36Flautoyou mean vbuzzer?
13:30.40SkymarshalHi, what is the best why to search for a year in a timebased include?
13:30.45Ateboyflauto: yes
13:30.47Flautoi am really not using it much for long distance
13:30.59Ateboyflauto: what do you use it for?
13:31.03Flautoi use it mostly for inbound
13:31.16Ateboythen, a cheap did :), works well?
13:31.17Flautoand calling toronto area code 416
13:31.33Flautoit has been working pretty well
13:31.53Ateboywell, the 500 error you get, I think only them can tell you what is going on...
13:31.57Flautothis problem started only a few days ago
13:32.09Ateboypossible
13:32.31Flautoit seems that now, they have two ip addresses
13:32.31Ateboybut since it is their servers that return this message, I think they're the only one that can answer you...
13:33.15Flautobut they don't support anything other than their softphone
13:33.30Flautoateboy, there is a new service is called icall.com
13:33.44Flautothat one provides free unlimited long distance
13:33.54Flautobut the problem is that it does not work with asterisk
13:33.54Ateboythanks
13:34.05Flautoyou have to use their softphone as well
13:34.17Ateboyok, so you're telling me you're not using asterisk?
13:34.30Flautoi tired to config is to my asterisk
13:34.34Flautoit did not work at all
13:34.42Flautoi am using it
13:34.55Flautobut can not make icall to work on my asterisk
13:35.18Ateboyif they force you to use their softphone, they're probably using a proprietary protocol, like skype
13:35.48Flautothey are using sip
13:35.58Ateboyhmmmi
13:36.41Ateboythey must be doing something else
13:37.29Ateboybut if you didn't get the 500 error on your asterisk, you shouldn't be asking on an asterisk list.... or at least say it (maybe you said it before I got in the room, though)
13:39.01Flautothat is okay
13:44.24Flautohope someone will soon figure out how to config icall to asterisk
13:44.37Mw3hm, is there any problem with fwd at the moment? it seems to be down :(
13:45.23RoyK~seen wasim
13:45.33jbotwasim <n=wasim@pdpc/supporter/active/wasim> was last seen on IRC in channel #asterisk, 3d 23h 43m 33s ago, saying: 'guigouz: an fxo/sip ata'.
13:45.33Flautoat beginning, i had problem with vbuzzer too but later, i figured it out and then, i saw the config on the wiki as well
13:45.44Flautomw3
13:45.46Flautolet me try
13:47.45Flautomw3, you are right
13:47.47Flautonot working
13:48.52*** part/#asterisk Cybertoy (n=maxim@dsl254-123-241.nyc1.dsl.speakeasy.net)
13:48.53Mw3even the web site is down as i see
13:49.23Mw3dns cannot resolve the addresses (iax.fwdnet.net fwd.pulver.com www.freeworlddialup.com)
13:51.54RoyKMw3: I can't resolv them from here either
13:52.13*** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se)
13:52.19Mw3ok thanks :) i hope it will be back soon
13:58.47*** join/#asterisk pa (n=Paolo@unaffiliated/pa)
14:01.40RoyKhej hej
14:01.56coppiceis this news?
14:02.06Flautohehe
14:02.15RoyK:)
14:02.39*** join/#asterisk jhiver_ (n=jhiver@89-114.206-83.static-ip.oleane.fr)
14:02.44jhiver_~seen tzanger
14:02.48jbottzanger is currently on #asterisk (9d 14h 53m 18s). Has said a total of 492 messages. Is idling for 11h 33m 55s, last said: 'yeah it's spitting out white noise and listening, right?  I think... it's been a while'.
14:03.03jhiver_~seen jhiver
14:03.04jbotjhiver <n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr> was last seen on IRC in channel #asterisk, 2h 28m 34s ago, saying: '~seen tzanger'.
14:03.19jhiver_:)
14:03.45jhiverhi all
14:04.06RoyK~lart jhiver
14:04.30jhiverblue or red?
14:04.33jhiverthe pill :)
14:05.00coppicesounds like an ad for 7-up :-)
14:06.34jhiverI've been playing with SER recently
14:06.46jhiverit's pretty good but its config files are such a mess
14:06.57coppiceah, everyone has a sad story to tell
14:07.00*** join/#asterisk juanjoc (n=juanjoc@248-32-235-201.fibertel.com.ar)
14:07.10jhiverand also the documentation is piss-poor / out of date... I guess I should contribute :)
14:07.38jhiverI'm thinking of writing a ser.cfg Perl generator but I'm wondering what a good syntax might bze
14:07.48coppicerefactoring code is fine, but most project need to much more seriously refactor their configuration scheme
14:08.00jhiverbecause actually this software doesn't seem to do very much at all :)
14:08.20*** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239)
14:08.30[TK]D-FenderGodd morning all.....
14:08.32jhivermind you that's also a strengh... do something simple and do it well
14:08.33jhiverhi
14:09.21[TK]D-FenderQuick question : has a recent release of * broken unauthenticated incoming SIP calls?  My config used to work, and those of a few people I talk with and none of ours seem to work any more...
14:09.59jhiveroh, sounds dangerous
14:10.12jhivermaybe I'll wait a little before I upgrade from 1.0.9 :)
14:10.35russellb[TK]D-Fender: when upgrading to 1.2 from 1.0, you need allowguest=yes
14:10.39Drukeni'm afraid to upgrade... i'd have to rebuild my entire dialplan... :(
14:10.41russellbin sip.conf
14:10.45[TK]D-Fenderrussellb : Always been doing that...
14:10.58jhivermind you the 'n' priority is just so right :)
14:11.19russellbI hope more and more people start using AEL in 1.4
14:11.30Drukenael?
14:11.44Maxxedmorning' folks :)
14:11.46jhiverrussellb, why? it doesn't look very good, too much like a programming language
14:11.51[TK]D-FenderAll Efforts Lost :)
14:11.59jhiverAs if SER configs wasn't enough :)
14:12.09russellbjhiver: it is a language
14:12.14Maxxedhey, is there a way to have a caller thats been in a queue for a long enuff time to get a voicemail box?
14:12.19jhiverwell
14:12.24Drukenser's config is fuct
14:12.38Maxxedie, caller sits in queue for 10min, they get vmail
14:12.52jhiverif you want a language for a dialplan then maybe use python or perl or something
14:12.57Maxxedi know i can gotoif a var n such
14:13.06russellbDruken: http://svn.digium.com/view/asterisk/team/murf/AEL2/doc/ael.txt?view=markup
14:13.08jhivernot some other stupid _new_ language
14:13.18Maxxedbut is there an easy, or rather nicer way to do it
14:14.08jhiverI think dial plans should look like access list
14:14.41jhiveras in: condition1 condition2 condition3 action
14:14.41[TK]D-Fenderrussellb : I get "SIP/2.0 407 Proxy Authentication Required" when I get the call.... though I have "allowguest=yes, and there is an exten for his "to:" header
14:16.38russellbtoo early to look a SIP trace
14:17.32*** join/#asterisk mutilator (i=WebChat@65.111.201.122)
14:17.33[TK]D-FenderWell there is a context in [general], there is a matching exten in extensions.conf, and in it an exten named "andrew" which is what he's dialing
14:17.47[TK]D-FenderThat sounds like all 3 pieces to me...
14:18.00[TK]D-Fender(and the fact it used to work...)
14:22.49blitzragemadness
14:23.30*** join/#asterisk uski (n=uski@ALagny-151-1-83-209.w86-198.abo.wanadoo.fr)
14:25.53[TK]D-Fenderugh..
14:26.43RoyKugh..
14:27.44Maxxedeh..
14:28.11Drukenblah....
14:36.20[TK]D-FenderYeah, its happening on 3 servers now...
14:36.23[TK]D-Fenderdammit
14:36.54*** join/#asterisk Assid (n=assid@203.115.64.8)
14:40.35[TK]D-FenderMight have something to do with SIP domain?
14:52.00RoyKhttp://freewlan.org/ <-- My latest project :)
14:56.01Maxxedso yeah, any idea how to time out a queue
14:56.15Maxxedsombody in the queue for 5min gets voicemail
14:56.37*** join/#asterisk Frogzoo (n=Frogzoo@202.155.165.25)
14:56.43Maxxedi have the time out set in the queue.conf
14:56.54*** join/#asterisk asteriskmonkey (n=phil@69.158.144.16)
14:56.54Maxxedbut that dont seem to work the way i think it would
14:57.19asteriskmonkeyhey , has anyone ever ordered the g729 codec licenses from digium before?
14:58.04Maxxedi havent
14:58.06Maxxedi thought about it
14:58.13Maxxedbut i dont know what all the hype is about
14:58.21Assidi have
14:58.22Maxxedlike really.. what makes it so great?
14:58.24Assidtakes around 10 mins
14:58.26*** join/#asterisk apardo (n=apardo@87.217.144.163)
14:58.29asteriskmonkeyim trying to find out if they email you the key or you have to wait for it in the mail
14:58.36Assidwait for the eamil
14:58.40asteriskmonkeysweet
14:58.46Assidi think they send out the email manually
14:58.47MaxxedAssid: is it really all that badass?
14:58.54Assidbadass? wouldnt know
14:58.57Assidjust a damn codec
14:59.07Maxxedyeah.. bump paying for just a codec
14:59.14asteriskmonkeyMaxxed: just a goode codec to use if you have little bandwidth
14:59.24Maxxedah
14:59.41asteriskmonkeyso if you wanna do 20 lines of voip on a dsl you would want g729
14:59.58Maxxeddamn, 20 lines on dsl
14:59.58Assidim thinking of that
15:00.01Maxxedis that even posible
15:00.04asteriskmonkeyyes
15:00.08Assidthere is only 1 issue
15:00.14Assidyour carrier should be 729 compliant
15:00.16asteriskmonkeyhave lots of people using audiocodes gatways with that :)
15:00.41asteriskmonkeyim a carrier trying to become g729 compliant lol
15:01.00Assidi actually wanna be a carrier
15:01.06Assidmaybe on a retail level
15:01.09asteriskmonkeydo you have a t1 or ds3
15:01.11Assidi know ic an get tons of business
15:01.19asteriskmonkeyAssid where are you located
15:01.27Assidme? personally ?india
15:01.40asteriskmonkeyah
15:01.49Assidhence why good opportunity to start up
15:01.50asteriskmonkeyto many grey routes for voip there
15:01.57Assidwell
15:02.01Assidim not gonna link into PSTN
15:02.06Assidso im doing it all legal
15:02.30asteriskmonkeyi guess but then you cant sell termination to pstn to other providers
15:02.38Assiderr.. no
15:02.43Assiddont wanna do that
15:03.10Assidinstead of letting the west communicate with east.. im gonna let east communicate with west
15:03.27ManxPowerCommunications is far overrated.
15:03.32Assidwell
15:03.39Assidso long as it pays my phone bills
15:03.40Assidhehe
15:03.48[TK]D-FenderJust got it... I have an ext configured mathcing the caller ID of the inbound call and it thinks its a LOCAL phone....
15:04.08Assidhuh?!?!?
15:04.13[TK]D-Fendernote to self : Make all []'s in sip.conf RANDOM looking...
15:04.26Assidrandom looking huh?
15:04.28Assidwhy
15:04.53[TK]D-FenderAssid : read up...
15:05.10ManxPowera MAC works
15:05.33[TK]D-FenderManxPower : Good for hardphones, less so for softphones, but at least it'd be meaningful where applicable...
15:05.49ManxPowerall softphones suck!
15:06.05Ateboylol
15:06.19Assidhow is your pbx getting a matching caller id as incoming call
15:06.42Assidyou using whole numbers per extension?
15:07.27Assidi wanna start playing with realtime soon
15:08.29asteriskmonkeyif you have a large did range just use a matchign expression
15:09.12Assidyou know.. we really should have a universal dundi and open access system.. to try and make things cheaper for everyone
15:09.22asteriskmonkeyexample _4165983XXX,1,Dial(SIP/${exten})
15:09.38asteriskmonkeyassuming you name your sip accounts as the did
15:09.58asteriskmonkeyassid: its called msn
15:10.06Assidhuh?
15:10.29Assidmsn doesnt do jack
15:10.52asteriskmonkeymakign a global dundi network would be a disaster
15:10.55[TK]D-FenderManxPower : True, but its what my mother can use in her case :)
15:11.07ManxPowerall softphones suck!
15:11.08asteriskmonkeyyou could 1)never assure quailty 2)never prevent idiots abusing it
15:12.01asteriskmonkeyif you want an example of uber crudy dundi check out voip discoutn with over 32 countrys to call for free and all you get is pops and audio drops
15:12.22Assidwell. quality should theoretically be better .. as its a voip-voip link without any traditional phones in the middle to mess it up
15:12.33Assidhrmm
15:12.40Assidaudio hasnt been droppng for me
15:12.43Assidalthough
15:12.54Assidthey arent supposed to be charging me for my calls to hongkong
15:13.15coppiceAssid: nobody charges me for calls to Hong Kong
15:13.31Assidcoppice: know any free providers?
15:13.38asteriskmonkeylots
15:13.50asteriskmonkeysipdiscount, voipdiscount, voipbuster
15:13.51coppiceYeah. the local telco. i live in HK :-)
15:13.52asteriskmonkeylots
15:14.04Assiderr.. thats all the same..
15:14.14asteriskmonkeyyep
15:14.15Assidsipdiscount/voipdiscount and voipbuster belong to the same guys
15:14.19*** join/#asterisk Hali_303 (n=surfk@dsl540096E3.pool.t-online.hu)
15:14.21asteriskmonkeywoo just got my g729 codecs
15:14.38Assidnice
15:14.42Assidenjoy:D
15:14.44Hali_303asteriskmonkey: how much do they cost?
15:14.52asteriskmonkey10$ per channel
15:15.04Hali_303and how is a channel defined?
15:15.11asteriskmonkeya call
15:15.39asteriskmonkeyan audio speech path :P
15:15.41Assidactive line is a channel
15:15.42Hali_303hmm i see. and why is it better than speex at low bitrate?
15:15.48Assidspeex sucks
15:15.55Assidsounds like robots
15:15.57asteriskmonkeybetter quailty
15:16.09asteriskmonkeyg729 sounds pretty good and only eats up 8kbps
15:16.14Jackeinteroperability  ;>
15:16.17asteriskmonkey+overhead
15:16.25Hali_303I used speex@7600bps and it was a bit robotic but not much
15:16.29asteriskmonkeyualw use nearly 90k with overhead
15:16.35coppiceif speex sounds bad at 8k, something is wrong with your setup
15:16.45asteriskmonkeytry explaining codecs to custoemrs.. they only care about quailty
15:16.51coppiceg729 uses 30K with overheads
15:16.58Hali_303asteriskmonkey: yeah that is  true..
15:17.03AssidG.729  8  31.2
15:17.12Hali_303asteriskmonkey: no hardware phones w/ speex inside?
15:17.18Assidethernet bandwith = 31.2
15:17.34Assidaround 3.9K/sec
15:17.47Assidso effectively 4K/sec up/down you have a clear call
15:17.56Assidthats good enough for even a 56kbps dialup
15:17.57RoyKcoppice: huh? IP+UDP+RTP overhead with 20ms packetization is 16kbps
15:18.01coppiceasteriskmonkey: If your customer care about quality, how come that will accept G.729?
15:18.02*** join/#asterisk pauldy (n=pauldy@24-155-86-154.ip.grandenetworks.net)
15:18.02RoyKcoppice: so 24 in total
15:18.12coppiceRoyK: try again
15:18.26RoyKwhy?
15:18.28Assidcoppice: effectively .. 729 is lossless codec
15:18.34Assidso its better than ul
15:18.42asteriskmonkeycoppice: all my customers are ulaw :) the g729 is for high volume places that want more than 6 concurrent calls on a dsl
15:18.51coppice729 is a *very* lossy codec
15:18.56Assidits lossy ?
15:19.13asteriskmonkey729 is good but you have to keep it under 150ms otherwise it goes to crap
15:19.15*** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se)
15:19.21coppice729 is *far* worse than ulaw or alaw, too
15:19.28Assidweird
15:19.30asteriskmonkeyulaw and alaw are the best
15:19.35asteriskmonkeythere the least compressed
15:19.36Assidi do a few calls on 729
15:19.37Assidworks fine
15:19.42coppiceAssid: where on earth did you get the idea 729 is lossless
15:19.56Assidsome stupid forum i read longggggg ago
15:20.00Assidguess it stuck with me
15:20.29asteriskmonkeynow i can take out my intel testing g729 ones :)
15:20.44Assidhehe
15:20.49asteriskmonkey729 is good if the distance is short
15:20.51Assidsee any different in translation times
15:20.54Hali_303Assid: you need 64kbps for telephone quality, which is lossless audio sampled at 13 bits (compressed into 8 bits) at 8khz
15:21.14asteriskmonkey729 however adds a lot of cpu usage to your box so you cant exactly go mad with it
15:21.22Assiderr...
15:21.27Hali_303I think the problem w/ speex was not with quality, but with delay!
15:21.29Assidasteriskmonkey: only if you transcode
15:21.43coppiceHai_303: not really. you can get *very* similar quality to ulaw at something like 25kbps. you can't get it at 8kbps, though
15:21.48Assidif caller/callee is 729.. no issues
15:21.52asteriskmonkeyAssid: voip client > pstn = transcode
15:22.05Assidoh.. you got a fxs interface
15:22.11asteriskmonkeylol no
15:22.11Assiderr.. fxo even
15:22.15asteriskmonkeytry a ds3
15:22.17Assidaah
15:22.22Assidthatd exaplain it
15:22.25Hali_303coppice: yes. very similar, but not the same!
15:22.39Assidso how many licenses did you buy
15:22.53coppiceHali_303: well you can do a lot better than ulaw at about 30kbps
15:23.16Assidhow the hell do you set ulaw to eat less bandwith?
15:23.46Hali_303Assid: you cannot. ulaw is a standard, which makes audio sampled at 13 bits into 8 bits
15:23.47asteriskmonkeyyou dont
15:23.50Hali_303you cannot set that
15:24.06asteriskmonkeyyou get a bigger internet connection
15:24.08Assidasteriskmonkey: when you gong live?
15:24.18asteriskmonkeyAssid: i am live
15:24.34Assidretailing?
15:24.44asteriskmonkeywholesaling to dealers
15:24.58Assidsite?
15:25.07asteriskmonkeyits uber plain www.massivetel.com
15:25.23Assid11:25:15 (1.28 MB/s) - `asterisk-1.2.7.1.tar.gz' saved [10554037/10554037]
15:25.27asteriskmonkeyill probably rejig it sometime soon but has been ok for past 8months :D
15:25.27Assidi think thats enough bandwith?
15:25.50asteriskmonkeythats downstram
15:25.55asteriskmonkeyit upstream that counts
15:26.11Assidhrmm
15:26.33Assidwell.. its a dedicated server..
15:26.37asteriskmonkeyso
15:26.56asteriskmonkeydosnt matter if its dedicated matters what connectivity is plugged into it and if its throttled or not
15:27.16asteriskmonkeytry sending something to another backbone server that will give you an idea
15:27.40asteriskmonkeythen you can just div you upstream by 90k to see how many users you can handle
15:27.48asteriskmonkeyttyl
15:28.04Assidhrmm around 800K up
15:28.17Assid> 1 if i do to my other box
15:28.22Assidoh well
15:28.48Assidhrmm.. as i was saying.. i wanna play with RT
15:31.25*** join/#asterisk viperdude (n=viperdud@84-45-168-60.no-dns-yet.enta.net)
15:31.58viperdudehi anybody got experience of using SER with Asterisk?
15:34.28Assidokay time to upgrade one of these old boxes
15:35.13Assiderr.. is it worth upgrading from 1.2.4 to current?
15:45.54dpryoIs it possible to get the number a call was transfered from? (c-number?)
15:47.18*** join/#asterisk Grubs (n=Miranda@c220-239-223-90.eburwd3.vic.optusnet.com.au)
15:47.36Assidwell.. you could set a variable before transferring the call
15:47.41Assidi think that could do it
15:48.05dpryoWell, I'm not doing the transfer, the telco is transfering a number in to me.
15:48.11Assidoh
15:48.18Assiddoesnt the caller id change then
15:48.32dpryoNope, I get the correct callerid
15:48.43dpryoI guess the telco sets it up..
15:48.49Assidthen i dont think you can
15:49.05dpryoI've seen some talk about 'c-number' containing it
15:49.12Assidnot sure..
15:49.16Assidask around
15:49.22dpryoBut i'm not sure how to debug it
15:51.24GrubsJust upgraded from 1.2.5 to 1.2.7.1 and I am seeing a new 2-3 second delay after dialing before asterisk responds (dialing and voicemail prompts). Any ideas on what to look for to eliminate it?
15:51.51GrubsuseCallerID=no so it isnt caller ID detection.
15:53.08Strom_Cthis is going to seem like a really silly question, but how do you get asterisk to start at boot-time?
15:54.45GrubsStrom_C: - http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x430.html
15:55.36Strom_Cthanks :)
15:55.57Flautofwd is down
15:56.09Flautocan not evenopen their website
15:57.50*** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se)
15:58.09Jackeanyone running asterisk on freebsd here?
15:59.02NuggetI do.  I don't recommend it.
15:59.39Jackewhy not?
15:59.50Jackeand which version of asterisk are you using.
16:00.32Nuggetthe zaptel-bsd driver lags the real zaptel stuff and has always been really flaky for me.
16:00.48Nuggetif you can live without zaptel it's pretty viable.
16:01.31Jackewell, with my TE410P i'm not in the position to live without it ;)
16:01.46Nuggetit's also hard to get support from the community if you have problems.  You'll be forced to listen to a 15 minute diatribe on how ubuntu is going to change the world and how asterisk works just great if you'd just switch to their favorite distro.
16:02.11dpryoNugget: It's true! You know it!
16:02.15Jackethe problem is i don't like their distros ;)
16:02.21SedoroxNugget: ahahhahaha
16:02.24JackeI don't like their operating system at all :)
16:02.33JackeNugget: Which version of asterisk are you running?
16:02.36Nuggetneither do I, but I tolerate it for asterisk.
16:02.57SedoroxI like fbsd for networking shit...
16:03.01DoktorGregDogma is the one true religion!
16:03.01Sedoroxnot for normal workstation...
16:03.12Jackeon the FreeBSD i mean.
16:03.17Nuggetof course not.  os x is for normal workstations.  ;)
16:03.28*** join/#asterisk CrummyGummy (n=wayne@dsl-145-99-210.telkomadsl.co.za)
16:03.30Nuggetfreebsd for servers, openbsd for firewalls, and linux for asterisk.
16:03.35Jackewell, as if an asterisk machine was going to be a workstation ;)
16:03.42SedoroxNugget: if I had a mac.. thats what I would run (of if osx86 ran on my laptop...)
16:03.44coppice* has *real* problems with OS/X :-)
16:03.57Nuggetreal problems spelling it?
16:04.04DoktorGregosx has 'realtime' problems i hear
16:04.16Sedoroxand look.. here we go on a tangent about operating systems :p
16:04.22coppiceall OSes have realtime problems
16:04.27Nuggethey!  I'm being a grammar troll, not an os troll!
16:04.33Sedoroxlol
16:04.33Nuggetget it right, Sedorox.
16:04.38Sedoroxno no.. I mean in general
16:04.40DoktorGregwhich begs the question
16:04.41Sedoroxnot you alone :p
16:04.46DoktorGregwhat is realtime?
16:04.54Sedoroxwe're going on about fbsd.. obsd.. linux... osx... etc.. :p
16:05.03DoktorGregcan someone splain it to me in a nutshell?
16:05.08coppicerealtime is something OSes don't do properly :-)
16:05.28Nuggetin an asterisk context, "realtime" means "database-backed asterisk"
16:05.39Nuggetin an os context, "realtime" means guaranteed response time
16:06.34coppiceand guarantees of response time are nearly impossible to achieve these days
16:06.35Nuggetgeneral purpose operating systems can't do true "realtime" processign, no matter how much ugly crap you bolt onto them, whcih is the point I believe coppice is making.
16:06.50Nuggetand even if it were feasible, you probably wouldn't want to
16:07.04*** join/#asterisk tier_1 (n=tier_1@c-24-9-75-234.hsd1.co.comcast.net)
16:07.11*** join/#asterisk Gamercjm (n=chris@pool-71-254-177-36.lsanca.fios.verizon.net)
16:07.15coppicethe processors can't do real time either, unless you turn off their caches
16:07.33tier_1http://pastebin.ca/50702
16:07.38tier_1have fun
16:07.51tier_1NANPA config
16:08.16Corydon76-homeWell, if you had enough spare cores, you could guarantee response time
16:08.39Nuggetnot with a bus to mitigate.
16:08.53coppiceCorydon76-home: sounds like you don't even begin to understand the problem
16:09.01Corydon76-homebut master/slave CPUs has already been demonstrated to be quite inefficient
16:09.03Nuggetyou have to stop servicing interrupts if you want to guarantee a response time
16:09.27DoktorGregso if i wanted an application to be 'realtime' I would have to design something from the ground up?
16:09.45Nuggetor you could use an off-the-shelf realtime system.
16:09.47DoktorGregor rather get an architecture that is designed to be realtime?
16:09.49Sedoroxclockless cpus!!!!!111elevenone
16:10.14Nuggetheh
16:10.31Sedoroxman... amd64 but clockless.... yummy....
16:10.33Sedoroxanyway...
16:10.33Sedorox:p
16:10.47*** join/#asterisk ketil (n=chatzill@217-131-74.5001.adsl.tele2.no)
16:11.29x86wow.... 3 year wedding anniversary today for my wife and I
16:11.38x863... long..... years ;)
16:11.41Sedoroxcongrats
16:11.49x86thanks :)
16:12.30Corydon76-homeYou can have an efficient system or you can have a realtime system
16:12.47Sedoroxanyone want to wite my persaisive (sp) speech on why rfid tagging is bad? :p
16:13.05coppiceyou can have a highly efficient real time system, but its far from general purpose
16:13.27Corydon76-homeTrue enough
16:13.29DoktorGregwell googling around
16:13.45DoktorGregit looks like fly by wire flight controll systems have to be real time
16:13.54Corydon76-homeSo, Realtime/Efficient/General Purpose... pick any two...
16:14.09DoktorGregand frame grabbers for science experiments have to be real time
16:14.17Nuggetno!  you're not allowed to pick.  everyone must run my favorite distro of linux otherwise I lose.
16:14.24Sedoroxlol
16:14.40Sedoroxlet me gues.s.... your a debian user? :p
16:15.05DoktorGregi am contemplating an OSX for my desktop
16:15.23Corydon76-homeI have a few nub's who have picked Debian for their systems and then expect me to be able to advise them how to administrate it
16:15.26DoktorGregbut i play a game a week out of every three months
16:15.33NuggetI'm a slut.  I use just about everything.
16:15.39Corydon76-homeHomey don't play dat.
16:15.39Sedoroxlol
16:15.47SedoroxI'm basically gentoo or fbsd now...
16:16.00fileNugget: SLACKER!
16:16.05Nuggetthat's me!
16:16.10DoktorGregAll will bow to the Hurd!
16:16.17Corydon76-homeSomebody sold them on Debian, then PROMPTLY DISAPPEARED.
16:16.19fileno, all will bow to OS/2
16:16.38blitzragehail OS/2!
16:17.01NuggetDoktorGreg: OS X runs world of warcraft and civilization. Those are the only two games I need.  :)
16:17.12Sedoroxahahah
16:17.19DoktorGregNugget, last game i played was DnD online
16:17.22Sedoroxand enemy territory... can't forget that one :p
16:17.36Nuggetah, yeah, I play call of duty when I want to shoot nazis.
16:17.46Sedoroxhehe
16:17.58DoktorGregI havent played a nazi shooting game for a while
16:17.59fileblitzrage: you're supposed to be working out
16:18.21DoktorGregoh i know i totally burned out on nazi shooting on bf1942
16:18.39x86Nugget: niether WoW or Civ have x86 binaries for OS X
16:18.44Nuggetwow does.
16:18.51x86Nugget: you sure?
16:18.54tier_1Freeciv
16:18.55Nugget100%
16:18.56Sedoroxyes
16:19.06Nuggetwow has been a universal binary for a few months now
16:19.06SedoroxI know a lot ofp eople with wow on osx86
16:19.42DoktorGregI more or less refuse to play games like wow since i lost a couple of friends to EQ
16:19.51Nuggetyeah, that's a tangible risk.
16:20.28NuggetI mitigated it by getting my partner hooked too, so we're both equally hopeless and it works out just fine.
16:20.34Sedoroxmy roommates tried to get me to play it when it was in beta
16:20.35Sedorox:p
16:20.37Sedoroxand free...
16:20.47SedoroxNugget: lol
16:21.00Nuggetshe's playing right now, in fact.  I can hear the music.
16:21.02SedoroxI wouldn't play tho.. I know I would get addicted to it.. so I didn't bother...
16:21.27DoktorGregI get bored with a particular game after two weeks max
16:21.45DoktorGregso problem with muds for me
16:21.46*** join/#asterisk dahunter3 (n=dahunter@pool-71-110-89-49.lsanca.dsl-w.verizon.net)
16:21.53Sedoroxwe play ET here at school a lot.. but its weird.. we'll go on kicks where for like 3 weeks straight we play nothing but it.. solid.. whenver we can...
16:21.56DoktorGregis they take about a month to really get into them
16:22.01Sedoroxthen we'll go three weeks where no one plays
16:23.02NuggetI never liked W:ET.  the controls and gameplay are too arcade style.  I like DoD a lot better where the combat is less "point and spray" and gameplay is more deliberate.
16:23.25DoktorGregapex of shooters was bf1942
16:23.42DoktorGregIMO
16:24.26DoktorGregand single combat was irrelevant
16:24.41DoktorGregyou had to work as team or het pwned
16:26.09NuggetI'm in an endgame guild in WoW which is a really neat experience.  20 and 40 player teams who all have to work in lockstep to even have a chance of succeeding at something
16:26.33Nuggetit's a real rush when everyone's focused
16:33.19*** join/#asterisk stoffell_h (n=stoffell@81.164.209.43)
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16:51.40Flautois anyone here using icall.com
16:51.48Flautoi mean using it with asterisk
16:53.12*** join/#asterisk jovan (n=jovan@host151-99.pool8711.interbusiness.it)
16:53.21jovanhi
16:53.56Flautohi
17:04.43*** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net)
17:08.27*** join/#asterisk ramo (n=ramo@59.92.200.207)
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17:15.09*** join/#asterisk SkramX (n=mark@admins.sentiensystems.net)
17:15.31*** join/#asterisk |dennis| (i=dennis@200.32.215.84)
17:15.36SkramXwhat would i need to do, to be able to handle a lot of calls and just do passthrough/call accounting? SER? no codec changes needed.
17:24.31*** join/#asterisk Skarmeth (n=Skarmeth@201008202192.user.veloxzone.com.br)
17:28.42*** join/#asterisk IceManRISK (n=kart@201.10.94.122)
17:31.22OliverX== Everyone is busy/congested at this time (1:0/0/1)  // Can anyone help me?
17:32.16Assidwell
17:32.34Assidif you just do SER, that would be exactly what you need
17:35.40SkramXAssid: okay.
17:35.49SkramXwhat kind of server resoirces are we talking?
17:37.45Assidwell.. not sure.. can handle couple hundred.. on a xeon
17:37.47Assidthats for sure
17:37.53Assid150 odd +
17:38.09Assidatleast thats what i read
17:38.12Assidi could be mistaken
17:38.27SkramXhrmm
17:40.02*** join/#asterisk Samoied (n=Samoied@201-3-227-215.fnsce7002.dsl.brasiltelecom.net.br)
17:41.22SkramXa xeon for only 150+?
17:41.33Assidumm.. whats the makefile config for K8 ?
17:41.36Assidathlon64?
17:41.47Assidnot sure SkramX
17:42.15*** join/#asterisk mmmmmToop (n=chatzill@dsl-165-166-229.telkomadsl.co.za)
17:42.27AssidPROC=k8 ?
17:42.38Samoiedhello all!
17:43.06Samoiedanyone have tested miax with bluetooth?
17:43.44SamoiedI have tried with a nokia 6600
17:44.12Samoiedbut miax try to use the phone as modem, not a handset
17:46.40Assidguess i will just set it to athlon
17:47.42IceManRISKAnyone here uses JIAX /
17:49.46SkramXso..
17:49.48IceManRISKAnyone here uses JIAX ?
17:50.51*** join/#asterisk Alric (n=nbowyer@ppp-db.1stel.com)
17:51.11*** join/#asterisk stakk (i=sted@85.10.196.41)
17:51.56Assidokay i gotta run
17:51.58Assidlaterz
17:53.40*** part/#asterisk BearPerson (i=karsten@freenode/staff/sourcemage.wizard.BearPerson)
17:54.31websaeanyone from Canada here?
17:55.09*** join/#asterisk BladeRunner05 (n=feelme@81-174-56-54.f5.ngi.it)
17:56.12Drukenyep
17:56.48Drukenwebsae: why ya lookin for canadians?
17:57.17websaejust to say, hello :)
17:57.31Drukenhello.....
17:57.33websaeI am in Wisconsin -- pretty close to Canada
17:57.42websaedo you have wholesale routes there?
17:57.55Drukenhow's the cheese? hehe
17:58.05websaemoldy :)
17:58.47Drukenwebsae: msg me priv with what your lookin for
18:09.34NewSolehmm
18:21.53*** join/#asterisk FreddyFeuer (n=email@p54AE01A6.dip0.t-ipconnect.de)
18:33.17dpryoIs it possible to "traceroute" PRIs? .. like see how many hops there is to the destination?
18:33.46macTijnmostly that's just 1
18:33.52macTijnwhat would you count as a hop ?
18:33.58dpryoA pbx
18:34.02dpryoor something
18:34.34dpryoMy calls are going through at least 3 different pbxes
18:35.00dpryocisco -> asterisk -> avaya -> ..
18:35.17tainted-dpryo that is pretty cool
18:41.45xbmodder_lappyhow is the stability of chan_bluetooth
18:43.53dlyneswebsae: I'm from Canada, too
18:44.00FreddyFeuerkann mir jemand bei meiner asterisk konfiguration helfen?
18:44.11Hmmhesaysput your hand up on your hip, i dip, you dip, we dip
18:45.15dlyneswebsae: Not able to talk right now, though...I'll be back in about an hour if you want to msg me about what you want
18:45.48VoIPMastaFreddyFeuer: Ich zweifele, daß Sie jemand finden, das Deutsches innen hier spricht
18:46.55FreddyFeuerhmm.
18:53.32*** join/#asterisk firestrm (n=no@S01060013105c1c69.gv.shawcable.net)
18:53.45firestrmhello
19:01.36firestrmanyone here good at wierd IAX problems?
19:01.46*** join/#asterisk pdunkel (n=pdunkel@213.235.231.189)
19:03.15*** join/#asterisk Poincare (n=jefffnod@amp89.ampersant.be)
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19:11.25doolphanyone know to fix this problem, when I try to enter any number in a ivr menu the system doesn't accept it
19:11.59*** part/#asterisk SkramX (n=mark@admins.sentiensystems.net)
19:18.25*** part/#asterisk Skarmeth (n=Skarmeth@201008202192.user.veloxzone.com.br)
19:19.43ManxPowerdoolph, many possible causes.  you may need to increase or decrease the rxgain or txgain on your ZAP ports.
19:20.06doolphI am using sip trunk and sip extension
19:20.09ManxPowerYou may also be trying to do inband DTMF on a call that uses a compressed codec and EVERYONE knows that won't work.
19:20.44[Airwolf]I both have users with IAX2 and SIP accounts.
19:21.21[Airwolf]But I want to create a macro that dials the account (iax2 or sip) on which the user has logged in.
19:21.49ManxPower[Airwolf], I know of no easy to do that.
19:21.50[Airwolf]This morning I got the tip to use the ChanIsAvail function.
19:21.55ManxPowerWhy not just dial them both?
19:22.11Hmmhesaysset a variable in sip.conf and use that
19:22.24[Airwolf]ManxPower, that is the solution, the problem is it isn't really good.
19:22.25*** join/#asterisk espino (i=espino@srv4.v-expressa.com.br)
19:22.38[Airwolf]exten => s,1,Dial(${ARG2}&${ARG3},${ARG4})
19:22.39ManxPowerHmmhesays, not going to do much good for OUTGOING calls.
19:22.47[Airwolf]I can solve it something like that
19:22.50HmmhesaysManx, why not?
19:23.02ManxPowerHmmhesays, because that variable won't be set.
19:23.08[Airwolf]But it creates an error on the channel where the user isn't logged in.
19:23.12Hmmhesaysnm
19:23.36ManxPowerSay you have a Setvar for a SIP account.  A call comes in via Zap and gets sent to that SIP account.  The variable will not be set.
19:23.39Hmmhesaysseperate out your extensions numbers, make 6XX sip and 7XX iax or osmething like that
19:24.04ManxPowerHmmhesays, the SAME user is allowed to register BOTH IAX and SIP.
19:24.28[Airwolf]That is a possibility, but that doesn't fit in my local dialplan.
19:24.40*** join/#asterisk wikityler (n=yo_tyler@d66-183-163-151.bchsia.telus.net)
19:25.02[Airwolf]I think it just has to be the quick & dirty solution by calling the channels at the same time and just leave the error to be.
19:25.08wikityleryay, im finaly in.
19:25.30ManxPower[Airwolf], other providers either 1) make a user PICK if they want IAX or SIP (and only 1 can be selected at a time) or 2) send all calls to BOTH SIP and IAX registration for that user
19:26.11wikitylerquich and dumb voip question: if im using an ata, can it be on the switch that the server is on, or does it have to be connected directly to a nic on the sever?
19:26.36VoIPMastawikityler: you can connect it to the switch
19:26.46wikitylerok, thankyou.
19:26.54VoIPMastawikityler: you're welcome
19:27.18[Airwolf]ManxPower, I get it.
19:27.52[Airwolf]Maybe there should be a function that just returns a true of false on the question if a channel is avalible.
19:28.05[Airwolf]Then the problem could be solved. :)
19:30.42mmmmmToophi...anyone here used Sirrix BRI cards before?
19:36.09*** join/#asterisk mindwarp (i=mindwarp@silenceisdefeat.org)
19:40.05mindwarpHello, I have a quick Asterisk question: I read a lot in tutorials about using SIP (which most IP phones use), but also about its problems with NAT. I was wondering if Asterisk allows me to use one protocol (SIP) for connecting my phone to the network and a different protocol (IAX) to carry that line over the net for use with an ITSP or such.
19:40.53mindwarpor do I have to commit to one protocol?  Any clarifications appreciated, thanks!
19:41.52xbmodder_lappyyou can use both IAX and SIP
19:42.56mindwarpSo I can use a SIP phone but actually use IAX to terminate calls etc.  That's great, thanks.
19:47.50[Airwolf]mindwarp, no, then you have to buy a phone that supports IAX2
19:48.31xbmodder_lappy[Airwolf], I think he means having his asterisk box use both IAX2 and SIP
19:49.47mindwarpah, no, [Airwolf] got it
19:49.56mindwarpthanks, that's what I was wondering
19:50.25mindwarpwhether there is any way to have Asterisk bridge the two protocols
19:50.36Maxxeddamnit
19:50.47Maxxedim getting this callerid@ipaddr of the pbx now
19:51.01Maxxedthat i upgraded my 7960's to the lastest sip firmware
19:51.16Maxxedanybody know how to turn that crap off
19:51.50[hC]cisco sure is getting a kick out of fucking up sip loads on their phones these days
19:52.00*** join/#asterisk bweschke (n=bweschke@c-68-36-51-213.hsd1.nj.comcast.net)
19:52.07Maxxedlol
19:54.11techieMaxxed: I have the same problem
19:55.04Maxxedfigure out a resolution yet?
19:55.19techieno, no docs on it either
19:55.24Maxxedgayness
19:56.45*** join/#asterisk MacDome (n=eseidel@c-71-198-177-144.hsd1.ca.comcast.net)
19:58.42xbmodder_lappyany VoIP providers here?
20:04.18*** join/#asterisk brockj49464 (n=brockj49@63.87.56.236)
20:05.05firestrmanyone here use *@home?
20:05.16*** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com)
20:05.27firestrmxbmodder_lappy, talk to websae..
20:05.37xbmodder_lappyYeah, I already havve
20:05.38xbmodder_lappyhave
20:05.58firestrmhe has been really good to me..
20:07.25firestrmok no asterisk@home people huh.. any one know of any paid support?.. im about to loose my sanity on this thing.. and even though i cant afford it, im still willing to pay to get it fixed..
20:08.47NewSolewhats wrong
20:10.06*** join/#asterisk hohum (i=corbe@snoop.burghcom.com)
20:10.42NewSoleo well.... I am going for some food
20:11.40hohumhow can I set up my asterisk box so it only does RTP pass thru?
20:11.44firestrmNewSole, i keep getting "the number you called is temporarly disconencted" when dialing outbound
20:12.05firestrmeven though the connection is going through..
20:14.04firestrmand the #)$(#*$)@#($* thing doesnt give any usefull debug information.. i have no idea why it is winding up there.,.
20:17.07*** join/#asterisk Dr-Linux (n=Linux@202.59.73.131)
20:17.16*** join/#asterisk DoktorGreg (n=Greg@70.91.121.89)
20:17.58RoyK~seen wasim
20:18.08jbotwasim <n=wasim@pdpc/supporter/active/wasim> was last seen on IRC in channel #asterisk, 4d 6h 16m 8s ago, saying: 'guigouz: an fxo/sip ata'.
20:18.51bonfire1Is there any way to use an old voice-modem as an FXO card?
20:20.18Igbothom_IIInot if you want good quality
20:20.28*** join/#asterisk synaptic (i=synaptic@68.62.176.196)
20:21.01bonfire1well and if I just want to play with asterisk ?
20:21.19bonfire1is it just like - installing the modem and configuring something in the zaptel.conf ?
20:21.23hohumhow can I set up my asterisk box so it only does RTP pass thru?
20:21.56*** join/#asterisk stkn (n=foobar@gentoo/developer/pdpc.active.stkn)
20:22.06Igbothom_IIIbonfire1, not that easy - you need drivers for your modem.  best way would be to look on ebay for cheap 1-port FXO clone cards - they cost around US$10 - 15
20:22.22bonfire1yeah but I live in israel
20:22.25bonfire1shipping is a bitch
20:22.37bonfire1they sell the digium ones for 49$ (!!)
20:22.41Igbothom_IIIthen ask locally  :)
20:22.54*** join/#asterisk marv (n=marv@12-219-145-181.client.mchsi.com)
20:22.57Igbothom_IIIthen spend $49 and be done with it
20:23.01bonfire1there isn't much of a market for voip supply.
20:23.13[Airwolf]bonfire1, there are enough cheap pci modems availible that are just the X100P, but you need to chage the driver a little bit.
20:23.54Igbothom_IIIafter all, the clone cards are basically a modem themselves
20:24.05synapticyou just need to change the drivers to the chipset of the winmodem to make it a x100p heh
20:24.50bonfire1so I basically plug a modem in, install winmodem's driver, and I'm good to go?
20:25.41Igbothom_IIIif you get the right modem, yes
20:26.38synaptici rather just buy a dev kit.   tdm11
20:26.57Igbothom_IIIas I suggested, get cards you know will work
20:27.03Igbothom_IIIsaves the buggerising around
20:27.17bonfire1maybe I'll do that... in my next trip to the US...
20:27.41bonfire1or with a company that supplies a US address and then ships it to Israel
20:31.15Igbothom_IIIlike the guys who supply you with weapons  :)
20:31.32Igbothom_IIIthey generally ship straight from the 'States  :)
20:32.37bonfire1yeah US army stuff is great
20:32.47[Airwolf]hmm, 2 nukes, 3 stingers, 1 X100P, 200 granades
20:32.49[Airwolf]:P
20:33.56hohumhow can I set up my asterisk box so it only does RTP pass thru?
20:34.23`Kevinasterisk should be able to become a slave via pri to a shortel ? we have this scenario and the d channel will not come up
20:35.27*** join/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
20:35.32*** part/#asterisk surfdue (n=tyler@unaffiliated/surfdue)
20:35.50brookshirehohum: by default
20:36.27brookshirenoreinvte=yes (would make all traffic go through asterisk and not between endpoints)
20:37.31brookshiremaybe it's canreinvite=no
20:37.33distortioncanreinvite=no
20:37.38brookshirei forget
20:38.01distortioncanreinvite=no forces rtp to pass through asterisk
20:38.34brookshirertp pass thru is kind of a bad question though
20:38.51brookshirebecause pass through simply means asterisk isn't doing anything with transcoding
20:39.30distortionrtp passthrough is needed to maintain billing acuracy
20:39.51distortioni've had horrible billing issues by turning it off when connecting to session controllers
20:40.04distortionerr by using canreinvite=yes rather
20:40.17filein Asterisk?
20:40.24distortionyep.
20:40.36distortionparticularly bad when connecting to nextones
20:40.45filethat makes no sense
20:40.50brookshireit doesn't make sense
20:40.58brookshirebecause the control should still be up and monitoring
20:41.04filereinvites might not work, so your audio might not flow... but signalling still goes through Asterisk
20:41.17hohumwell this asterisk box is being set up as a signaling gateway
20:41.24Maxxedok im out for now fellas
20:41.25ManxPowerYeah.  reinvites only does AUDIO.
20:41.25hohumneed to take SIP and spit out H323 to a vendor
20:41.27hohumso
20:41.30distortionfor some reason there are always hung calls on the provider's end which cause billing issues
20:41.31Maxxedil harass yall later ;)
20:41.33hohumit doesn't need to touch the RTP stream
20:41.49filethen device might not like reinvites for whatever reason
20:41.52brookshireh323 = ewh!
20:41.53brookshire:)
20:41.56fileer the device
20:42.11distortionnextones are very good with re-invites
20:42.26fileit still makes no sense that it would affect billing
20:42.54fileoh well
20:43.19distortioni'd love to find out too- then i could stop spending thousands on my bandwidth bill since i proxy all the damn rtp
20:43.26*** join/#asterisk swm_ (n=admin@digitaldatabits.net)
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20:51.07*** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
20:55.15ghenryWhat do you guys think of this: http://cgi.ebay.co.uk/Asterisk-PBX-Hardware-and-Full-Support_W0QQitemZ9716930873QQcategoryZ34165QQrdZ1QQcmdZViewItem ?
20:55.20ghenryRecommended?
20:55.25ghenryOr have go myself?
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20:59.43*** part/#asterisk ghenry (n=ghenry@mail.suretecsystems.com)
21:00.50lesouvage.
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21:05.34*** part/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-251.claranet.co.uk)
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21:25.28NoName32hi all i am still pretty new to asterisk got a problem i cant figure out .. working on the one touch record using the featuremap with automon => *1 but it doesnt seem to be reconizing the *1 if i change it to ** it works any ideas/ sugestions i am using asterisk 1.2.6
21:31.14*** join/#asterisk apardo (n=apardo@87.217.144.163)
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22:17.39Drukenwhat is it about icecream that no matter what, it makes ya feel better?
22:20.16SpaceBassanyone using freedigts.com?
22:20.29SpaceBassfreedigits that is
22:22.12*** join/#asterisk saftsack (n=saftsack@p54A7E109.dip.t-dialin.net)
22:23.34Drukenlooks like a virtual peering with upgradeable dids
22:23.58SpaceBassknow what it is, but just having problems routing my incoming numbers
22:24.46Drukenoh, wuts the problem?
22:24.54SpaceBassthey come in as guest sip calls
22:25.03SpaceBasswith out a DID
22:25.24Drukenreally... thats strange....
22:25.35SpaceBassso I can have all guest calls go to my incoming context but I'd really prefer to treat each one differently
22:26.11Drukenare the seperate accounts per did? or multipul dids per account?
22:26.24SpaceBassseperate accounts
22:26.40*** part/#asterisk opus_ (n=opus@dahphish.org)
22:26.56Drukenalrighty
22:28.10Drukenare you registering?
22:28.35SpaceBassno, nerdvittles.com suggested that you could, and I tried their settings, but it didnt work
22:28.49SpaceBassfreedigits's tech support said to just forward the sip calls
22:29.01SpaceBassso thats why they are coming in as guest calls
22:29.53Drukenforward the sip calls? how so ?
22:29.58*** join/#asterisk Aurs (n=Aurs@host-81-191-123-189.bluecom.no)
22:30.27SpaceBasslike I put in the IP of my * box and the port (5060)
22:34.42IceManRISKAnyone here uses JIAX ?
22:35.27X-RobSpaceBass, if you care about calls from a specific guest, they should have a type=peer, which means they'r eno longer a guest.
22:35.27DrukenSpaceBass: you got dialing with them working?
22:35.36X-Roba guest means, by definiton, something you don't know about and dont' care about.
22:36.52*** join/#asterisk saftsack (n=saftsack@p54A7E109.dip.t-dialin.net)
22:38.04Drukeni just grabbed a number from them, not sure if it's working, i'd need a test call... if it works, then i'll let ya know how to fix it :)
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22:45.24*** join/#asterisk DoktorGreg (n=Greg@70.91.121.89)
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22:52.04*** join/#asterisk chiardon (n=chiardon@200.71.58.39)
22:52.27chiardonHi
22:53.41chiardonI have an E1 connected to an * box, yet my E1 provider seems to be signalling me with PCM-31.
22:53.55Qwellchiardon: Have them change it..
22:53.57chiardonStrangely enough the link sometimes works
22:54.00NuggetMy cat's breath smells like cat food.
22:54.21chiardonQwell --> Yes, but in the meantime (they say that takes about 1 week)
22:54.29chiardonis there anything that can be done?
22:54.36Qwellchiardon: You could pay somebody to add support to *
22:54.41Qwell(which would take more than a week)
22:54.52QwellOr, you could have tested before you went into production. :)
22:55.31chiardonThere is a BRI patch around ... could that work for my pri?
22:55.41Qwellno
22:56.27DoktorGregBRIStuff does not work on PRI
22:56.31*** join/#asterisk robust (n=robust@c-567472d5.01-167-70697410.cust.bredbandsbolaget.se)
22:56.36DoktorGregin no way shape or form
22:57.13chiardonOh well thanks.
22:58.14chiardonBtw I am looking for a CHEAP SIP (physical) phone
22:58.32DoktorGregget an ATA
22:58.36QwellCHEAP as in crap, or cheap as in inexpensive?
22:58.43DoktorGreglol
22:59.00Qwellgrandstream for the former, linksys spa941 for the latter
22:59.07chiardonhow crappy can 'crap' be?
22:59.17Qwelltwo words...
22:59.20Qwellbarbie tones
22:59.55chiardonand in the inexpensive range... we are talking about how much?
23:00.03Qwell$100?
23:00.07Qwell$150 tops
23:00.13chiardonfor one?
23:00.18Qwellyes
23:00.23*** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au)
23:00.24Qwelltelephony aint cheap
23:00.35chiardonUuuh I think I joined the wrong channel ... you guys are expensive here :P
23:00.47DoktorGregATA is also option
23:00.47Qwellchiardon: Try #nortel.
23:00.58QwellYou'll wet yourself when you hear what a nortel phone costs
23:01.14chiardonDoktorGreg --> How much can a decent ATA cost?
23:01.21Qwelldecent?  ~$90
23:01.25DoktorGregI am contemplating selling all my nortel equiptment and replaceing it with the inexpensive voip stuff
23:01.31SplasPoodQwell: Is there even a #nortel? :)
23:01.35QwellSplasPood: maybe :p
23:01.42Qwellprobably on something silly, like dalnet
23:02.01chiardonDoktorGreg, --> What nortel thingies are you selling?
23:02.04SplasPoodHeh, that just sounds so funny being said on freeload
23:02.12DoktorGregplaying with idea
23:02.17DoktorGregI have about 40 phones
23:02.29DoktorGregplus a MICS system with all the bells and whisles
23:02.34chiardonDoktorGreg, --> And I hope you're selling them at a barbie-toned price :P
23:02.46QwellDoktorGreg: Only $25k for the base, and $300 for the phones?  used?
23:02.52Qwellif so, good deal
23:03.01DoktorGregThats what i was thinking
23:03.22chiardonas I said before when you guys talk about money I get scared
23:03.29chiardonso I think I'll just run home to mommy =)
23:03.33chiardonHave a nice day
23:03.40chiardonGood bye =)
23:03.46SplasPoodhahahaa
23:03.49SplasPoodthat was hillarious.
23:03.52DoktorGregwhat about softphone?
23:03.56DoktorGregoh he left
23:04.04SplasPoodit was hilarious too
23:04.19DoktorGregwell a nice analog desk phone still cost ~80-90
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23:06.29project_2501is it possible to chang the ${EXTEN} channel variable?
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23:31.19Ariel_hello eveyone
23:31.28Ariel_eveyone/everyone
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23:55.33camonzhi
23:55.55camonzi was wondering if i could get some help registering * with a sip proxy
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23:57.21camonzi'm doing register => name@201.211.65.2/1006
23:57.27camonzin my sip.conf
23:57.56camonzi sucesfully register but i'm not getting any calls from that server
23:58.30tainted-maybe no one is calling
23:58.37tainted-try sip debug
23:58.46tainted-which sip proxy
23:58.46camonzthey where calling
23:58.56tainted-are u behind a nat
23:58.59camonzit's a setup between some friends and i, we're testing it now
23:59.03tainted-what does 'sip show peers' say
23:59.09camonznope.., i'm in the dmz of my lan
23:59.17*** part/#asterisk demigod2k (n=joey@cpe-24-210-97-162.twmi.res.rr.com)
23:59.27tainted-ok
23:59.28camonzplus i've got correctly configured externip and localnet
23:59.39tainted-what does sip debug say
23:59.52tainted-when a call is placed through the sip proxy

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