irclog2html for #asterisk on 20040801

00:27.23*** join/#asterisk jero (~boo@modemcable055.101-131-66.mc.videotron.ca)
00:27.29jerohi
00:40.18lwcdo any of the FXO cards work of line power from the phone line, even if the PC dies calls process?
00:40.33tzangerlwc um no
00:40.38dougheckahah
00:40.46lwcthere is one card that does
00:40.54lwcgetting specs
00:40.58dougheckabut, what would answer the phone?
00:41.03lwc911
00:41.08tzangerlwc use a $50 UPS and save yourself the headache
00:41.10dougheckaI mean, the software needs to be there to answer the call
00:41.11tzangerlwc that's called passthrough
00:41.14tzangerthe card isn't doing shit
00:41.41dougheckaah, then you mean the FXS/FXO card
00:41.45dougheckaaka TDM400 :)
00:41.50dougheckabut it doesnt do it
00:49.10*** join/#asterisk jmhunter (~jacob@adsl-68-123-41-113.dsl.pltn13.pacbell.net)
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00:55.37jmhuntermds
00:55.45jmhunterls
00:55.46jmhunterls
00:55.49jmhunterls
00:55.51jmhunteroops
01:09.28letholanyone with voicetronix hardware experience?
01:10.34letholasterisk cli goes on some kind of loop when asking for outside line.. and when incoming call gets picked up by asterisk
01:13.29*** join/#asterisk hcir (~hcir@64.4.231.24)
01:15.11*** part/#asterisk hcir (~hcir@64.4.231.24)
01:26.24*** join/#asterisk jr99 (trilluser@adsl-065-005-202-014.sip.gnv.bellsouth.net)
01:34.33*** part/#asterisk jr99 (trilluser@adsl-065-005-202-014.sip.gnv.bellsouth.net)
01:36.58Legend`so this sayson 480i is interesting
01:36.58*** join/#asterisk jero (~boo@modemcable055.101-131-66.mc.videotron.ca)
01:37.00Legend`you can't send DTMF if you are conferenced
01:37.09Legend`rather embarrasing
01:37.21Legend`"could you please dial these DTMF tones pls?"
01:38.22*** join/#asterisk ellery (~ellery@65.3.112.225)
01:38.45*** join/#asterisk Dave`` (dave@ool-4352e324.dyn.optonline.net)
01:46.59DrRighteouscan someone explain the point of a context statement within a peer entry in sip.conf?
01:47.08DrRighteousoviosuly in a user entry its for inbound routing. but in a peer statement?
01:47.19PatrickDKoutbout routing
01:47.23*** join/#asterisk implicit (~implicit@ip68-7-154-222.sd.sd.cox.net)
01:47.23PatrickDKoutbound
01:47.28implicitsup
01:47.31DrRighteouspls explain.
01:47.35PatrickDKspecify ip address and password, and ....
01:47.42PatrickDKvalid codec's to use
01:47.51PatrickDKsame shit as inbound
01:47.57DrRighteousIt limits the context which can call a Dial commant with the SIP entry?
01:48.18PatrickDKoh, e, context
01:48.36PatrickDKprobably only used when pressing transfer
01:48.41PatrickDKif it is used than
01:48.52DrRighteousvery confusing...
01:49.03DrRighteousnot sure it answers my questions :)
01:54.41letholdamn.. what whould be the normal CLI output when a vpb answers?
01:54.58letholCLI output goes into some kind of loop
02:03.47*** join/#asterisk coppice (~Steve_Und@135.195.17.210.dyn.pacific.net.hk)
02:05.07*** join/#asterisk jmhunter (~jacob@adsl-68-124-65-96.dsl.pltn13.pacbell.net)
02:05.07*** mode/#asterisk [+o jmhunter] by ChanServ
02:06.09jmhuntermy qos is rockin
02:06.34jmhuntercant get wshaper to work on my wrt command right... buit i figured out hwo to tweak the web ui
02:09.24jmhunterit is dead in here today
02:10.49Legend`this isn't #wrt54g
02:10.50Legend`;D
02:10.56jmhunterbuzz off
02:11.02jmhunterVOIP
02:11.04jmhunterqos
02:11.06twistedwheee
02:11.07twistedgoing
02:11.09twistedgone
02:11.11jmhunteris there a #wrt54g?
02:11.14Legend`yeah
02:11.32Legend`jmhunter: btw, just posted my feedback of the 480i on the -users list
02:12.16jmhunterwhat do u use for a router legend
02:12.31Legend`jmhunter: at home?
02:12.39jmhuntersi
02:12.48Legend`some little SMC thing, its a few years old
02:12.56jmhuntersweet
02:13.02jmhunterwhere r u ?
02:13.06Legend`have a WRT54G under my desk at work, waiting to be molested, but haven't had the time
02:13.10Legend`jmhunter: bahamas
02:13.16jmhuntersweeet
02:15.35jmhunterlegend on the redial thing.. maybe u have to setup standard dialout lengths in the phone's dialplan (not asterisks)
02:17.04deddLegend`: nice indepth review
02:17.54jmhunterit is good
02:22.01jmhunterlegend r u from the states
02:32.26DrRighteousdo I need username/secret in a IAX user context if there is a register string?
02:37.33*** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc)
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02:41.20Ahewesanyone found that T1 crossover cable that is *too* short will not work with their X100P?
02:41.33*** join/#asterisk MustDie (~voip@ool-18b91fa7.dyn.optonline.net)
02:42.13AhewesI just had a weird X100P experience where a properly configured 1' cable between an X100P and an Adtran did not work, and a 6' cable *did* work.
02:42.20coppiceAhewes: depends what you set in the zaptel.conf file for the line buildout
02:42.25AhewesThe 1' cable was picking up all kinds of spurious stuff.
02:42.40AhewesWell, I used the shortest buildout for both.
02:43.01Aheweslets see...span=1,0,0,esf,b8zs
02:43.53Ahewesanyway, just an odd thing.  I'm using cat 5, so a 1' cat 5 cable might be slightly out of spec.  I think cat 3 is recommended.
02:45.56atacommlol, i keep watching that JibJab video thats on Slashdot
02:47.21letholwhat could be wrong with xlite/asterisk setup if other end cannot hear me?
02:50.18bkw_ok who in here told someone that type=friend is BAD BAD BAD?
02:50.41bkw_because who ever you are.... YOU'RE A FUCKING IDIOT.  99% of the time type=friend will do the job and do it well.
02:50.46bkw_NEXT!!!
02:51.04PatrickDKhmm, 600ohm wire is recommanded, cat5 is 100ohm
02:51.40Corydon76-homebkw_:  It's a habit of JerJer to tell people that type=friend is bad
02:52.18coppicebkw_: and that attitude has nothing to do with configuring * :-)
02:52.51letholatacomm: it is a great video
02:53.14coppice"You can't say nuclear. That really scares me" :-)
02:53.28atacommlethol: yeah it is
02:54.17Corydon76-homeGeneralizations are generally wrong... including this one...
02:54.23bkw_haha
02:54.51atacommi like "Mass-Oh-Chew-Sits"
02:54.54bkw_well dont confuse people... friend is fine 99% of the time.
02:55.05bkw_the way asterisk does sip peer/users confuses people sometimes
02:55.16coppicefriend is fine 99%, but lover is best all the time :-)
02:55.46Corydon76-homeFriend is good, but fuck-buddy is even better
02:55.51bkw_hahahaha
02:55.58atacommi also like how they make fun of Fox News
02:56.01coppiceatacomm: I thought the traditional bad spelling was the dental version - massive-chew-sets
02:56.10atacommLOL
02:56.50kramyes, friend is bad in SIP
02:56.56krambecause of SIP's broken authentication model
02:57.07postelIs there a way to tell my cisco ATAs 186 to stfu and dont flood my wires with CDP requests?
02:57.21kramyou don't have separation of callerid and authentication
02:57.24visik7what's the user:pass of sourceforge cvs ?
02:57.45Corydon76-homeanonymous:<none>
02:58.03visik7thanks
02:59.43visik7Corydon76-home there is a common path ?
02:59.54Corydon76-homevisik7: nope
02:59.54visik7cvs -d :pserver:anonymous@cvs.sourceforge.net:/usr/cvsroot/sipp login
03:00.15Corydon76-homeEvery repository has its own directory
03:00.41atacommwhy doesnt' digium have a web cvs?  i use a 3rd party site, but it would seem like digium should have that
03:01.39implicitatacomm: because they have only a little bw
03:01.49implicitand they think you should be able to handle command line
03:02.03implicitand i think your name is fucking hilarious
03:02.05blitzrageyah.. get used to the CLI with Asterisk :)
03:02.15atacommyeah, if you want to download it...... if you want to quickly look at a file, web cvs is nice
03:02.24implicitatacomm: i agree
03:02.30implicitbut it doesn't change your name
03:02.31Corydon76-homeOr run your own CVS mirror
03:02.44implicitCorydon-home: he barely has a computer, lol
03:03.08atacommpff, we own like 10 servers, implicit probably does IRC off a calculator
03:03.12implicitlol
03:03.22implicitatacomm: no joke, it runs linux
03:04.12implicityea
03:04.21implicityour name is pretty chill just funy
03:05.26implicitbbl
03:05.34atacommoh please done come back
03:06.09DrRighteousanyone have a good/cheap source for a QoS router for the home? like a WRT?
03:06.23atacommlinksys makes one what has QoS....never tested it though
03:06.44DrRighteoushey atacomm you have IAXy's in stock?
03:07.18atacommnope, we've never been told by digium that they are generally available.... (of course, on that regard, Digium doesn't seem to ever keep resellers up to date, we usually find things out from customers first)
03:07.32DrRighteouseak!
03:08.40*** join/#asterisk |Vulture| (~Vulture2@adsl-154-193-8.jax.bellsouth.net)
03:09.02atacommthey provide us about as much documentation as they do with their products
03:10.09coppiceatacomm: excellent promotional technique you have :-)
03:10.21atacommlol, what do you mean?
03:11.04coppiceatacomm: it sounds a bit like "we sell stuff that sucks". Did you ever see the rise and fall of Reginald Perrin"?
03:11.15atacommlol, no i havent
03:11.40atacommalthough, i would have to partially agree with that statement...... I think Digium could do alot more to please their customers and not act like a basement operation, especially on $1500 cards
03:12.10coppiceFamous British TV comedy series. He bacame very rich running a chain of stores called Grot, that sold utterly useless things. Bill Gates might have used him as a role model.
03:14.10*** join/#asterisk angler- (~angler@angler.digium.sponsor.pdpc)
03:14.44atacommcoppice: i mean, even the one sheet thats included with the TE410P....i looked at it the other day, it wasn't even printed straight.... they photocopied it
03:16.23*** join/#asterisk |Blaze| (dirc@d142-59-242-112.abhsia.telus.net)
03:16.35coppiceyeah. A sheet or two, with maybe the odd picture, might make it look like they are a more substantial operation.
03:17.16atacommwell i mean, the parts on the card are sub-$100, but the card costs $1500.  The least they could do is provide documentation, maybe a free CD with asterisk, i dunno, doesn't seem like a whole lot to me
03:18.10coppicein large quantities the card would be $100. Do Digium have that volume, though?
03:18.21bkw_ok ok
03:18.30Umarobkw_, !!!!!
03:18.38Umarobkw_, did you get around to fixing app_controlplayback?
03:18.43bkw_no
03:18.45bkw_i'm lazy
03:18.48atacommwe've priced the parts out for cards we are working on, so yes, in qty of 100, it would be sub-100 for them
03:18.56bkw_I have no idea why its doing it
03:19.06bkw_Umaro open  bug and attach that bt output please
03:19.33Umarobkw_, ok
03:21.14file[laptop]magical like
03:21.24blitzragelool
03:21.28blitzrage-o
03:24.48bkw_poose menis
03:26.41file[laptop]well said!
03:27.10Corydon76-homeHmmm, didn't take me all that long to set up cvsweb
03:27.59|Vulture|moose!
03:28.15file[laptop]|Vulture|: half credit on that
03:28.42|Vulture|esoom sinep!
03:29.02file[laptop]-100% credit
03:29.10*** join/#asterisk orignal (~chatzilla@CPE0040f43f9688-CM013319902665.cpe.net.cable.rogers.com)
03:29.13|Vulture|lol
03:29.34kentsterevening all
03:30.13coppiceit should be evening some, morning some, afternoon some
03:30.27|Vulture|yea, and worktime for all
03:30.48file[laptop]jbot: ugt?
03:30.49jbotwell, ugt is Universal Greeting Time. It states that it is always morning when person comes in, and it is always late when person leaves.  Local time of any member of channel is irrelevant.
03:30.51file[laptop]:)
03:31.32coppicejbot: you just made that up, didn't you :-)
03:31.33jbotcoppice: KCI error, or a problem with the Keyboard-Chair Interface.
03:32.52|Vulture|hahahah
03:34.01*** join/#asterisk jmhunter (~jacob@adsl-68-123-40-240.dsl.pltn13.pacbell.net)
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03:34.10jmhunterhey everyone
03:34.25jmhunterso i pulled out my x100p card yesterday, removed all of its configuration...
03:34.37jmhunterbut now i get this error on my zap fxs line
03:34.47jmhunterARNING[1201788480]: chan_zap.c:1137 reset_conf: Failed to reset conferencing on channel 1!
03:34.51jmhunterWARNING*
03:35.13jmhuntereverytime that zap chan disconnects it gives thate error
03:38.40*** join/#asterisk mitcheloc (trilluser@69-169-62-35.anhmca.adelphia.net)
03:38.48mitchelocoing the change now
03:42.03jmhunteri canr
03:42.05jmhuntercant
03:48.01|Vulture|what do you put in zapa.conf so that it doesn't answer ZAP calls? I just want to use it for outgoing
03:48.27jmhunteri make a context that answers it and hangs up
03:48.44|Vulture|oh good idea
03:48.57jmhuntergoto(killer,s,1)
03:50.03MustDieJul 31 22:02:10 WARNING[49167]: chan_zap.c:1137 reset_conf: Failed to reset conferencing on channel 1!
03:50.09MustDiefresh checkout
03:51.18*** join/#asterisk adker (adker@216.130.231.1)
03:54.22*** join/#asterisk _tekati_ (~captain@cpe-66-75-215-63.bak.rr.com)
03:57.49|Vulture|strange the ZAP line is set to auto forward calls to the voip line, but the * box detects that it rings and opens the ZAP device
03:58.13DrRighteousbye all
03:58.48*** part/#asterisk DrRighteous (~poll49@ool-435717de.dyn.optonline.net)
03:59.36*** join/#asterisk ariel_ (~Ariel@fl-nked-ubr2-c6c-4.miamfl.adelphia.net)
03:59.52ariel_Hello all
04:01.10*** join/#asterisk mitcheloc (trilluser@69-169-62-35.anhmca.adelphia.net)
04:01.30bkw_http://tinyurl.com/3zo2y
04:01.40jmhunterwhat is it
04:02.00dougheckaquad 15 inch display
04:02.06dougheckaONLY 2400 dollars
04:02.16|Vulture|nice
04:02.18|Vulture|not bad
04:04.33ariel_Actually for 2400.00 you can get 4 17" LCD's and put them together and still have money left over. But it would be nice in any case.
04:05.07dougheckayea
04:05.52coppice$250 buys a 15" display, so $2400 is a bit high for adding a mounting bracket.
04:05.59dougheckalol
04:06.22coppiceWhy would anyone put 4 x 15" together, when 4 x 17" would be so much nicer at a modest extra cost
04:06.23|Vulture|yea we got our $15 for $100 with the free upgrade program through dell
04:06.42|Vulture|they had buy a PC and get a free upgrade to LCD from CRT
04:06.58dougheckahah
04:07.29coppiceViewsonic's 3 x 17" side by side unit is quite nice
04:07.53|Vulture|I have a 17" hercules I love it, great for digital photography
04:08.24dougheckaI read that as "I have a 17" hercules I love it, great for digital pornography"
04:08.38|Vulture|hahahaha
04:08.41coppiceshh. that's the real use
04:08.46dougheckaoh
04:08.46|Vulture|no....
04:08.47doughecka:P
04:08.50|Vulture|thats what the laptop is for
04:13.44*** join/#asterisk E-Mind (E-Mind@c-24-6-226-189.client.comcast.net)
04:14.56E-Minddoes asterisk support cygwin?
04:16.06coppiceno
04:16.29coppicethough I think there is work going on with cygwin
04:16.47coppiceprobably not with the drivers for the cards, though
04:17.06E-Mind:-/
04:17.35E-Mindhow about knoppix?
04:17.50dougheckaknoppix should work, as its linux
04:17.53ariel_file tried and gave up. He had part of it working.
04:18.03coppicewriting drivers from windows is deeply masochistic
04:18.14doughecka~dict masochistic
04:18.23doughecka....
04:18.57ariel_there are some that is the way they get there kicks....
04:19.07dougheckaheh
04:19.09dougheckain the rear
04:19.24E-Mindhmmm... I have a shuttle XPC
04:19.35E-MindI wonder if the built-in soundcard is supported
04:19.51JohnWayneanyone use 800today.com for the toll free numbers?
04:19.57coppiceIf linux supports it, * will
04:20.12coppicesome sound cards are pretty nasty for latency, though
04:20.14implicitohaosdyf
04:20.16implicitahfohafoihaf
04:20.18implicitf
04:20.19dougheckaamen
04:20.21implicitgoing insane
04:20.25E-MindI also have a SB audigy
04:20.25dougheckagood
04:20.30dougheckawelcome to the club
04:20.50E-Mindhaha
04:21.28coppiceif being insane is good enough from the president of the USA, its good enough for you
04:21.56E-Mindso, I need a modem to use the software or can I use the soundcard?
04:22.53*** join/#asterisk cuban (~djimenez@border0-hou.cuban.cc)
04:23.06ariel_E-Mind, modem, sound card.  Wow, You don't really need any of them. It really depends on what your going to do with it.
04:23.15cubanah0y
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04:23.48ariel_E-Mind, you do need an timing device or if your connecting to your pstn line a fxo port like a x101p.
04:24.01ariel_cuban, hello hope all is well.
04:24.13E-Mindthanks ariel
04:24.24*** part/#asterisk nowork (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com)
04:24.25E-Mindare you from Israel?
04:24.38jmhuntersup cuban
04:24.41cubanyo
04:25.02jmhunteri am sooo tired
04:25.19ariel_E-Mind, no I am from the good old USA.
04:25.27*** join/#asterisk nowork (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com)
04:28.47ariel_Ok everyone here where should I start to learn to program. Via PHP or Perl first or directly into C programming? I really need to start learning this stuff.
04:30.27jmhunterAnypne besides me having braodvoice issues?
04:30.47|Vulture|jmhunter: they just started a second ago
04:30.55|Vulture|ringing into nowhere
04:31.04|Vulture|showing registered
04:31.22jmhunterdo a busy forward to a nufone line.. in is still working for me, i cant make calls out
04:32.33JohnWaynecan I have two simultaneous but unique Dial statements in a context?
04:32.57JohnWayneWhen a call comes in I want to set a specific caller id name for one set of extensions, but not for another
04:33.03JohnWaynebut I want them all to ring at the same time
04:33.18E-MindAriel - start with Kernigan & Ritchie (Ansi C)
04:33.45ariel_E-Mind, thanks I will check them out.
04:35.30E-Mindhttp://www.amazon.com/exec/obidos/tg/detail/-/0131103628/102-8147261-2654533?v=glance
04:36.12*** join/#asterisk denon (proxy@synapse.subneural.net)
04:36.30|Vulture|jmhunter: my call forwards aren't working :(
04:36.53E-MindIs there a Knoppix distro with asterisk built in?
04:37.58E-Mindis it still possible to spoof CID/ANI with asterisk?
04:38.08|Vulture|depends on your voip service
04:38.20E-Mindglophone?
04:38.35implicitE-Mind: lol
04:38.37E-MindI use a web based voip
04:38.37|Vulture|duno about them
04:38.41coppiceE-Mind: the spoof issue is not a * problem. Its a PSTN operational problem
04:38.47implicite-mind what does web based voip mean?
04:38.48implicitlol
04:38.54E-Mindwell..
04:38.56E-Mindhaha
04:39.13|Vulture|is that like broadvoice's little java app?
04:39.23E-MindI know... it sounded funny... what I meant was that I am not using any sort of hardware besides my PC
04:39.30|Vulture|jmhunter: resource unavil. from BV's website
04:39.34E-MindI don't know what the rest of you are using
04:39.38|Vulture|softphones
04:39.48|Vulture|thats what its called
04:39.50E-Mindif you sign up for any special service providers or what not
04:40.09E-MindI just found this voip provider that I use on my computer with a headset
04:40.10|Vulture|I believe nufone and vpc pass CID but Im not sure
04:40.23E-Mindhttp://securityfocus.com/news/9061
04:40.36E-MindNatas used Asterisk in conjunction with the NuFone Network for his demonstration of Caller I.D. unmasking. NuFone chief Jeremy McNamara didn't return phone calls for this story.
04:40.54implicite-mind lol
04:42.24|Vulture|hahaha "hack"
04:42.41|Vulture|wtf they call everyone a hacker now
04:42.53implicityeah man
04:42.56|Vulture|people who cheat on video games are "hackers"
04:42.56implicitit is pretty retarded
04:42.58ariel_E-Mind, Funny in that NuFone will allow you to set your own callid but if you check out most PRI providers will allow you to do the same thing.
04:42.59JohnWayneE-Mind: you will always be able to set your number on outgoing calls over a PRI or CT1 because of the laws that require specific location data to be transmitted to 911 PSAPs for facilities that are over 50,000ft^2
04:43.15implicitariel_: basically all pri providers, hehe
04:43.31E-Mindhaha
04:43.46E-Mindyah... the design is kinda stupid
04:43.55E-Mindthere is no handshaking
04:44.02|Vulture|urg and now BV is totally fucked
04:44.04E-Mindno authentication
04:44.06ariel_implicit, not all nuVox will not allow that. They want papers on all the numbers you use and names..
04:44.17coppicein most countries you cannot spoof your CID
04:44.39implicitariel_: yeah there are a few who dont
04:44.51JohnWaynetraditionally, it hasn't been for spoofing, but just for what I just said, but now with VOIP people are using to spoof
04:45.33coppiceJohnWayne: it has been used for spoofing. that is why most countries stopped it
04:45.33|Vulture|yea but were all "hackers" because we use *
04:45.56ariel_What we use it for is multi company installations. Like my last job we had 3 different companies own by same people but used same network and phone system. Asterisk was perfect for them.
04:46.02JohnWaynecoppice: but spoofing hasn;t been a huge problem until people could get $8/mo VoIP lines
04:46.05|Vulture|great... implicit you know what this means now right?
04:46.13JohnWaynecoppice: before you had to pay $500+/mo for a PRI
04:46.35|Vulture|were going to get a bunch of 12 year olds that want to make prank calls trying to use * and come in here looking for help
04:46.42JohnWaynehehe
04:46.50JohnWayneexaclty, before now 12 year olds didn't have PRIs
04:46.54coppiceJohnWayne: I think it was spoofing by call centres that provoked most places to stop it
04:46.56E-MindI guess the definition of a hacker is in the eye of the beholder
04:47.06ariel_rofl 12 year olds. we get that with the over 20 already....
04:47.08mtplol
04:47.17|Vulture|hahaha
04:47.20mtpi'm 17
04:47.29mtpand yeah at PRI
04:47.35ariel_young wipper snapper....
04:47.43JohnWaynecoppice: laws in the US have changed to require telemarketers to send ANI information, they used to not send any to prevent people from knowing who they were or blocking their number
04:47.47|Vulture|its good to learn young
04:47.50mtpwork pays for the PRIs there
04:47.57|Vulture|Im 21
04:48.01mtpbut i'm root@{santa-fe,espanola}.pbx
04:48.08mtpwhich both have PRI
04:48.17coppiceJohnWayne: 5% of people live in the US. Most others have CLI spoofing stopped at the local exchange
04:48.20ariel_boy 2 that are in the same age group as my kids.
04:48.56JohnWaynecoppice: how do other countries handle sending specific locationd data for large facilities to the PSAP?  do they have the telco handle it?
04:49.57JohnWaynethe original idea was if you had a large factory or something, you would send phone numbers that correspond to your specific location in thet factory.  then if someone calls 911, the operator will see "acem factory, building 12, floor 2, room 3" instead of just "acme factory"
04:50.03coppiceJohnWayne: If you send no CLI a default for your PRI gets inserted at the local exchange. If you give one which you are not entitled to it depends on the operator. Some reject your call. Some change the CLI to the default.
04:50.04E-MindFunny thing I noticed is that when my parents came to visit from Israel - they had a roaming GSM phone - and while I usually get weired numbers on my caller-ID when they call me - I now saw their complete (Israeli) number
04:50.14ariel_mtp espanola.... your spanish?
04:50.20|Vulture|can someone try 314-989-9997?
04:50.26mtpariel_: new mexican
04:50.31JohnWaynecoppice: emergency operators will reject your call?
04:50.56JohnWaynecoppice: oh I see, the telco allows you to send only the block you are assigned
04:51.10ariel_mtp, what is a new mexican.
04:51.17mtpariel_: new mexico, USA.
04:51.20mtpyou know
04:51.20JohnWaynecoppice: yes, I've seen some that do that here, and it doesn't affect legitimate users, only hackers
04:51.21mtpthat state
04:51.26coppiceIf you place an invalid PRI call it gets rejected. Where is the problem in that. Anything else wrong in your call setup will have the same effect.
04:51.27mtpit's in between arizona
04:51.29mtpand texas
04:51.38|Vulture|someone that is not an old mexican :P
04:51.42ariel_Oh yes the one by what is it called Tejas.
04:51.46mtpys.
04:51.48mtpi live there
04:52.00JohnWayneyou live in New Mexico?
04:52.26ariel_I forgot that there is a New Mexico in the States. I must be getting old.
04:52.52|Vulture|lol, aren't you in miami ariel_?
04:53.00coppiceGood place for a suntan
04:53.25ariel_|Vulture|, yes I am. Actually 16 miles south west of Miami.
04:53.45|Vulture|I like lauderdale
04:53.56|Vulture|its always work when Im in miami
04:54.08ariel_I might be moving soon. I am looking for a new job.
04:54.23|Vulture|Im looking to kill broadvoice...
04:54.33JohnWaynewhy?
04:54.41|Vulture|they are messed up right now
04:54.52JohnWayneI am waiting for them to be able to port numbers in my area so I can switch most of my lines to them
04:54.56|Vulture|getting a busy tone on inbound and they aren't forwarded me calls
04:55.09|Vulture|they are good when they are working but this is the second problem in a week
04:55.17|Vulture|hopefully this one doesn't take a day to resolve
04:55.36|Vulture|ariel_: can you pull more than one line on that?
04:55.43JohnWayneor I want voicepulse to offer LNP
04:56.36|Vulture|didn't know if you could just keep opening new ZAP channels... didn't know if it worked that way
04:57.33JohnWayneso, back to my original question:  when a call comes in, can I ring extensions 1/2/3 with the caller id as sent, but send a special caller id name to extensions 4/5/6?
04:57.41ariel_|Vulture|, Packit8 will turn on the 2nd port if you want to.
04:57.54|Vulture|oh nice
04:57.56ariel_JohnWayne, yes
04:58.27JohnWayneariel_: would I have just three priority 1 commands? or would I ring them with no timeout and then use a Wait command?
04:58.32|Vulture|Im going to bed and will wish to the magic fairies that BV comes back up
04:58.32ariel_JohnWayne, asterisk has great power via the extensions.conf and the dial macro's
05:00.12ariel_JohnWayne, there are many different ways you can asign a caller ID to zap ports, to sip connections or even they them if you want to use xxx dial 9 first if you want ZZZ dial 8 first and so on.
05:00.16mtpJohnWayne: YES I LIVE IN NEW MEXICO
05:00.23JohnWaynentp: sorry
05:00.23mtplike i just said
05:00.24mtptwice
05:00.24JohnWayne:)
05:00.52JohnWayneariel_: yeah, that's not really what I'm asking, but I'll figure it out
05:01.44ariel_JohnWayne, Ok but if you tell us more we might come up with a way. I was just being general in my reply's.
05:02.42JohnWayne<JohnWayne> so, back to my original question:  when a call comes in, can I ring extensions 1/2/3 with the caller id as sent, but send a special caller id name to extensions 4/5/6?
05:02.48JohnWayneI thought that was pretty specific
05:04.28ariel_not really is 1/2/3 going to be ringing 4/5/6 at the same time. or is the 123 inbond and 456 outgoing. or are they in there own groups?
05:05.07JohnWayneyes, I want them all to ring at the same time
05:05.08JohnWayneBUT
05:05.17JohnWaynei want the caller as sent to appear on 1/2/3
05:05.28JohnWayneand I want a string that I specify to be sent to 4/5/6
05:05.52JohnWaynethis is so single line cordless phones can know which line the call is coming in one
05:05.53JohnWayneon
05:06.13JohnWayneif it comes in on line 1, I'd like to send "LINE 1" as the CallerIDName
05:06.20JohnWaynebut JUST to the cordlesses
05:06.32JohnWaynethe deskphones have multiple line appearances, so it will be obvious
05:07.18ariel_JohnWayne, Ok I see. That will take some doing in the extensions.conf For this one I will have to think about it and put a flow chart together.
05:07.41JohnWayneI think I have it almost worked out
05:09.49ariel_ok well it's late anyway and I need to go to bed. Wife is calling.
05:10.10JohnWayneok
05:10.45*** join/#asterisk pfn_ (~3fc50337@pfnguyen.best.vwh.net)
05:11.12pfn_man, this screen is so bright
05:12.10jmhunterpfn are u currently suffering the wrath of bv like the rest of us?
05:13.33kramlol!
05:13.41*** join/#asterisk advorak (~advorak@cn8.ischool.washington.edu)
05:13.56jmhunterwhat is it funny that im straight?
05:14.19coppiceits actually a jealous induced giggle
05:14.20jmhuntertheres still a few of us around hhere
05:14.43coppicei'm straight, but wifey is 10,000km away :-(
05:15.45coppicemost things in the US are big, but baths are small. don't americans like a nice friendly bath? :-)
05:15.59kramlol
05:16.04kramno it's just funny
05:18.06pfn_coppice yep, my fiancee is 8000mi away
05:18.10pfn_.....
05:18.27pfn_as for whether or not broadvoice is up for me, I don't know
05:18.32pfn_I'm not at home
05:18.37pfn_ircing via cgiirc
05:19.33jmhuntersweeet
05:19.45jmhunterwaiting for my wife to find shorts
05:19.49jmhunterfor me
05:19.57pfn_you don't know where your shorts are?  :p
05:20.01pfn_mmmm, hot tub action
05:20.04coppiceclothes? how unromantic!
05:21.03*** join/#asterisk jorist (hidden-use@62-177-191-186.bbeyond.nl)
05:22.06pfn_indeed
05:26.40|Vulture|jmhunter: still having BV probs?
05:27.46*** join/#asterisk brc_ (~root@ip24-251-182-226.ph.ph.cox.net)
05:29.39*** join/#asterisk nmkha (~nmkhaus@203.210.217.198)
05:29.42brc_http://www.brc007.com/images/3.jpg
05:30.05Legend`bastard
05:30.11Legend`i have never gotten a good picture of fireworks
05:30.34coppicebrc_ fireworks, or a really bad hairstyle? :-)
05:31.09|Vulture|no tripod brc?
05:31.26|Vulture|or long exp?
05:31.29brc_it's 'artistic' boi
05:31.44brc_also try 2.jpg and 1
05:31.44izohey did any of u had problems with X100P that stopeed answering phone calls (reload of wcfxs module helps solve the problem)
05:32.06|Vulture|maybe a clone?
05:32.06coppiceI can see the new DisneyLand from here, so next year I will be able to see fireworks every night. How nice, eh? [answer: no, it isn't]
05:32.24Legend`izo: X100P doesn't use wcfxs
05:32.27Legend`maybe that is the problem
05:32.56*** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com)
05:32.57Legend`|Vulture|: there are fireworks about twice a month less than a mile from my house
05:32.58pfn_that sure is an ugly picture of fireworks
05:33.10izoi ment wcfxo
05:34.06*** join/#asterisk Bekar (bekar@darkwood.bekar.id.au)
05:34.20Bekarhowdy.
05:34.21kentsterany AGI masters here?
05:34.45BekarI was wondering if anybody could help me get a Voicetronix OpenLine 4 working with Asterisk.  I seem to be missing the part that tells Asterisk that one exists..
05:34.53|Vulture|oky I think 2 is cool, but I don't think you meant to do it lol
05:37.04kentsterI have the line "RECORD FILE test gsm 1 20  BEEP"  and when I hit one to exit recording it just beeps at me again.. it seems to be in a recording loop.  Is the syntax corrent?
05:37.33atacommwhats the basic trouble shooting procedure for moh?  i installed mpg123, edited musiconhold.conf, but its saying :
05:37.33atacomm-- Executing MusicOnHold("SIP/atacomm_1000-fffa", "default") in new stack
05:37.33atacommAug  1 00:36:07 WARNING[1251156800]: res_musiconhold.c:334 moh0_exec: Unable to start music on hold (class 'default') on channel SIP/atacomm_1000-fffa
05:37.59coppiceBekar: onky a few people use the Voicetronix cards with *. You might need to hunt a bit to find an expert.
05:39.29Bekarcoppice: that's what I was afraid of ;)
05:39.36|Vulture|shit
05:39.40|Vulture|anyone here a BV user?
05:40.12BekarI guess my biggest confusion is I can't see anwhere to tell * that it should use a given device as a channel.. :-/  I can't se eit loading the configuration file that's in /etc/asterisk/, or how to tell it to load a specific section..
05:41.04coppiceBekar: /etc/asterisk/vpb.conf would be a good starting point to look.
05:41.11|Vulture|urg I just called BV and they said they didn't know there were any problems right now
05:41.32Bekarcoppice: yea, configured that.  But can't see * as having looked at it.  There's no messages in the startup, even under -vvvvcg
05:41.56Bekarah well.  I guess I'll keep poking it.
05:42.34coppiceBekar: Does the vpb module get built by default?
05:42.55JohnWayneatacomm: musicclass = default in your sip.conf
05:43.41Bekarcoppice: I can't see one with *, unless it's not part of the 'asterisk' cvs tree.
05:43.44brc_http://brc007.com/images/fireworks.html
05:44.10coppiceBekar: chan_vpb is only built if vpbapi.h is on your machine. Is it?
05:44.37|Vulture|what camera brc_?
05:44.41Bekarchecking.
05:45.07coppicebrc_: where was the firework display? Looks like a fairly small affair
05:45.22brc_rawhide
05:45.27brc_phoenix,az
05:45.36brc_it's a local tourist trap that has been around FOREVER
05:45.39Bekaryes it is.. but it's possible it wasn't there when I compiled *.  Will try re-compiling, see if it does it.  Thanks.
05:46.04coppicebrc_: the summer lightning in pheonix is rather more spectacular :-)
05:46.12brc_yeah it is
05:46.20brc_we get some _nice_ sunsets
05:46.31brc_coppice you get down this way often?
05:46.39coppicebut the lightning is often silent. really weird
05:46.51brc_yeah
05:46.55brc_funny you mention that
05:46.59brc_we just had a nice storm
05:47.06brc_but my tripod is busted
05:47.13|Vulture|www.the-vulture.com/gallery
05:47.14brc_one of the legs sinks
05:47.16|Vulture|my gallery
05:47.41coppicebrc_ I spent a few weeks in pheonix about 12 years ago, when I worked for Motorola. az is a very beautiful place
05:48.11brc_my mom worked for motorola
05:48.21brc_EE
05:48.33coppicelots of people *used* to work for Motorola :-)
05:48.37brc_:p
05:48.55brc_did you hear they've started to tear down one of the plants here?
05:49.02brc_can't remember what it's called
05:49.12brc_what'd you do for moto
05:49.17*** join/#asterisk sudoer (~toy@pool-68-160-154-250.bos.east.verizon.net)
05:49.36coppiceI didn't hear, but it seems likely. They are China bound these days.
05:49.57JohnWaynetheir new radio products suck
05:49.58brc_yep
05:50.01coppiceDSP in the semi division. Speech coding mostly
05:50.03JohnWayneI sell motorola radios
05:50.04brc_and yep
05:50.10brc_frs?
05:50.15JohnWaynethe stuff they;ve come out with lately is horrible
05:50.30JohnWaynebrc_: CGISS
05:50.37coppiceits mostly OEM from HK and China companies
05:50.58Bekarcoppice: many thanks, that was it.
05:51.13coppicethe BOM is extremely low, so don't blame the Asians for the quality
05:51.17brc_JohnWayne, carry ham stuff?
05:51.30JohnWaynebrc_: sure do, I am a ham myself
05:51.37pfngoddamnit, speex fucking annoying
05:51.41brc_cool
05:51.44brc_I'm a tech
05:51.46brc_:\
05:51.48pfncalling x-lite with speex enabled fucks shit up
05:51.52coppicespeex is very good
05:51.52brc_never finished learning code
05:51.53pfnunless I install speex in asterisk...
05:51.53pfnugh
05:51.57brc_got about half way
05:52.06JohnWayneme, too. I've been meaning to upgrade for 10 years now :)
05:52.15brc_got my ticket in 98
05:52.23brc_or was it 97
05:52.26JohnWaynebrc_: shop.waltel.com is where all the goodies are, in case you were wondering
05:52.26brc_one of those two
05:52.30|Vulture|pfn: you use broadvoice?
05:52.30brc_okay
05:52.30JohnWayneI got mine in 94
05:53.00|Vulture|nvm it seems like its back up
05:53.03coppicebrc_: code? Unicode or Morse? :-)
05:53.10brc_morse
05:53.23coppicehow very un-i18n :-)
05:53.25brc_.... ..
05:53.37brc_#hamradio
05:53.42coppicemorse for Chinese is fun
05:53.45ddougghey new crowd
05:53.52ddougganyone interested in setting up my asterisk @home?
05:53.59JohnWaynebrc_: on efnet or freenode?
05:54.02brc_*giggle*
05:54.08ddouggi'm looking for someone wanting to work on a per-project fee basis...
05:54.10|Vulture|ddougg: just takes a few hrs to learn the basics
05:54.12brc_is there a larger croud on efnet?
05:54.14ddougghttp://aurora.telerama.com/as-setup.txt
05:54.24JohnWaynebrc_: I used to hang out in there, there was maybe 20-30 people
05:55.13brc_do you carry the orinoco pigtails?
05:55.55file[laptop]I usually charge per hour
05:55.59|Vulture|ddougg: get a FXO card, X100P card
05:56.25|Vulture|ddougg: get approx. 1ghz pc
05:56.59ddouggokay vulture, just dump the proper config files to me and we'll be done...
05:57.27JohnWayne* Inbound hunt   that's something you're gonna have to setup on the provider's end
05:57.34Legend`ddougg: the configs aren't hard, why don't you do it on your own?
05:57.56ddougggot other demands on my time
05:58.01ddouggmakes more sense to outsource things.
05:58.19JohnWaynedo you have the phones yet?
05:58.21|Vulture|ddougg: I can write that some time tommorow all you need is something to record the sound files
05:58.33|Vulture|then just a basic voice prompting
05:59.04ddouggall i've got so far is the grandstream fxs that broadvoice sends on with their basic package.
05:59.23|Vulture|hmm I don't know about that if no one answers and someone is there part
05:59.41|Vulture|duno if it can be done without trying to ring that person's extention you are checking
05:59.54JohnWayneyeah, I dunno about that
06:00.13|Vulture|ddougg: what phones are you using SIP enabled?
06:00.18JohnWayneyou'd need a "meta extension"
06:00.21ddouggasterisk can't figure out if a line is in use?
06:00.36|Vulture|ddougg: VoIP lines work differently
06:00.38brc_that part might take some magic
06:00.45ddouggi'll have to buy some stuff.
06:01.06brc_|Vulture| you can use qualify and iirc there is a manager event
06:01.32|Vulture|oh didn't think about using that never had a need for it
06:01.40brc_:)
06:01.46JohnWaynesounds like this project is getting more complicated
06:01.51|Vulture|yea
06:02.10|Vulture|well that stuff is but the basics are almost extentions.example.conf
06:02.14coppiceJohnWayne: how many projects get less complicated? :-)
06:02.18brc_haha
06:02.20|Vulture|so true
06:02.31JohnWayneI'd bid it if I didn't have 23523 other projects open at the moment
06:02.40|Vulture|Web interface for VM never used that but Im sure its not that hard if its a * module
06:02.54JohnWayneI tried the web interface, it was buggy and ugly
06:03.15|Vulture|JohnWayne: good to know... email works well
06:03.30JohnWayne|: yep, that's what I use if I'm near email
06:03.48JohnWayneI have it send a text messaage to my phone and send the message to my email
06:03.57|Vulture|oh my they brought back bevis and butthead on MTV
06:04.04JohnWaynereally?
06:04.09|Vulture|yup MTV2
06:04.12JohnWayneah fond memories
06:04.17JohnWayneI wonder if I get mtv2
06:04.25|Vulture|lol they are making fun of smashing pumpkins
06:04.34|Vulture|"today" I like that song...
06:04.57atacommso I'm curious, why does MOH pickup where previously was left off, i.e. middle of the song, thats like creepy...lol, you keep putting someone on hold and its like they havent missed any of their life inbetween being on hold
06:05.04brc_btw FANTASY #2 might be doable IF you use analog phones....or cisco7960's
06:05.11JohnWayneatacomm: it is always playing
06:05.24Legend`atacomm: if nobody is on hold, the mpg123 process is paused
06:05.34brc_night
06:05.35atacommLegend: AHHH, ok, that makes sense
06:05.39brc_nighteynight
06:05.47|Vulture|night
06:06.24|Vulture|yea #2 wouldn't be that hard with cisco 7960/40s
06:06.24Legend`atacomm: it is kinda annoying, if you don't have a high call volume, the same song can be playing for a week
06:06.29|Vulture|just make a callplan to redirect
06:06.51coppiceI used to work with someone who would break off a conversation when we parted and pick up at precisely the same point next time we met. No introduction. No "What we were talking about the other day". Really weird
06:06.54JohnWayne|Vulture|: I do that in my dialplan
06:07.11JohnWayne|Vulture|: the second and third choice trunk selection thing, right?
06:07.19|Vulture|yup
06:07.23JohnWayneyeah, that's easy
06:07.31JohnWayneusing n+101 priorities
06:07.46|Vulture|fun stuff lol
06:08.10JohnWayneexten => _NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
06:08.10JohnWayneexten => _NXXXXXX,102,SetCallerID(806-698-1346)
06:08.10JohnWayneexten => _NXXXXXX,103,Dial,IAX2/UdS47XfH13@voicepulse/1806${EXTEN}
06:08.13|Vulture|wait #2... join a call in progress
06:08.35|Vulture|wow that looks familliar JohnWayne
06:08.36JohnWaynetries to use TRUNK, which is Zap/g1, if that's busy then it uses voicepulse IAX trunks
06:08.54JohnWayne|Vulture|: oh yeah?
06:09.18ddouggpart of what's driving me to outsource it is that i haven't been able to find a definitve reference on asterisk conf files...
06:09.35JohnWayneyou been to the wiki?
06:09.38Legend`ddougg: voip-info.org
06:09.41JohnWayneit has 90% of what you need
06:09.43brc_~asterisk wiki
06:09.44jbotsomebody said asterisk wiki was http://www.voip-info.org . A wiki is a user-editable website
06:09.52brc_~useful asterisk docs
06:09.53jbotmethinks useful asterisk docs is (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the "Unnoficial Links") and http://www.voipinfo.org/wiki-Asterisk (the Wiki), and http://www.fnords.org/~eric/asterisk (ManxPower's site), and http://asteriskdocs.org, also, read all files in /usr/src/asterisk/doc
06:10.10brc_more like 98.284% I'd say
06:10.13|Vulture|voip-info is a massive source
06:10.37JohnWaynebrc_: some of the directives for config files have no links to click on for more info
06:10.42|Vulture|anyone see Napleon Dynamite?
06:11.01brc_ddougg see also: http://www.asteriskdocs.org
06:11.36JohnWaynein the time it took you to write that RFP you could have gotten a good start on programming your setup
06:13.16|Vulture|can anyone tell me if the first one is correct or the second one: http://www.pastebin.com/88043 or http://www.pastebin.com/88044
06:13.59|Vulture|yea good org. though all my dialplans start on a napkin or something throw away
06:15.23JohnWayneI don't think either of them are
06:15.34JohnWayneunless you are trying to dial out on multiple trunks at once
06:15.43|Vulture|no
06:15.48|Vulture|just trying to failover
06:15.58|Vulture|urg
06:15.58JohnWayneexten => 123,1,Dial(firstchoice)
06:15.59ddouggyeah, i've gone over the wiki...
06:16.04JohnWayneexten => 123,102,Dial(secondchoice)
06:16.11JohnWayneexten => 123,103,Dial(thirdchoice)
06:16.17|Vulture|oky
06:16.21JohnWaynelike that
06:18.28JohnWaynelet me rethink that
06:18.34JohnWaynestandby
06:18.40|Vulture|http://www.pastebin.com/88046
06:18.46|Vulture|that looks better
06:19.48JohnWaynehttp://www.pastebin.com/88047
06:20.22JohnWaynebecause on busy, it will go to n+101
06:20.23|Vulture|ah yes +101
06:20.29JohnWayne102+101 = 203
06:20.42*** join/#asterisk BoRiS (boris@S01060050da67299b.wp.shawcable.net)
06:20.47JohnWaynetry that
06:21.04Legend`that whole numbering system is so antiquated
06:21.10JohnWayneyeah
06:21.20Legend`there must be a better syntax for extensions.conf
06:21.33JohnWayneit can get hairy if you have some complicated failover or LCR situations
06:21.41Legend`yeah
06:21.45E-MindDoes * work in linux under vmware? (does the sound-card work ok?)
06:21.50|Vulture|I think I can get rid of the double BV
06:22.01Legend`even something as simple as basic line numbers, so you can slip in a line 15 if you need to
06:22.16Legend`and n+101 would have to be expressed some other way
06:23.08JohnWayne|Vulture|: oh yeah, you can
06:23.36JohnWayneI'd like to see it like BASIC where you number your lines 10 20 30 etc
06:23.36*** join/#asterisk ^ClaSsY (~^ClAsSy@203.175.66.68)
06:23.51Legend`JohnWayne: thats what i meant
06:24.05Legend`its not perfect, but its easier than renumbering 10 lines
06:24.12JohnWayneyeah
06:24.46JohnWayneor use an interactive system like nortel's BARS
06:25.04JohnWaynesomeone would have to rewrite the whole * configuration paradigm though
06:25.23|Vulture|JohnWayne: I think its what you said, but with 2,Hangup instead of 5,Hangup
06:25.44JohnWayneyeah, that makes more sense
06:26.33JohnWaynealthough it would probably work with 5
06:26.52JohnWayneI dunno is asterisk panics if it can't find n+1, or it just wants to see something >n
06:27.12JohnWaynewhich would open up the possibility of BASIC style line numbering
06:27.34JohnWayneexten => s,10,dosomething
06:27.37JohnWayneexten => s,20,dosomething
06:27.38JohnWayneetc
06:27.45JohnWaynethen it's easy to add something later
06:30.51atacommhow do you set a call queue to go to voicemail at a certain point?  i've got a timeout specified, but all it does is restart the timeout lol
06:31.12*** join/#asterisk Urgo (Feh45@pcp0010126810pcs.alxndr01.va.comcast.net)
06:31.55JohnWaynetheoretically, a call queue shouldn't go to VM unless the caller presses a number
06:32.12JohnWayneie "please continue to hold, or press 1 to leave a message"
06:32.34atacommJohn: in theory yes, but what if you wanted it to?  Otherwise it can be a pain to wait for ever, lol
06:32.58|Vulture|there we go all configured, no errors
06:33.19JohnWayneatacomm: I honestly don't know the answer
06:33.38*** join/#asterisk twisted (~twisted@twisted.active.supporter.pdpc)
06:33.38*** mode/#asterisk [+o twisted] by ChanServ
06:33.45atacommhow do you do extension time of day filtering? i cant remember
06:34.31|Vulture|GotoIfTime
06:34.53|Vulture|example GotoIfTime(8:00-17:00|mon-fri|*|*?open,s,1)
06:34.54*** join/#asterisk jmhunter (~jacob@adsl-68-123-41-202.dsl.pltn13.pacbell.net)
06:34.54*** mode/#asterisk [+o jmhunter] by ChanServ
06:34.59|Vulture|wb jmhunter
06:35.17|Vulture|I called BV and they had no clue it was not working... lol started working like 5min later
06:35.21jmhunterhye
06:35.27jmhunterhahha
06:35.32jmhuntertypical
06:35.44|Vulture|I think I would rather them tell me they were working on it
06:35.47jmhuntermosquitos took over my house while i was in the hottup
06:36.00|Vulture|need to get one of those high voltage zappers
06:36.14JohnWayneanyone use 800today.com for their toll free?
06:36.19JohnWayneor have another suggestion?
06:36.32jmhunternufone
06:36.38coppicejmhunter: you have shorts on, so at least you have some protection.
06:36.45JohnWaynehow much is nufone?
06:36.50coppicewe would be totally naked in similar circumstamces
06:36.50jmhunterno im ully clothed
06:36.56jmhunter2 cents a min john
06:37.01JohnWayneany MRC?
06:37.03coppicestill, wifey is a good mosquitoe attractant
06:37.04|Vulture|oh man a mosq. bite on the balls... that would ruin my day
06:37.05JohnWayneor NRC?
06:37.11*** join/#asterisk evilbuny (~evilbuny@192-172-93-202.dsl.nbdsl.net)
06:37.27jmhunterwhats that john
06:37.45JohnWayneMonthly Recurring Charge and Non-Recurring Charge
06:37.52jmhunternone at all
06:38.04jmhunterits prepaid, and ur prepaid ammount never expires
06:38.20JohnWaynecan you get me a specific number?
06:38.24jmhunterthey do iax
06:38.51jmhunterif u paypal 5 bucks to sales@nufone.net you should have an account tomorrow morning
06:38.59*** join/#asterisk Moc (~Moc@modemcable161.105-70-69.mc.videotron.ca)
06:39.02JohnWayneso I can have my tollfree come in on a IAX trunk? that would be nice
06:39.07file[laptop]yes
06:39.15jmhunteryes, u can order numbers with them... i think there is a $20 fee for vanity 800
06:39.16MocJohnWayne, nufone offer that service
06:39.17JohnWayneI was gonna have it go in on a Voicepulse DID, but that would be an extra step
06:39.31jmhunterMoc pay attention
06:39.35jmhunteralready done
06:39.35Moconly problem, those 1800 doesn't work from canada :(
06:39.36JohnWaynehehe
06:39.47JohnWaynethen you can call the regular number!
06:39.54jmhunterya joh for 800 nufone is the wya
06:40.05jmhuntergafachi plans on have tol free did in the near future
06:40.09JohnWaynedoes nufone do DIDs?
06:40.11|Vulture|night guys
06:40.17jmhunterwhere r u john
06:40.24jmhunternigh vulture
06:40.26JohnWayne806, LUBBOCK rate center
06:40.34JohnWayneI need my numbers ported
06:40.34jmhunterwhat state
06:40.36JohnWayneTexas
06:40.44jmhunterwhy would i know were lubbock is
06:41.11JohnWaynedo you know where new york or los angeles or houston is?
06:41.17jmhunterwell i dont think they do in tat area... id use connect.voicepulse.com, for ur local did through IAX.. and nufone for 866 DID in
06:41.20JohnWayneeveryone knows where Lubbock is!
06:41.27jmhunternooo
06:41.52jmhunteri know fwhere la sf and ny and houston is
06:42.52JohnWayneI need one "unlimited" service trunk from like VP or BV for my LD, and then just some ported DIDs for incoming
06:42.58JohnWayneI am trying to get rid of SBC
06:43.04jmhunter/is also
06:43.11Moceveryone hate SBC..
06:43.18jmhuntersbc can bite my cballs
06:43.25bkw_who doesn't hate SBC
06:43.31jmhunterexcept they have me on dsl contract until march
06:43.48JohnWaynethe problem is that VP is in my area, but doesn't do LNP.  BV isn't in my area but they do LNP
06:43.53jmhunterbkw might bit my balls if given the opportunity
06:43.56Legend`JohnWayne: is that a single?
06:44.07bonbon-homeif i want to test an agi script from the command line then is this possible? What does agi->readparse do?
06:44.08JohnWayneJames Jones at BV told me that they would be able to port my numbers late summer or early fall
06:44.18JohnWayneLegend`: is what a single?
06:44.41Legend`JohnWayne: sorry, that was for jmhunter
06:45.09JohnWayneoh ok
06:45.52JohnWayneI'd probably use VoicePulse connect for everything if they could port my numbers
06:46.11jmhunterlegend i like em all.. i really like that new one.... first straw
06:46.28JohnWayneI do about 2000 minutes of LD a month, so I'd like ap lan with unlimited LD, which I have now from SBC, but costs $60/mo
06:46.30jmhunterbv is inconsistent.. i have them
06:46.35jmhuntertheyre a contant gamble
06:46.43Legend`jmhunter: might need to check that out
06:46.59JohnWaynejmhunter: so you wouldn't reccommend them for my business lines?
06:47.24jmhunterNOOOOOO
06:47.30JohnWaynethe only thing I hate about BV and VP is that I don't want threeway, call waiting, voicemail, etc
06:47.33jmhunterwhat kinda business u run
06:47.42JohnWaynecommunications equipment sales and consulting
06:47.43jmhunteri wouldnt do too much of any business on voip
06:47.46JohnWaynewww.waltel.com
06:47.49JohnWaynehehe
06:48.01jmhunterinternet isnt consistent enough
06:48.07jmhunterx100p
06:48.31JohnWaynemy internet has been down a total of maybe 10 minutes this year
06:48.41JohnWayneof course, there's a lot between BV and me
06:48.54jmhunterwell if anyone bin between u and ur provider goes down ur sol
06:49.00jmhunterexactly
06:49.02JohnWaynesome providers will let you specify a backup number in case their stuff goes down
06:49.06jmhunterbv has been shitty
06:49.13jmhunterthey were just down for 2hrs
06:49.23jmhunterand didnt even know it until vulture called
06:49.27JohnWayneI don't care too much if they're saturday night at 10pm
06:49.34JohnWaynebut it was monday at 10am I would be PISSED
06:49.50JohnWaynemy business lives and dies on the phone
06:50.26jmhunterthey switched servers for asterisk to use last week and didnt tell anyone that they did
06:51.01JohnWayneyeah, from what I've gleamed from the mailing list, they seem to do stuff and then not tell anyone
06:51.15JohnWayneand then claim that it's been working all along when it hasn't been
06:51.20JohnWaynemuch like a shady ISP
06:51.37JohnWayneVoIP providers are the dial up ISPs of the decade
06:51.42JohnWaynenew decade
07:00.44atacommanyone know how to use call queue agent penalties?  its mentioned but not documented
07:14.10jmhunterdont get me wrong johnwayne i pay for the service i get
07:14.25jmhunteri pay 9.99 a month for unlimited instate calling.. california is a big state
07:14.31jmhunter30 some odd areacodes
07:14.47jmhunterim off to bed
07:14.49jmhunternight
07:15.28atacommok, how the heck do i get a call queue to terminate on timeout?  i've got the options set yet it keeps ringing and ringing and ringing
07:26.21*** join/#asterisk snewpy (~markl@203-217-39-140.dyn.iinet.net.au)
07:46.04*** join/#asterisk Asger (~asgerstru@0x3ef37c86.hrnxx2.adsl.tele.dk)
07:56.46*** join/#asterisk Jas_Williams (~Jason@host217-43-100-163.range217-43.btcentralplus.com)
08:10.43*** join/#asterisk eightman (~karbon@wbar9.lax1-4-11-193-072.dsl-verizon.net)
08:42.00*** join/#asterisk markstripes (~stripes@book.riviera.org.uk)
08:47.34*** join/#asterisk cursor (~kevin@andromeda.office.cursor.biz)
08:49.15cursorhello all
09:10.12pfnso why does asterisk come with speex?
09:10.20pfnwhere does the license conflict
09:10.41cursor<pfn> so why does asterisk come with speex?
09:10.46pfns/does/doesn't
09:10.47cursorAsterisk does not come with SpeeX
09:11.32cursorSpeeX is easy enough to install
09:13.13cursorGet the SpeeX source and untar it somewhere
09:13.17cursorconfigure
09:13.19cursormake
09:13.20cursormake install
09:13.23*** part/#asterisk jorist (hidden-use@62-177-191-186.bbeyond.nl)
09:13.28cursorthen go to the Asterisk source and compile
09:13.46cursorit'll spot speex.h in /usr/include or /usr/local/include
09:13.53cursorand will make chan_speex.so for you
09:19.43*** join/#asterisk RaffiOl (RaffiOl@dynadsl-080-228-75-091.ewetel.net)
09:21.25cursorzzzz
09:33.02cursorOnly 19 reports left in the bug tracker
09:33.07cursorand most have patches attached
09:49.09*** join/#asterisk af_ (~af@62.94.148.227)
09:55.17twistedcursor, wtf are you smoking?
09:55.36twistedthere are 87 open items
09:55.59cursoroh - I had it restricted to 'core' :-)
09:56.00cursoroops
09:56.03twistedheh
09:56.17cursorI thought we were "getting somewhere" :-)
09:56.22twisteddude
09:56.27twistedif there were only 19 items i'd be afraid
09:56.34cursorI was amazed
09:56.37cursorfor 15 mins
09:56.38cursor:-)
09:56.58twistedhell
09:56.59cursorNow I'm back to normal
09:57.00twistedlet's face it
09:57.07twistedif there were only 19 items, i'd be out a job :P
09:57.16cursorhaha
09:57.33twisted(it's a non-paying job, but a job none the less)
09:57.57cursoryes
09:58.14cursor1 - 50 / 1581
09:58.15cursorack
09:58.22cursorSlightly more than 19
09:58.22twistedlol
09:58.28twistedput the check in the resolved box.
09:58.42cursoroh
09:58.54cursorbetter
09:58.57cursorclosed loads of bugs
09:58.59cursoronly 88 left
09:59.03twisted88?
09:59.08twistedthere were 87 like 5 minutes ago
09:59.12cursor1 - 50 / 88
09:59.19cursorprobably me
09:59.42cursorI've been clearing out my "annoyances" wishlist
10:00.18cursorThat transfer bug was the worst annoyance of all
10:00.45cursorI looked into it a couple of weeks ago
10:00.47cursorand gave up
10:01.13cursorI looked at it again today and almost gave up
10:01.21cursorI was going to raise that report and then give up on it
10:01.32cursorBut I just couldn't stop myself :-)
10:01.53cursorMainly because it was #1 on my list
10:03.48*** join/#asterisk altamic (~altamic@212.141.97.189)
10:04.29cursorIt looks as if "Asterisk does not hang up SIP call" is the new bug that crept in while you weren't looking
10:04.34cursorNot one of mine
10:04.57cursorYou can close 0002189
10:05.05cursoralready fixed in CVS
10:06.01cursorbrbwt...
10:07.36twistedcursor, there's a reason 2189 hasn't been closed yet.
10:08.48twistedalso, there hasn't been a related change in cvs since that bug was posted.
10:09.45*** join/#asterisk kiso79 (~kiso@81.198.225.172)
10:10.16kiso79Hi all
10:10.29twistedcursor, nevermind
10:12.48*** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
10:14.00twistedwhen i start overlooking cvs commits, it's time to sleep
10:14.19twistedg'nite
10:18.32*** part/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
10:18.50cursor<twisted> also, there hasn't been a related change in cvs since that bug was posted.
10:18.53cursorthere has
10:18.57cursorI'll find it
10:19.54cursorModified Files:
10:19.54cursormodule.h
10:19.54cursorLog Message:
10:19.54cursorDon't hard code the RTP payload type to 101 (bug #2192)
10:20.16cursorThe person just didn't qute the bug number
10:20.23cursorquote the correct bug number
10:25.55*** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
10:29.42cursorzzzz
10:44.15*** join/#asterisk Mjkeay (~m@ns2.uk.lastminutehosting.com)
10:49.46*** join/#asterisk Dobaj (~ryanair@avonstreet.plus.com)
10:50.36cursorIt's a bit quiet in here today
10:51.59cursorah well
10:52.03cursorI tried
10:52.57cursor:-)
10:53.20Mjkeaycanned! :P
10:54.55miller7~blast cursor
10:55.04miller7~fire cursor
10:55.09jbotI'm sorry, cursor, but it just isn't working out.  Pack up your stuff.
10:55.17cursorhaha
10:55.20miller7ooops
10:55.27miller7~nuke cursor
10:55.29miller7wtf
10:56.05miller7sorry man, no brain cells left
10:56.13cursorI know the feeling
10:56.23miller7yep too much wild life lately
10:56.35miller7ie sitting 20 hours in front of pc
10:56.43cursor:-)
10:56.53cursorOnly 20?
10:56.57cursorLight weight
10:56.59cursorhaha
10:57.25miller7:)
11:02.50visik7when I try to connect to asterisk via sip it says SIP/2.0 404 Not Found
11:03.51cursorgreat
11:03.52cursor:-)
11:04.06visik7?
11:04.09cursorYou'll probably have to provide more info
11:05.07cursorFor instance...
11:05.14cursorhow are you trying to connect?
11:05.17cursorwhat phone?
11:05.25cursorwhat does your asterisk config look like?
11:05.27cursoretc.
11:05.45cursorWhat have you tried so far to correct it
11:05.53cursorand what difference did your changes make
11:05.58cursorall these things help
11:06.05cursorIt's sometimes hard to guess
11:06.53cursorI suspect that you probably got the hostname/IP wrong in the client config
11:07.00cursoror the port number etc.
11:08.20cursoror perhaps it's connecting but can't find the context you specified
11:08.28cursorI have no idea at this point
11:08.34cursorextra info is always a bonus
11:17.11sudhir492Anyone using queues here?
11:17.15cursornot me
11:17.21cursorbut I've played with them a bit
11:17.49visik7I'm unsing SJPhone
11:18.00sudhir492is there a way to ring phones without agents logging in stayin off hook?
11:18.21cursor(visik7) ok - can you check your sjphone's config against your Asterisk config
11:18.26sudhir492Or logging in but not staying off hook?
11:18.35cursorstart by checking the hostname, port, username and password
11:18.36visik7cursor sure
11:18.43visik7all it's ok
11:18.58cursor(sudhir) you don't need agents to have queues
11:19.05cursoryou can specify the phones in queues.conf
11:19.18cursormember => SIP/2101
11:19.18cursormember => SIP/2111
11:19.20cursorlike that
11:19.48sudhir492cursor: really? After I have defined queues in queues.conf, how to put calls in a queue?
11:20.21cursorQueue(somename)
11:20.21cursorexten => 1234,1,Queue(...
11:20.45visik7cursor do you want read sip debug log ?
11:21.01cursornot at this point :-)
11:21.11visik7:/
11:21.27cursoractually, yes - post it to the pastebin and I'll take a look
11:21.39cursorhttp://pastebin.ca/new.php
11:21.47sudhir492cursor: I have at present defined as member => Agent/@4
11:21.57cursorok
11:22.05cursorthen that user will have to sign in as that agent
11:22.15sudhir492How do I specify everyone in call group 4 to be in the queue?
11:22.26cursorUnless the users move around or share phones, you may as well put the phones in the config
11:22.30cursorrather than agents
11:22.41cursoryou don't
11:22.47cursoryou specify the phones
11:22.51cursornot groups etc.
11:23.15sudhir492cursor: ok, specifying the phones is a good idea. let me try that.
11:23.50visik7sip
11:23.53sudhir492cursor: thanks
11:24.45visik7cursor sipdebug is the title
11:24.51cursor(visik7) can you pastebin your sip.conf block for one of the phones
11:25.02cursoredit the password out first
11:25.17cursorI think you may just need to add "nat=no" to the config
11:25.32cursoroops
11:25.36cursorI mean nat=yes :-)
11:27.05cursorhttp://pastebin.ca/389  <-- my Cisco phone
11:29.16visik7cursor sipconf
11:29.42cursorok
11:29.53cursorcompare that to mine and experement with the directives that are missing
11:30.00cursorespecially the "nat"
11:30.16cursorand remove that defaultip
11:30.23visik7nat ?
11:30.32cursorhttp://pastebin.ca/389
11:30.45visik7there is no nat between client and asterisk
11:30.53cursorright
11:31.06cursorBut you're on a private network, yes?
11:31.10visik7yes
11:31.12cursorYour Asterisk and your phone
11:31.14cursorok
11:31.23cursorso you want it to stay that way
11:31.46cursorremove the defaultip
11:31.48cursoradd the nat
11:31.49cursorrestart
11:31.54cursorand see if that makes a change
11:32.16visik7ok
11:32.18visik7I try
11:33.11visik7same
11:33.22cursorDid you restart Asterisk?
11:33.27visik7reload
11:33.39cursorstop and start
11:33.49visik7same thing
11:33.52cursorok
11:34.08cursorI just never trust anything other than a stop/start ;-)
11:34.17visik7:)
11:34.29cursornext
11:34.33cursoradd the following:
11:34.38cursordisallow = all
11:34.41cursorallow = ulaw
11:35.01visik7only this ?
11:35.09cursorfor now
11:36.01*** join/#asterisk HiTech69 (~hitech@34-29.202-68.tampabay.rr.com)
11:36.24visik7no way
11:36.26visik7404
11:36.38cursorok
11:37.24cursorcan you strip all passwords from your sip.conf and pastebin the entire file
11:37.32cursormake sure you strip all of the passwords
11:37.45cursorincluding any in your register directives (if any)
11:38.18cursorI really just want to look at your [general]
11:38.29cursorI assume you've been editing the sip block for your phone
11:39.39*** join/#asterisk inspired (mikael@a217-118-63-4.bluecom.no)
11:39.53visik7cursor /391
11:40.39cursorDoes your machine have more than one IP address?
11:40.44visik7yes
11:40.48cursorok
11:41.12cursorAdd this to the [general]
11:41.12cursoroh
11:41.12cursorare you using a late CVS version?
11:41.19cursoror a very old version?
11:42.07visik7CVS-05/31/04-22:00:51
11:42.11visik7is it old ?
11:42.21cursorquite old, but it may be ok for this
11:42.29cursorupdate it later today :-)
11:42.31cursorlocalnet = 192.168.0.0/255.255.255.0
11:42.36cursorAdd that to [general]
11:42.54visik7it's a packaged version
11:42.57visik7anyway
11:43.02cursoryuck
11:43.05cursorpackaged CVS version
11:43.24cursorJust get it from CVS - it's much easier than fighting with packages
11:43.36cursorand waiting for others to catch up with the latest bug fixes
11:43.44cursorok
11:43.49tzangercursor: I pull my CVS on a build machine and use checkinstall to make packages for my two * machines, neither of which have any kind of dev environment on them
11:44.07cursorearlier versions (I forget how old) required localnet to use two directives
11:44.12cursorone for the net and one for the mask
11:44.21cursorlater versions allowed them to be combined
11:44.31cursorI forget the variable names for the old method
11:45.04cursor(ztanger) ok
11:45.08visik7Invalid localnet keyword: 192.168.0.0/255.255.255.0
11:45.15cursorok
11:45.17*** join/#asterisk nextime (~nextime@danex.i-m-c.it)
11:45.18cursorold version then
11:45.40cursorAnd I've forgotten the old keywords
11:45.43cursorit was that long ago
11:46.27visik7got it
11:46.31cursorok
11:46.33visik7localnet and localmask
11:46.38cursorright
11:46.49cursortry that
11:47.54visik7no
11:47.58visik7same 404
11:48.00cursorok
11:48.01visik7:/
11:49.49cursorDo you have nat=yes in your soft phone?
11:50.27*** join/#asterisk altamic (~Michele@212.141.103.240)
11:50.40cursorI think you may have to dump that soft phone and download x-lite or something
11:51.04cursoror try connecting a real phone
11:51.20visik7a real phone ? I don't have a real sip phone
11:51.35cursorok
11:51.52cursorback in a moment
11:51.59cursorcan you post a new sip debug trace
11:52.02cursorbrb
11:52.17visik7sure
11:53.24visik7/392
12:00.45cursorYou don't have nat=yes
12:01.29visik7I ensure you that nat=yes is there
12:01.42cursorSending to 192.168.0.5 : 5060 (non-NAT)
12:01.55visik7this is the client
12:02.08cursorin the server
12:02.21cursorput nat=yes in the [test] block in your sip.conf
12:02.25visik7is there
12:02.32cursorok
12:02.44visik7look /391
12:02.49cursorand you restarted?
12:03.02visik7sure
12:03.21cursorI don't have a clue then - I suggest that you upgrade your Asterisk version
12:03.42visik7mmm I think the same
12:05.28cursorApart from the IDs, your old and new sip debug traces are identical
12:05.44cursorAre you sure you're changing the sip.conf
12:05.49cursorand not something else
12:05.55cursorlike another sip.conf somewhere else
12:06.00cursorin /tmp or something
12:06.03*** join/#asterisk kiso79 (~kiso@81.198.225.172)
12:06.09visik7:)
12:06.12visik7yes I'm sure
12:06.20cursorhttp://pastebin.ca/388
12:06.22cursororiginal
12:06.27cursorhttp://pastebin.ca/392
12:06.36cursornew after apparently making changes
12:06.48cursorload them into your browser and press forward/back to compare
12:06.56cursorthey are identical except for the IDs
12:07.08cursorEven the codec list
12:07.13cursoroh
12:07.26cursordoes your soft phone only allow alaw
12:07.27cursor?
12:07.28sudhir492visik7: I see you have been struggling for a while. Can you take out the password bits, and email me sip.conf and extensions.conf files? I should be able to help you out. sudhir@cequip.com
12:07.35cursorwhich codecs does it do?
12:07.53cursorhe's pastebinned them
12:07.57cursorthe sip.conf parts
12:08.14visik7sudhir492 I try to upgrade asterisk
12:08.28cursorhttp://pastebin.ca/391   <-- his sip.conf
12:08.59cursoroh
12:09.04cursoradd this to your [test]
12:09.11visik7maybe this cvs version has some problem with sip
12:09.23visik7and we're loosing time
12:09.24cursornever mind
12:09.30sudhir492doesnt matter. I have 4 different versions of asterisks running and the latest is going to be in production tomorrow - with one of th emost recent versions
12:09.36cursorI was going to suggest something silly
12:09.37cursor:-)
12:09.51cursortomorrow?
12:09.57cursorThat's cutting it a bit fine
12:10.11cursorYou'll need some time to get used to it before committing to supporting a live server
12:10.47cursorTry the latest CVS
12:10.49sudhir492cursor: I have been testing this one for a week. Call volume is no problem so far.
12:10.53cursorand try my sip.conf entry
12:10.54cursorhttp://pastebin.ca/389
12:11.25cursorok
12:11.39cursorSorry, I'm getting you two confused now :-)
12:11.51cursorremembering which one is where :-)
12:12.30cursorWhat sort of call volume do you have?
12:12.40cursorand on what hardware?
12:12.52visik7i4l
12:13.02visik7is what you mean ?
12:13.34cursorno, but never mind
12:14.01cursorYou need to get the latest CVS source compiled and installed
12:14.18cursorThen you'll have a better chance of getting it working
12:14.35cursorI have no idea what that soft phone is though
12:14.39cursorI don't have it here
12:16.41sudhir492cursor: around 1000 calls a day (9:00 to 6:00), Pentium 2.8GHz, 1GB Mem
12:17.05visik7sudhir492 how many concurrent call you can handle ?
12:17.56sudhir492visik7: PSTN interface is T1, and 23 simultaneous calls go fine.
12:19.17sudhir492G729 codec, no problems at all.
12:19.31cursorHow many licenses?
12:19.54visik7license for what ?
12:19.58cursorG.729
12:20.06sudhir49223 G729 licenses.SIP to SIP calls have no transcoding, hence they can be even larger
12:20.17cursorok
12:21.10miller7sudhir492: how many minutes are those 1000 calls?
12:22.23sudhir492miller7: On an average, around 5 minutes.
12:22.52sudhir492I will have better statistics in a week
12:24.01sudhir492miller7: I meant 5 minutes per call.
12:24.30miller7so you're not utilizing the T1 too much
12:25.30sudhir492Actually, most of the calls are on T1. In peak time, 20 incoming calls is not abnormal
12:25.44miller7ic
12:26.21sudhir4925000 minutes of incoming call on a T1 in 8 hours is pretty good utilization
12:27.02tzangersudhir492: that's under 50% utilization
12:27.27miller7sudhir492: depends on your peaks actually
12:28.45sudhir49210% utilization is good. 50% utilization is great. Have no choice but to go for T1 because of peak number of calls
12:28.46coppicefor 8 hours its a bit low, but I expect you have a couple of really peaky hours if you get near to 23 channels in use
12:29.03*** join/#asterisk suma (~suma@81-86-93-203.dsl.pipex.com)
12:29.16coppicewhat is your target grade of service?
12:30.08coppicetzanger: trying to hook it into mrtg, or similar?
12:30.10sudhir492coppice: I sure want it be perfect :-)
12:30.13tzangercoppice: exactly
12:30.26tzangerI guess catting /proc/zap/1 and parsing it every min or 5 would do
12:30.34coppicesudhir492: that isn't engineering
12:32.10sudhir492coppice: I know. Had no choice but to go for it.
12:33.02sudhir492I have a backup server, identical configuration
12:33.08coppiceI don't follow. You mean you given a pipe, and have no choice about its size?
12:33.42tzangercoppice what did you decide to do re your post early last month about revamping the fax subsystem?
12:33.57sudhir492coppice: Are we talking about two different things? What are you asking about?
12:33.59*** join/#asterisk zotz (~zotz@24.231.36.159)
12:34.07coppiceeh? what? what revamping?
12:34.25coppicesudhir492: do you understand what grade of service means?
12:34.45tzangercoppice: you'd made a post to -users about what to do about fax, whether to try and make a t38 channel for asterisk or to interface with hylafax or whatever else was discussed
12:35.31cursorjust use eFax
12:35.40cursorMuch less effort
12:35.51sudhir492coppice: I am not that pedantic about grade of service. I just go by subjective measure of reliability. However, will sure appreciate if you explained grade of service
12:35.52tzangercursor: I'm talking about incoming more than outgoing
12:35.58cursorso am I
12:36.06cursoreFax == fax->email
12:36.20tzangercursor: We already have a well-esatablished fax number
12:36.28cursorok
12:36.42cursorThen I'd get a fax modem and tie it to a serial port on a server
12:36.50cursorand run a fax server from there
12:36.52coppiceOh, that's not a revamp. Its what else goes in. T.38 will be going in, but now much has happened so far. A class 1 fax modem interface is in progress, but I'm not clear of the best way to present the modem interface with Unix98 pty interface in linux 2.6
12:36.58tzangerI'd like ot take a crack at writing a telnet<->pstty driver so I can hook pu my Ascend Max to Hylafax
12:37.53tzangerthe max will throw you on a modem if you telnet ot port 5000...  so I want to open up 24 connections to port 5000 and have HylaFax see them as serial ports
12:38.06tzangerthere are lots of serial port to telnet gateways but none quite do that :-)
12:38.39coppicesudhir492: you will find subjective judgements turn out to be utterly useless. Grade of service is the percentage of calls resulting in busy trunks. Try a google search with words like "erlang" anf "grade of service" and you should find some simple equations that will tell you what grade of service to expect.
12:39.20coppicetzanger: do those connections look like AT modems?
12:39.26tzangercoppice: exactly like at modems
12:40.15coppicetzanger: then it should be simple. You face the same issue as me, though. Unix98 ptys do not give  the kind of stable predictable device names that things like HylaFAX expect
12:40.56sudhir492coppice: Thanks. Thats what I was doing just now :-) As I told, in peak times, (like Monday morning, around 20 incoming calls is not uncommon) but mostly it is quite sustanined at 5 simultaneous calls
12:41.45coppiceif 20 is not uncommon, hitting the limit is not rare either. Do the calculation, and work out what you can really expect.
12:42.01tzangercoppice: indeed, I just haven't sat down and done it yet
12:42.09tzangercoppice: how do other apps that require stable device names cope?
12:42.17cursorIf I'm really unlucky, I'll get two calls in a day
12:42.26cursorOn a normal day, I'll get none
12:42.28cursor:-)
12:42.33*** join/#asterisk |^Angel^| (~angel@port453.ds1-alb.adsl.cybercity.dk)
12:42.38cursorI tend to leave my phone on DnD most of the time
12:42.47cursorPesky customers bugging me all the time
12:43.07sudhir492coppice: so grade of service is in 90s. closer to 100, of course provided asterisk does not go down
12:43.09cursorI tend to pick up the VM and call them back when I've finished what I'm doing
12:43.21coppicedunno. The T.38 modem in openh323 doesn't seem to have been adapted yet. It still uses the BSD interface. You can build that into linux 2.6 for backwards compatibility. The distros don't seem to have it by default, though, and its clearly on the way out.
12:44.00cursorWhich service providers allow T.38?
12:44.04coppicesudhir492: 1% or 2% are the kinds of figures most business users expect
12:44.17cursorperhaps this calls for a chan_t38
12:44.31sudhir492coppice: In addition, if there are a few signals, that is not a big deal as far as business is concerned.
12:44.33coppicecursor: I should hope most of them do
12:45.00cursorI don't know of any fax providers except for eFax
12:45.01sudhir492coppice: busy signals
12:45.02cursorI probably get two faxes per year
12:45.07cursorand send one every two years
12:45.20tzangerI can't reliably send faxes over to nufone...  mind you we have a Canon IR3300 which is a picky fax to begin with
12:45.28coppicenot fax providers VoIP providers. If you call and use a FAX modem most of them should automatically go into T.38 mode
12:45.30tzangerso I don't believe it to be a problem with nufone at all
12:45.32cursorSo I only keep a number to hand for that very rare time when someone will ask
12:46.03cursorI have a 1-877 from NuFone
12:49.12cursort.38 is probably just a codec
12:50.07coppiceT.38 is effectively a codec. It demodulates the modem tones, and sends the actual binary data across the IP channel.
12:50.20cursorright
12:50.23cursorsounds interesting
12:50.29cursorlike OOB DTMF
12:50.44tzangernow that's a nifty idea
12:50.58coppiceNothing else will get FAX through 100% reliably. Even using u-law or A-law has lots of issues.
12:51.15cursorjitter buffer etc., yes
12:51.33tzangercoppice: yup...  I'm 8 hops from nufone with about 30ms lag (sometimes that much jitter) and it's hit-and-miss
12:52.17coppicesamples dropped and inserted for a variety of reasons. See http://www.opencall.org/faq
12:52.21cursorhaha
12:52.26cursorthis is my NuFone: OK (122 ms)
12:52.46cursorMy local provider: OK (9 ms)
12:53.06cursorFWD is 91 and IAXtel is 109
12:53.14coppiceAny slip of even one audio sample kills a FAX
12:53.22cursoryes
12:53.31miller7cursor: what is your local provider?
12:53.40cursorTelAppliant
12:53.52cursorLondon
12:56.19cursorRFC 3362
12:56.25*** join/#asterisk folsson_ (~filip@62.209.162.178)
12:58.48chuji,
12:58.53cursor.
12:59.07chuji:P
13:00.16cursorDo you know of any SIP/IAX providers that do T.38?
13:00.28cursorIf there are some then a channel may be worth writing
13:00.35cursorif not then it'll be a waste of time
13:00.42cursorand it couldn't be tested anyway :-)
13:01.08coppicethat argument doesn't hold any water
13:01.53cursorThat's good
13:02.03cursorWater and IP networks don't mix well
13:02.35coppiceThere are people doign T.38 with SIP. There won't be any doing it with IAX until someone provides some support. Implement it, and they will come.
13:02.44chujihmmm... I thought Incoming SIP calls used the [general] context in sip.conf? Is that not correct?
13:02.58chujiI thought the explicit contexts were for outbound
13:02.58cursorWell, if it works over SIP then it'll work over IAX
13:03.03cursorespecially if it's just a codec
13:03.22coppiceIt isn't just a codec. Its somewhat like a codec
13:03.33cursorIt could be implemented as a codec
13:03.39cursorin theory
13:04.25cursorI'm assuming the T.38 data could be sent as RTP packets, over a session initiated via SIP or IAX etc.
13:04.36cursorAs long as it had its own codec number
13:04.39coppicewe are looking at that. There are problems. The code scheme in * assumes the directions are independent. With T.38 the directions are links. The transport needs to be new, as T.38 doesn't use RTP.
13:04.43cursorand people could allow=t38
13:05.17cursorok
13:05.37coppiceThere is a proposed T.38 over RTP option from Cisco, but the current standard allows for UDP or TCP, neither being RTP.
13:05.44cursorI don't really have a lot of interest in, or use for, fax technology
13:05.56cursorit just sounded like a cool thing to do
13:06.03cursorand a nice addition to Asterisk
13:06.48coppiceit is actually a very important addition to *. Without filling in all the gaps it has a hard time making headway in a lot of uses.
13:06.53cursorI've only got three patches in Asterisk so far
13:07.03cursorAlthough I have another three outstanding in the bug tracker
13:07.54cursorPresumably we could have something like this:
13:08.05cursorexten => fax,1,FAX(${EXTEN})
13:08.21cursorassuming the EXTEN != 'fax' :-)
13:08.39cursorso * detects a tone and bounces the call to the fax extension
13:08.57tzangerit already does that
13:08.59cursorassuming the original EXTEN is still valid, we could call a chan_fax to dial and do T.38
13:09.07cursorI know it does
13:09.09tzangeryes that woudl be the new part :-)
13:09.16cursorI have exten => fax,1,Hangup :-)
13:09.31*** join/#asterisk Dave`` (dave@ool-4352e324.dyn.optonline.net)
13:09.35Dave``ave\\
13:09.37tzangerso something like exten => fax,1,Dial(T38/${T38PROVIDER}/${EXTEN}) ?
13:09.37Dave``er.
13:09.50cursoroops
13:09.51cursoryes
13:09.57cursorchan_fax would be like that, yes
13:10.01cursoror chan_t38
13:10.19cursorIt'd be a channel, called via Dial()
13:10.40cursorAssuming ${EXTEN} is still correct
13:10.48cursorand hasn't been replaced with 'fax'
13:11.05cursorwhich would be silly anyway
13:12.27cursorBut we'd still need a T.38 service provider to test against
13:12.52cursorSomeone who would be willing to let us send rubbish a few times
13:19.33*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
13:23.06chujicurser: When a new call comes in via a sip provider, does it get it's extensions context from the [general] or from the individual peer section?
13:23.31cursorthe individual user section
13:23.34cursornot peer
13:23.58cursoruser == incoming
13:24.01cursorpeer == outgoing
13:24.06cursorfriend == both (evil(
13:24.14chujihmm, so should I have two sections in sip.conf for my broadvoice?
13:24.22cursoryes
13:24.31chujiok, so that is what is hosing me
13:24.38chujiI for user, 1 for peer
13:24.40chujigotcha
13:24.50chujiMy shit works sometimes, and not others
13:24.50cursorand you'll probably need a "register" directive too
13:24.56chujiI have the register
13:24.59cursorok
13:24.59chujiI'm cool there
13:25.02chujiThanks!
13:25.05*** join/#asterisk sauber (~ask@Gf7dc.g.pppool.de)
13:25.06chujiLemme go try that
13:29.40cursorI wonder if it could be possible to set up a "not for profit" telco
13:29.49cursorPeople's bills would be tax deductible
13:30.04chujiHeh
13:30.09cursorcharges would be cheap or free anyway because the corp wouldn't be hoarding a vast profit
13:30.34cursorJust make enough money to cover equipment costs etc.
13:30.42cursorand vast salaries
13:30.45*** join/#asterisk iguy (~iguy@dsl093-197-234.mke1.dsl.speakeasy.net)
13:30.46cursorand that's it
13:31.28cursorcould be fun to apply and see :-)
13:32.38visik7recompiled asterisk
13:32.40visik7and got this
13:32.40saubercursor, just one problem.. wheter you need people before you start to buy equipment or you buy it yourself and get money back afterwards .. i guess the second is the only thing possible. so you need money first :)
13:32.41visik7[chan_modem_i4l.so]/usr/lib/asterisk/modules/chan_modem_i4l.so: undefined symbol: ast_unregister_modem_driver
13:32.41visik7Loading module chan_modem_i4l.so failed!
13:32.43visik7sorry
13:33.00visik7what's wrong ?
13:33.15cursor(sauber) yes - there's always a fly in the ointment
13:34.53cursorEdit your modules.conf file
13:34.57cursornoload => chan_modem.so
13:35.12cursornoload => chan_modem_aopen.so
13:35.12cursornoload => chan_modem_bestdata.so
13:35.12cursornoload => chan_modem_i4l.so
13:36.47cursorbrb
13:43.52cursorI think it's time for me to go
13:44.29cursorA nice sunny Sunday afternoon and I'm stuck in my office
13:44.40cursorAlbeit a nice air conditioned office :-)
13:44.50tzangerwow
13:44.52tzangerit's only 10amhere
13:46.17cursorJust look at your cursor on your screen
13:46.27cursorit's always one step ahead of what you type :-)
13:46.36wasimheh, its a fat blob on my screen
13:46.41wasim:P
13:46.43cursorbah!
13:47.04Bobby_Ewingcursor - where ya from
13:47.10cursorLondon
13:47.20cursorEngland
13:47.28cursorIs there anywhere else?
13:47.40Bobby_EwingScarborough, North Yorkshire
13:47.53cursorDamned northeners ;-)
13:47.57Bobby_Ewingheh
13:48.07Bobby_Ewingi was down in brighton last weekend, but i prefer home
13:48.18cursorI was in Worthing last weekend
13:48.29cursorOnly a few miles away from Brighton
13:48.36cursorI'll be there next weekend too
13:49.07cursorI try to get away from here every couple of weekends
13:50.22*** join/#asterisk mhnoyes (~mhnoyes@user-2ivfjsu.dialup.mindspring.com)
13:51.08visik7cursor I need chan_modem_i4k
13:51.09visik7cursor I need chan_modem_i4l
13:51.16cursorwhy?
13:51.29visik7I use a i4l card
13:51.35cursorok
13:51.41cursorI don't know anything about those
13:51.48DaminI need chan_acousticadapter
13:52.12tzangerhahaa
13:52.13coppicewhat about chan_charlie
13:52.15visik7;)
13:52.23tzangerit's like 3pm there is it not?
13:52.29cursoryes
13:52.32cursor2:52pm
13:56.18visik7is not possible
13:56.29cursorAnything's possible
13:57.37wasimincluding stricking a matchstick on a wet soap bar
13:58.14cursorWith a lot of effort
13:58.52coppiceI've never stricked a match, so for all I know it may well be possible :-)
14:03.38*** join/#asterisk drumkilla (~russelb@130-127-109-131.thornhill.resnet.clemson.edu)
14:06.17cursor(visik7)
14:06.22cursorgo to your asterisk source
14:06.31cursoredit "channels/Makefile"
14:06.41cursorfind these lines:
14:06.41cursor#chan_modem.so : chan_modem.o
14:06.41cursor#       $(CC) -rdynamic -shared -Xlinker -x -o $@ $<
14:06.45cursorremove the comments
14:06.50cursormake asterisk and install
14:07.00cursorThat'll probably help
14:07.06cursorI haven't tried it myself
14:07.13cursorbut it looks likely to me
14:09.44cursorof course, it might not help
14:10.00cursorbut I can't tell from here because I don't have a voice modem
14:10.32*** join/#asterisk usam (~alx@203.147.59.129)
14:11.59*** join/#asterisk sudoer (~toy@pool-68-160-154-250.bos.east.verizon.net)
14:13.14usamhello, I couldnt have the Clarent CPG101 to wirk with asterisk. I can hear the dial-tone but the the unit wont detect the dtmf pressed .. Has any1 got the CPG101 working with asterisk? which firware?
14:13.56visik7cursor thanks I try
14:14.23cursorI've never heard of Clarent
14:14.26cursorsorry
14:15.37usamits a DG-104S clone
14:17.39cursorSorry - don't help :-)
14:17.47cursorWhat does it do?
14:18.50cursorThat's original
14:19.24cursorIs the CPG like a Sipura SPA-2000 or a Cisco ATA?
14:19.36cursori.e. plug phones into it and it spits SIP out the other end?
14:19.50usammgcp
14:20.13cursorok
14:20.26*** join/#asterisk eGnarF (mrbk@eris.hb.lu.se)
14:20.28cursorI haven't use MGCP, which probably explains why I haven't heard of that kit
14:20.45usamHM....
14:20.49postelAnybody knows how i turn off CDP on cisco ATA 186? Keeps sending packets on my wire trying to discover other cisco "friends"
14:21.21cursorperhaps you can switch some sort of 'mgcp debug' mode on in the asterisk cli
14:21.27cursorand see if the box is trying to talk
14:21.33eGnarFpostel: telnet to it and say "no cdp activate" or something.
14:21.39cursoror use tethereal to do the same sort of job
14:23.16posteleGnarF: connection refused, it fully works tho, and i can see the web config on ip/dev
14:23.18usamcursor: I will update you later if i get it working with the "reset" ..
14:23.35cursorok
14:23.42*** join/#asterisk nmkha (~nmkhaus@203.210.217.198)
14:23.45eGnarFpostel: Ok.. I don't know about the ata. But checked with a cisco router I have. The command "no cdp run" disables it.
14:23.54eGnarFpostel: Maybe the ata doesn't have a cli...
14:24.37posteleGnarF: xm, lets google a bit more, thanks anyway dude
14:24.45eGnarFpostel: no problemo.
14:27.42postel0x6A in the opflags on the web interface or ivr menu 323 and then 106
14:27.46postelfound it
14:27.54postelstaight from the horse's mouth
14:28.38postelif theres a bot round feed it the info, im sure some other poor soul would look for it at some point ;)
14:28.52wasimpostel: wiki it
14:31.26cursor~seen my lunch
14:31.27jbotcursor: i haven't seen 'my lunch'
14:32.02cursor~seen George Dubbya's brain
14:32.03jbotcursor: i haven't seen 'george dubbya's brain'
14:32.16cursorThat's about all the bot does
14:32.18robert_wwlhahahaha
14:32.28DarkFlib~seen my weiner?
14:32.29jboti haven't seen 'my weiner', DarkFlib
14:33.00visik7cursor no way :/
14:33.12cursorYou need to see this animation: http://www.jibjab.com/
14:33.23DarkFlib~asterisk docs
14:33.37DarkFlibjbot asterisk
14:33.38jbota PBX (Private Brance eXchange) and telephony toolkit. http://www.asterisk.org, or very cool, or awesome
14:33.49DarkFlibjbot asterisk help
14:33.50jbotDarkFlib: KCI error, or a problem with the Keyboard-Chair Interface.
14:34.04DarkFlibjbot asterisk docs
14:34.05jboti haven't a clue, DarkFlib
14:40.57*** join/#asterisk dnc (~duncan@duncan.wireless.org.yu)
14:41.08cursor~fire jbot
14:41.10jbotI'm sorry, jbot, but it just isn't working out.  Pack up your stuff.
14:41.40cursor~hump jbot
14:41.51cursorbah!
14:44.28*** join/#asterisk biot (~bert@sumner.biot.com)
14:44.52cursor~lobotomy
14:44.53jbotI feel different somehow.
14:45.03cursor~unlobotomy
14:46.25tzangerhahaa
14:48.29tzangeroh
14:48.38tzangerI thought you humped him and he felt different somehow
14:48.46cursorhaha
14:48.46Corydon76-home~fuck cursor
14:48.51jbotHumps cursor's leg
14:48.52cursorcookie
14:49.04cursor~cookie
14:50.20cursor~say stop asking me to fuck cursor, Corydon76-home
14:50.22jbotstop asking me to fuck cursor, Corydon76-home
14:50.31cursor:-)
14:50.45Corydon76-homeUh huh
14:50.51Corydon76-homeYou know you like it...
14:51.14cursorhaha
14:51.28cursor~fire Corydon76-home
14:51.30jbotI'm sorry, Corydon76-home, but it just isn't working out.  Pack up your stuff.
14:51.46Corydon76-home~abuse cursor
14:51.48jbotACTION smacks cursor across the face. "Take that, Bitch!"
14:52.26Corydon76-homeHey, hey... jbot called you his bitch... we now know how it is between you two...
14:52.37cursor:-)
14:52.42visik7cursor solved
14:53.03cursorgood
14:53.25cursor~asterisk-related stuff
14:53.31*** join/#asterisk tclark (~TC@S01060080ad113f15.gv.shawcable.net)
14:53.34cursorNope - doesn't know any
14:53.56Corydon76-home~oink
14:53.57jbotrumour has it, oink is the sound Ned Beatty makes when being sodomized by hillbillies
14:54.12cursor~moose
14:54.13jbotMOOOOSE PENIS!!!
14:55.05*** part/#asterisk altamic (~Michele@212.141.103.240)
14:55.26cursor~dict moose
14:56.01*** join/#asterisk Tekati (~captain@cpe-66-75-215-63.bak.rr.com)
14:56.16PilotPTK-Home~dict sodomy
14:56.23inspiredis zaphfc better than chan_capi? what should I go for when I'm buying an isdn card
14:56.25PilotPTK-Home~dict hillbilly
14:56.43cursorIt probably depends upon the card
14:57.04cursorIf the card is CAPI compliant then you'd probably be best with chan_capi
14:57.15cursorprobably
14:57.23cursorIf one doesn't work, try the other :-)
14:57.35inspiredI just ordered a Billion card which claims capi 2.0 is supported, but I also know that it has a hfc chipset
14:57.47cursorok
14:57.58cursorAt least you have a fallback
14:58.00DaminHmm...
14:58.01inspired(at least it should have) I'll try capi then
14:58.01Daminhttp://www.tedas.de/english/ip_dect.htm
14:58.06cursorI hear that chan_capi works well
14:58.10DaminAnyone seen one of these in action?
14:58.18inspiredok
14:58.32sudoer~dict cockblocker
14:58.39cursoroooooo - I want one
14:58.58cursorThe DECT kit, you idiot
14:59.02cursornot the cockblocker
14:59.03cursorhaha
14:59.18cursorbefore someone tries to imply something
14:59.20cursor:-)
15:00.15cursorActually, I have two DECT phones plugged into a Sipura SPA-2000, so I don't really need one
15:04.35Damincursor: Riiiiiigggghhhhtttt.....
15:04.35DaminOff to lunch..
15:04.35cursorAh - lunch
15:04.35cursorI almost forgot
15:04.37cursor4pm now
15:04.37cursorLunch time
15:04.37chuji4pm? where are you?
15:04.38cursorhere
15:04.38cursorLondon
15:04.38chujiGotcha
15:04.38cursorperl -e 'print qq^9(A)gzk\b1,\\\^^^qq^rM7@\tZ<i]_4T^'
15:04.53*** join/#asterisk Darwin35 (~darwin35@pool-141-158-115-33.pitt.east.verizon.net)
15:08.10*** join/#asterisk Tekati (~captain@cpe-66-75-215-63.bak.rr.com)
15:08.22cursorRight - gotta go
15:08.27cursorlater, guys
15:08.32cursorDon't work too hard
15:09.53*** join/#asterisk Cybo (cybo@CPE0004e2277f2d-CM014400110916.cpe.net.cable.rogers.com)
15:10.56usamcursor -- OK .. its working after i did a factory reset .
15:16.20*** join/#asterisk DrRighteous (~poll49@ool-435717de.dyn.optonline.net)
15:16.38*** join/#asterisk k3nnyg (~k3nnyg@pcp06942264pcs.nrockv01.md.comcast.net)
15:17.43k3nnygNew to this < What is the minimum hardware needed to run asterisk ?
15:17.43drumkillanone
15:17.43k3nnygnone ?
15:17.44drumkillayep.
15:17.51drumkillait depends on your application of asterisk
15:18.14k3nnygI am talking about the computer itself.....intel
15:18.19drumkillaah.
15:18.27drumkillai thought you meant telephony hardware
15:18.29k3nnygmemeory anfd CPU
15:18.34k3nnygSorry
15:18.37drumkillano problem
15:18.46*** join/#asterisk RoyK (~roy@213.115.144.116)
15:19.11drumkillawell, i'm not sure about that ...
15:19.56k3nnygWhat do you find it runs best on ???? (memory/CPU)
15:20.03Corydon76-homeIt depends on what you're trying to do
15:20.07drumkillaright
15:20.15drumkillai have it running on a 400mhz/128 MB machine
15:20.16Corydon76-homeLess number of concurrent channels requires less CPU
15:20.21drumkillabut it doesn't have to handle very much
15:20.55Corydon76-homehome machine, single physical channel
15:21.15k3nnygThats what I thought. More intensive application the more hardware nec. Just trying to get a baseline basically
15:21.27Corydon76-homeWhat is it that you want to do?
15:22.06k3nnygSmall poffice....ext. dialing (PLAR), and maybe least cost roiuting between the 3
15:22.08DarkFliba 350Mhz CPU works fine for 4 channels here as long as the codecs don't require much cpu power
15:22.12k3nnyg3 offices that is
15:22.23k3nnygWOW a T1 on that :)
15:22.33k3nnygGotta love Linux !
15:22.40Corydon76-homeGranted, the T1 was all data
15:23.06RoyKnot 24 B's then...
15:23.29k3nnygShoot....wife is calling me for breakfast.......gotta run.....bbl
15:23.29drumkillaawesome
15:23.43drumkillaand k3nnyg gets cracked by the whip!
15:23.46drumkilla:p
15:23.51Corydon76-homeRoyK: no, nethdlc
15:24.17Corydon76-homeI'm sure the P200 could handle a PRI just fine
15:24.36Corydon76-homeIt's the codec transcoding that really takes a toll on the host CPU
15:24.58Corydon76-homei.e. voip
15:25.23Corydon76-homeOh, and... it's a P200MMX
15:25.50Corydon76-homeDon't think it could survive without the MMX optimizations
15:26.25Corydon76-homebut that would be a neat test to try
15:26.57Corydon76-homeI suppose the real problem might be trying to conference all those channels... the echo cancellation would probably kill it
15:32.01RaffiOlAny Idea where i can get a 7970G Image?
15:32.59RoyKhm
15:33.00RoyKshit
15:33.19RoyKI have GRANTed and so on, but I still can't connect from cdr_odbc or _mysql
15:48.39*** join/#asterisk bigmurr (~mlisook@VDSL-130-13-95-214.PHNX.QWEST.NET)
15:58.18DrRighteousCan someone exsplain why in inbound SIP calls, its always trying to direct the call to default context, even when there is a different context specified?
15:58.42DrRighteousI have context=inbound_1 in my user section
15:58.52DrRighteousbut debug shows its trying to go to default
16:07.06Corydon76-homeDial(SIP/exten@context@host)
16:07.53Corydon76-homeOr is that SIP/user@host/exten@context ?  I always forget which
16:10.29*** join/#asterisk {Sean} (sean@seandesk.sean.net)
16:11.33DrRighteousSince non of you can answer my questions... No Pizza for anyone!
16:11.51*** join/#asterisk snewpy_ (~markl@203-173-45-53.dyn.iinet.net.au)
16:13.56*** join/#asterisk pointer-gaim (~pointer@pcp03533773pcs.summit01.tn.comcast.net)
16:16.58*** join/#asterisk zotz (~zotz@24.231.36.159)
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16:20.35DrRighteousCan someone exsplain why in inbound SIP calls, its always trying to direct the call to default context, even when there is a different context specified?
16:20.35DrRighteousI have context=inbound_1 in my user section
16:20.35DrRighteousbut debug shows its trying to go to default
16:21.01Corydon76-homeDrRighteous: I responded 5 minutes ago
16:21.41DrRighteousI thought you were talking to someone else
16:21.49Corydon76-homenope
16:21.54DrRighteousthis is for INBOUND connection, Im not trying to Dial anyone
16:22.03Corydon76-homeWhose service?
16:22.07DrRighteousBV
16:22.33Corydon76-homeDo you have a context set in the general section?
16:22.37DrRighteousno
16:22.41Corydon76-homeDo that.
16:22.47DrRighteousI want seperate contexts for each provider
16:23.01Corydon76-homeNot all providers support sending a context
16:23.16Corydon76-homeI know voiptalk, in particular, does not and will not.
16:23.40DrRighteousIve never gotten voiptalk to work anyways.. I paid for an account but it never worked
16:23.52Corydon76-homeI wouldn't be surprised to learn that BV is the same way
16:24.02Corydon76-homethough I've never tried to work with them
16:24.13DrRighteousbut isnt it the job of the context= statement in the providers block in my sip.conf to determine where its directed?
16:24.18Corydon76-homeYeah, voiptalk uses the context specified in the general section
16:24.40DrRighteousI mean its comming into MY box, why cant I direct it to the context I want?
16:24.41Corydon76-homeNo, the job of the context statement is to say which contexts the user is allowed into
16:24.48Corydon76-homeYou can specify multiple contexts
16:24.59DrRighteousmultiple?
16:25.09Corydon76-homeYep
16:25.17DrRighteousso its the provider that determines which inbound context they go into?
16:25.23Corydon76-homeYep
16:25.37DrRighteousyou can see how thats a little strange, eh?
16:25.54DrRighteousI mean that I cant direct the traffic the way I want on my own box
16:25.55Corydon76-homeAnd they generally don't specify a context, in which case, the default context or alternatively, the context specified in the general section
16:27.01Corydon76-homeNobody said chan_sip was perfect, that's just the way it works right now
16:27.30DrRighteoushehee
16:28.33Corydon76-homeI'm sure there's a good reason for it... probably they're forbidden from doing anything else according to the SIP spec
16:28.34*** join/#asterisk brc_ (~root@ip24-251-182-226.ph.ph.cox.net)
16:29.03DrRighteousgrrr
16:29.14*** join/#asterisk RoyK (~roy@213.115.144.116)
16:29.47*** part/#asterisk lethol (~lethol@201.128.129.176)
16:30.26RoyKwhat's a good sip client for linux/X?
16:30.56Corydon76-homeHaha, you used good and sip in the same sentence...
16:31.14RoyKs/good/decent/
16:31.22RoyKs/decent/ok/
16:31.27RoyKs/ok/usable/
16:31.30RoyKsomething
16:32.15Corydon76-homeDunno, I use hardphones
16:32.34*** join/#asterisk Halog3n (~weechat@209.159.235.241)
16:33.13Halog3nanyone know any freeware windows voip software I can use to test things out with?
16:34.00*** join/#asterisk postel (~postel@host217-42-116-137.range217-42.btcentralplus.com)
16:34.39RoyKHalog3n: x-lite
16:34.45RoyKHalog3n: works with SIP
16:35.03*** join/#asterisk postel (~postel@host217-42-116-137.range217-42.btcentralplus.com)
16:35.16RoyKHalog3n: or iax phone for windoze http://www.sokol-associates.com/IaxPhone.htm
16:36.10deddiaxcomm from sf works too
16:36.14deddor firefly
16:36.38RoyKor telnet
16:37.28*** join/#asterisk dcox (dcox@63.252.229.9)
16:38.32*** join/#asterisk MustDie (voip@205.247.13.202)
16:38.36*** join/#asterisk HiTech69 (~hitech@34-29.202-68.tampabay.rr.com)
16:39.20*** join/#asterisk ^ClaSsY (~^ClAsSy@203.175.66.124)
16:48.22*** join/#asterisk gr0mit_home (~wendolene@wendolene.txrx.org.uk)
16:48.53gr0mit_homeanyone having problems with sipgate.de calls to the uk?
16:57.05*** join/#asterisk ScaredyCat (~ScaredyCa@f26162.upc-f.chello.nl)
17:01.52*** join/#asterisk RoyK (~roy@213.115.144.116)
17:06.05*** join/#asterisk Dobaj (~ryanair@avonstreet.plus.com)
17:08.28*** join/#asterisk binar_ (~benedikt@merkur.benedikt-wildenhain.de)
17:08.32binar_re
17:08.46Dobajanyone help with a dial in and out problem using FWD
17:09.10deddDobaj: sip or iax2?
17:09.23Dobajsip... now worked out iax2 yet
17:09.32deddok
17:09.34dedd<PROTECTED>
17:09.50deddand please put any large pastes on pastebin.ca and just put the link here
17:10.36DobajI setup the system so I dial 8 then hte FWD number... I can see the call going out then I get a message within 2 secs saying no one answered
17:11.01binar_I am trying to use an i4l-card with asterisk, but when trying to call an voip-phone or the mailbox from outside, I only get Unable to find a path from UNKN to SLINR; any ideas where this can be fixed?
17:12.54Dobajerror message is on pastbin.ca/393
17:19.17*** join/#asterisk sung (sung@producto-valvo.com)
17:20.53*** join/#asterisk MamboKing (~mambo@d141-65-140.home.cgocable.net)
17:20.55MamboKinghey
17:21.32MamboKingquick question, I just updated my rH9 to FC2 zaptel barfs on make
17:21.57MamboKingi followed the procedure for linking to the new kernel sources, but it still doesn;t like it
17:22.12MamboKingI'm wondering if there is a syntax error on my part creating the softlink
17:22.41MamboKingln -s linux-2.6.6-1.435.2.3 ./linux-2.6
17:23.03*** join/#asterisk bonbon-home (~happy@81-86-0-190.dsl.pipex.com)
17:23.20MamboKingcat /proc/version is 2.6.6-1.435.2.3
17:23.59MamboKinghowever my kernel-source is 2.6.5-1.358 everything has been update with yum
17:24.25MamboKingwould there be any issues linking linux-2.6 to the .358 kernel-source?
17:24.42*** join/#asterisk altamic (~Michele@212.141.96.53)
17:25.29MamboKingnever mind
17:25.31MamboKinggot it
17:25.34MamboKingciao
17:26.35bonbon-homewhat's the best way to do command substituion in extensions.conf?
17:30.44izodude
17:30.48*** part/#asterisk izo (~izo@izo.warpl.ipxxi.pl)
17:30.58*** join/#asterisk izo (~izo@izo.warpl.ipxxi.pl)
17:31.39bonbon-homehow can I make the command in a string in extensions.conf be contained completely in a variable?
17:38.01visik7mmmm
17:38.22bonbon-homeit doesn't seem to work
17:38.35*** join/#asterisk Agent_sx (Chilling_c@12-220-162-81.client.insightBB.com)
17:42.04bonbon-homedoes anyone know if it's possible to do something like: exten => _X.,1,${DS1} ?
17:42.14bonbon-homewhere {DS1} is the complete command string
17:42.18bonbon-homeit doesn't seem to work
17:42.40bonbon-homepbx.c:1274 pbx_extension_helper:  No application '${DS1}' for extension
17:46.50citatsbonbon-home: no there is no substitution on that part, but you can make it happen
17:47.14citatsuse the Exec application that corydon put together and its cake
17:47.39bonbon-homeah
17:48.10bonbon-homeyou mean  the Exec app built into asterisk?
17:48.45bonbon-homewhat about Eval?
17:49.12bonbon-homecitats: you use Exec?
17:57.00binar_I am trying to use an i4l-card with asterisk, but when trying to call an voip-phone or the mailbox from outside, I only get Unable to find a path from UNKN to SLINR; any ideas where this can be fixed?
17:58.31*** join/#asterisk sudoer (~sudoer@65.75.148.190)
17:59.16^ClaSsYi am trying to use asterisk on a large level , i have a central office and 9 branch offices , i have 2 lines hunted to one number at each branch office , i want to know if i can route calls coming to these numbers to the head office somehow?
17:59.21^ClaSsYusing multiple boxes?
17:59.50*** join/#asterisk decode (mikael@24.241.192.254)
18:03.45tclark^ClaSsY: what architecture do you have in mind, will each office have an asterisk box & be connectted t the pstc for each of those hunted numbers ?
18:04.03tclarkpstc == pstn
18:04.23^ClaSsYtclark: yes , i can have box @ every office
18:04.57^ClaSsYbut i want to skip the box , use something small , like ata or something ...
18:05.03^ClaSsYi dont have any problem of bandwidth!
18:05.09visik7how can I handle sip call to asterisk like a normal call through i4l device ?
18:07.04^ClaSsY?
18:07.24tclark^ClaSsY: if you have a pstn interface requirement at each location & you dont want a asterisk box there just a ata type device then those ata would nned some type of fxo interface, in that case why not have all line come to central office & just route out the ata device
18:07.33visik7with the default configuration when asterisk is running and I call my phone number asterisk start with demo
18:08.07visik7I want the same on sip channel
18:08.21tclarkline get a t1 at the central office & assign 2 did's to each location then send those voip to the ata devices
18:08.37^ClaSsYtclark: they are at remote locations ... we dont have t1's here
18:08.50tclarkno I say get t1 at head office
18:08.56DrRighteousCan someone tell me the proper way to signup for SIPPhone for my * box?
18:09.08^ClaSsYi have like remote offices , 2 lines each , people calling in the number , i want it to ring @ main office
18:09.39dedd^ClaSsY: so route the pstn #'s back to the central office over voip
18:09.57^ClaSsYdedd: are * boxes necessary for that?
18:10.08tclarkdedd: read up he does not want * box at the remote
18:10.17deddtclark: right but the spa 3000 can do it
18:10.24tclark1 fxo only
18:10.31deddu are looking at about 200 boxes an office though
18:10.34deddto cover the 2 lines
18:10.39deddw/ 2 sperate units
18:10.48deddbut i dunno any other stand alone fxo method
18:10.53deddthen an ata w/ fxo
18:11.21^ClaSsYhmms
18:11.54deddother option is to crap the pstn and get remote dids delivers to asterisk and then back to the offices as needed
18:12.09^ClaSsYthis is Pakistan :)
18:12.14^ClaSsYVoIP = illegal
18:12.32implicitLOL!
18:12.34implicitwho gives a shit?
18:12.39implicitillegal stuff is fun
18:12.47^ClaSsY:)
18:12.52^ClaSsY7 years of jail
18:13.02dedd^ClaSsY: then the ata route would do ok
18:13.14implicitClassy, what is voip defined as in Pakistan?
18:13.35^ClaSsYso it would be like pstn -> ata -> head office asterisk thru internet?
18:14.17deddyup
18:14.46^ClaSsYhmms nice .. which ata do u suggest
18:14.55tclarkhis cost is going be more than a pc with 2 fxo cards in in with the 2 sipura 3000 :(
18:15.26tclarkwhen you include shipping & import to pk
18:15.35^ClaSsYah .. i need cost effectiveness too :(
18:18.58sudoerdedd, i compiled * on my  freebsd bozx last night, havent had time to actually test it though
18:20.36DrRighteousregister lines have to go in \[general]?
18:20.42^ClaSsYhow much is a sipura 3000?
18:22.31Mjkeay^ClaSsY, I saw one for 124 USD. Waiting to see how cheap they are in the UK :(
18:23.44*** join/#asterisk jero (~boo@modemcable055.101-131-66.mc.videotron.ca)
18:23.52jerohi
18:23.56^ClaSsYthats like affordable
18:23.58^ClaSsYwhere did you see it
18:24.19Mjkeay^ClaSsY, er not sure.. i think a quick google for sipura 3000 buy :p
18:24.37^ClaSsY:)
18:24.38*** part/#asterisk altamic (~Michele@212.141.96.53)
18:24.56Mjkeaywe have to wait till sept~ for them over here :((
18:25.14^ClaSsYah
18:26.24tclarkthat be 100% import duties in pk so you prolly $300-400 us a ony p200 must be ~100us an 2 clone x100 are 15/unit :)
18:26.48gsutterUS$124 at asteriskmall.com (n.b., I am affiliated)
18:27.17^ClaSsYhmmms
18:27.20^ClaSsYno we dont have any duty!@
18:29.55tclarkwhat does an older pc cost in pk 300Mhz to 1Ghz say ?
18:30.21^ClaSsYPIII 1ghz , old pc 256megs ram , 20gb hdd , dvdrom
18:30.24^ClaSsYethernet
18:30.36^ClaSsY233.34$
18:31.43^ClaSsYPII 400 Mhz
18:31.54^ClaSsY83$
18:34.58tclarkso you could do if $100/location with the cloners
18:35.34*** part/#asterisk decode (mikael@24.241.192.254)
18:35.54^ClaSsYhmmms yeah .. i guess
18:36.06^ClaSsYcloner = 5.8$ here
18:36.14^ClaSsYin local market , with absolutely no echo
18:36.33tclarkdont need hard drive either, just use a knnpix install
18:36.52^ClaSsYi need something stable
18:37.01^ClaSsYthat can reboot/ run automatically again
18:37.42tclarkyah that is why a knopix verion of * would be perfect here
18:38.09^ClaSsYhow can i do a knoppix version of *
18:38.14^ClaSsYand one more thing
18:38.15tclarkkeep the cd in the drive just reboot perfect reset to know install
18:38.55^ClaSsYhmms .. how can i do knoppix version of install
18:39.34tclarkthere is project out that taht has it done for ya on the wiki
18:39.45^ClaSsYi see
18:39.57bkw_ok WTF is PATA vs SATA?
18:40.02^ClaSsYone more thing , if i guy g729 codes for each location for each call
18:40.09^ClaSsYSATA = Serial ATA
18:40.24tclarkhttp://www.automated.it/asterisk/
18:40.26DrRighteousIAXTel says Im already registered, is there a way to find out my existing number?
18:41.20FuzzyCatPATA = Parallel ATA
18:41.24*** join/#asterisk mag`` (~mag__@c-24-18-138-47.client.comcast.net)
18:41.26^ClaSsYthanks tclark!
18:41.32^ClaSsYreally helpful talk
18:41.39mag``anyone want to buy two snom 105's on the cheap?
18:41.47tclarkor http://knopsterisk.com/
18:41.48DrRighteousmag: how cheap :)
18:41.52FuzzyCatno, but u can buy my 100's
18:42.05mag``$200 for two w/power
18:42.15mag``toss in shipping
18:42.41*** join/#asterisk Tili (Tili@202-133-65-162-dialup.sat.net.pk)
18:43.40DrRighteousmag: sorry I think Im going to save for a Cisco ...
18:43.50DrRighteousvoipsupply.com buys them used
18:44.05mag``thanks for the link!
18:44.11DrRighteousanyone selling ciscos or qos routers?
18:44.20tclarkthese 480i's show some nice promise when they get down to a production release
18:45.10sudoerare there any hardware phones that also work with power over ethernet?
18:45.48deddDrRighteous: gtsinc.com covers cisco gear
18:48.07tclarkFuzzyCat: yah i iknow should say live, has live cd == knoppix
18:49.43*** join/#asterisk suma (~suma@81-86-93-203.dsl.pipex.com)
18:50.15FuzzyCatbut it aint build on knoppix, that's what I'm sayin'
18:50.25FuzzyCatbuilt
18:50.30DrRighteousCan someone call my SIPPhone # for a moment, debugging something
18:50.30DrRighteous17476684598
18:50.33FuzzyCatpah stupid brain...
18:51.45tclarkFuzzyCat: what i am say is yah i know, i just had a brian fart where live cd == knoppix :)
18:52.17*** join/#asterisk bikram (~bikrams@68.160.74.243)
18:52.36FuzzyCatahh ok... no worries, I get brain farts 90% of the time lately...
18:52.57bikramhi twisted .. how are you ?
18:54.41bikramlooks like he is not there .. anyone else who can help me with a norstar asterisk inegration issue?
18:55.32sumai want to have a query which will give the best match to a string from the database, can anyone help me how can i do the same. e.g., if i have 123456789, if i have 1, 12,123,1234, i need to get 1234 for the same.
18:55.41Dobajanyone know why on an about going call I get about 30 secs talk time then is sounds like wind on the line and the call is dropped???? Using FWD SIP
18:56.29sumaDobaj: are you behind NAT ?
18:56.34DrRighteousAnyone here have SIPPhone?
18:57.21sumaDrRightrous: yes, what is your problem ?
18:57.31Dobajsuma: kinda... I have a pool of 8 but don't know how to force * to use one of them... incoming calls don't work either
19:01.33drumkillawhat is "Title" for on the disclaimer?  http://www.digium.com/disclaim.changes
19:03.42*** join/#asterisk altamic (~altamic@212.141.109.233)
19:09.07altamicdo you know if there is an english version of destar?
19:16.41altamicwhat program do you advice in order to configure * almost without manipulation of .conf files?
19:16.56*** join/#asterisk visik7 (visi@host197-36.pool80182.interbusiness.it)
19:22.22*** part/#asterisk altamic (~altamic@212.141.109.233)
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19:37.20gr0mit_homehi a anyone having problems with sipgate.de to the UK today?
19:45.49*** join/#asterisk Shaneful (~spring_ra@S0106000347085d53.vc.shawcable.net)
19:47.25*** join/#asterisk |Vulture| (~Vulture2@adsl-154-193-8.jax.bellsouth.net)
19:51.53pfndrumkilla it's your job title
19:52.16drumkillayeah, i got it eventually, heh, thanks  :)
19:54.57*** join/#asterisk daroz (~notforu@alb-24-92-56-206.nycap.rr.com)
19:56.23darozAnyone have a few minutes to talk about a multi-SOHO office project I might be tasked with?
19:56.38ariel_Hello all
19:56.41darozHey there
19:57.13ariel_daroz what's the question?
19:57.55darozHere's what I'm going to get questioend with on Monday -- We have a multi office project comming up - we need to connect N offices (where N will be somewhere between 3-12)...
19:58.12daroz... the problem is the offices will be SOHO - max 5 people each (HQ ofc.)
19:58.36|Vulture|daroz: whats your bandwidth at each office?
19:58.41darozI don't have a problem using * to do it -- my real concern is good PSTN connectivity methods and bandwitdh between the offices...
19:59.24darozIt'll be at least cable modem 1M/384... so I figure with a little outbound QoS work I can do 1 call.. MABYE 2... (g711)...
19:59.42|Vulture|daroz: you could do a coloc server for that...
19:59.54darozPerhaps DSL at the offices -- but it's all new - they are looking at a very rapid buildout in the next few months.
20:00.19darozI was thinking that -- Use a Start config to route calls through a coloed server (Ala rackshack/rackspace/etc.)
20:00.26daroz(start=star)
20:01.00darozOnly problem there is single point of failure. (And a biggie at that)...
20:01.22|Vulture|redundant servers
20:01.24pfndon't use g711u
20:01.44|Vulture|ulaw will take up a lot of bandwidth going to a colo
20:01.53|Vulture|726 perhapse?
20:02.28darozI thought about that but then I'm dealing with managing a potential sh*t load of licences... The colo server + each local * box.
20:02.40darozWhat is it about $20/ea now?
20:02.43|Vulture|726 is free
20:02.49PatrickDKwhyroute them all through one colo?
20:02.50|Vulture|729 is $10
20:02.58darozAh... Phones are 7960s tho.
20:02.58pfnpatrickdk ip centrex, why not
20:03.13PatrickDKit would take alittle more mangement, but manage them at each office
20:03.20pfncrazy
20:03.21pfnscrew that
20:03.21PatrickDKheh, no single point of failure
20:03.23darozI have to put a * box at each office...
20:03.24PatrickDKcompany wide
20:03.41|Vulture|daroz: oh then you might want a server per office, then 711u to the * server and 726 between * servers
20:03.43daroz... at leat to get PSTN access.
20:03.55ariel_daroz at each office you will need at least one PSTN line even for DSL service so each office you should put something like one Sipura the rest via sip to the main servers would work.
20:03.59pfndaroz you can do voip and forget POTS access
20:04.01darozRight... So bandwidth is better -- jitter might kill me on cable but...
20:04.24|Vulture|ariel_: yea we use PSTN for our telemarketers and VoIP for interoffice/LD
20:04.25darozpfn: I wish I could find someone reliable...
20:04.27ariel_daroz, the Sipura 3000 has one pstn port and one analog which the analog is a backup or a fax line.
20:04.43darozariel: I've seen it -- waiting to get one myself.  :)
20:04.46pfndaroz failover, home skillet
20:04.56ariel_daroz, where are you located I am now looking for work.
20:04.57pfnand if the * box is colo'd, get a T1 dropped in for cheap
20:05.00PatrickDKdaroz, psdn on sipura3000 is not usable voip
20:05.24PatrickDKpstn :)
20:05.28darozWell my second problem is getting the PSTN -=> * -- in a good way...
20:05.30ariel_PatrickDK, it's usable and it's mainly a backup.
20:05.42|Vulture|X100P?
20:05.46PatrickDKit's a usable incoming sip channel?
20:05.48pfndaroz use a PRI for incoming, and voip for outbound
20:06.00deddi have a sipura spa 3000
20:06.02deddworking nice
20:06.04*** join/#asterisk Dave`` (dave@ool-4352e324.dyn.optonline.net)
20:06.08PatrickDKhmm
20:06.09darozWell the PRI I thought of -- the problem is how to justify a PRI for 4 lines -- 6 tops.
20:06.21daroz(I'd actually prefer it -- it's a cost justification issue)
20:06.22pfnwhat are you talking about
20:06.26*** join/#asterisk markit (~marco@host119-245.pool8172.interbusiness.it)
20:06.26pfnit's 3-12 offices
20:06.28ariel_daroz, you only need the pri in the colo
20:06.36ariel_the office only get one pstn line
20:06.42darozYes.. Each office is going to get local phone service as well...
20:06.43pfnscrew getting a pstn line to the office
20:06.59darozI fear backhauling PSTN access from 6 states.
20:07.12ariel_pfn you have other issues which need them like 911 and dsl service.
20:07.25PatrickDKdsl doesn't need pstn
20:07.26darozBesides I walk right back to single point of failure again.
20:07.29pfnit's an office, screw 911
20:07.30pfn:p
20:07.34daroz(Redundant servers aside for a min)
20:07.44ariel_PatrickDK, yes it does it really depends on location.
20:07.47pfnit's soho, they can take the failure  :p
20:07.58darozSOHO, but not unimportant. ;)
20:07.59PatrickDKhmm, we don't need pstn for dsl here
20:08.01ariel_pfn no 911 is needed and it can get you in trouble.
20:08.15pfnhow is 911 needed?
20:08.17deddpfn: did u get my pm
20:08.41|Vulture|I forward users to a prompt to make sure they really wish to enter 911 calling mode
20:08.41pfndedd ah, yes
20:08.51ariel_pfn figure it out don't open your self to the law suites. have a backup just in case will save you later in life.
20:08.52*** join/#asterisk dnc (~duncan@duncan.wireless.org.yu)
20:08.56darozmost offices will hold (at start) 1-2 people in a soHO enviroment. It will roll to SOho in a retail/office complex enviroment with up to 4-5 people within 8 mos after launch.
20:09.06pfnariel I thought if you don't claim to support local service then 911 isn't required
20:09.29PatrickDKif you have employees, you need 911
20:09.30ariel_pfn, you setting offices up your opening your self into a can of worms.
20:09.32chujiDoes your sip config look at the "Via" ip address of the "From" IP address when a call comes in?
20:09.38darozpfn: we'll see what the FCC does eventually... nevermind the senate.
20:09.40chujiof=or
20:10.04ariel_I need to go daroz if you need help email me I will see what I can do for you. arielb27@hotmail.com
20:10.06darozYeah, 911 is going to be required.
20:10.11darozThanks ariel.
20:11.34Bobby_Ewingi dont want to try it, but will exten => 999,2,Dial(Zap/1/999) work ?  when you dial 999, it dials 999 on zap 1 on the pstn line?
20:11.44izoanyone experienced dropping D channels on TE405P with higher load ?
20:11.48|Vulture|yea
20:12.33|Vulture|as long as you have a 1 of course
20:13.05filetwisted: I've got pizza pockets... could make those
20:13.24darozPass some around?
20:13.47Bobby_Ewingvulture: i normally do this for outgoing on the pstn, dial 9 then the number: exten => _9.,3,Dial(ZAP/g1/${EXTEN:1})
20:14.03Bobby_Ewingso in the 999 exten, should i change /1/ to /g1/
20:14.20|Vulture|if you want to hunt from group1
20:14.26bkw_~seen bonbon-home
20:14.28jbotbonbon-home is currently on #asterisk (2h 51m 25s).  Has said a total of 11 messages.  Is idling for 2h 25m 16s
20:14.40chujiwhat's easier to use to debug sip info, Ethereal or tcpdump? Opinions?
20:14.41DarkFlib~seen darkflib
20:14.41jbotdarkflib is currently on #asterisk (1d 13h 17m 24s).  Has said a total of 11 messages.  Is idling for 0s
20:15.13pfnchuji depends on what you need
20:15.20darozchu: single device (and a terminal window that scrolls) tcpdump... else Etherreal hands down.
20:15.23pfnethereal has a lot of information, but doesn't make some of it easily displayed
20:15.35pfntcpdump provides a lot of pertinent packet information up-front
20:15.38*** join/#asterisk KryoStoffer (~kri@194.255.56.225)
20:15.43pfnand sip debug is good for purely the sip messages
20:15.58darozpfn: true -- but when you need to weed through a bunch of messages at a glance it can help...
20:16.00chujidaroz: Yeah, it's a remote device
20:16.36darozof cource you can always "tcpdump -i xxx.... port 5060 > /tmp/file"
20:16.40chujiwell I've got a sip packet coming from broadvoice that is addressed sip:mynumber@192.168.0.3;user=phone
20:16.53darozYep - I've got BV here as well.
20:16.55chujiI don't know where the hell that 192 address is coming from
20:17.13chujiI don't use that IP scheme anywhere
20:17.17chujiand that is my To: address
20:17.26chujiI'm seeing that in Sip Debug
20:17.40chujiSo, it looks like bv is sending me that
20:18.21|Vulture|chuji: 1s Ill check my servers
20:19.39darozchu: What kind of packet, INVITE?
20:20.12chujidaroz: Yep
20:20.24DarkFlibthen its the other end that is at fault
20:20.26darozLemme test real quick...
20:20.43chujiJust making sure it's not getting mangled somewhere when it hits my box
20:20.49darozAre you using the sip@broadvoice.com host name?
20:20.57chujithat's why I wanted to look at it with an anylizer
20:21.09chujidaroz: THis is inbound
20:22.07darozYes... All my To: lines read sip:1npanxxxxxxx@sip.broadvoice.com -- which is my I asked.
20:22.13darozJust tested here and it's fine.
20:22.40chujiwhat's the 1npan?
20:22.42darozI'd check your REGISTER packets and see what is being sent... make sure you don't have an externip set in sip.conf either.
20:22.52daroznpa-nxx-xxxx (the phone #)
20:23.34chujiHow do I see more verbosity in my tcpdump results?
20:24.02chujiI'll check my register
20:24.19pfnyou can do tcpdump -x
20:24.19daroztcpdump -i eth0 -s 1500 -xXn port 5060
20:24.26darozreplace eth0 as appropriate.
20:25.20chujithanks, trying...
20:25.23daroznp
20:26.04PatrickDKheh, the ap7920 is kind of nice
20:26.21chujiholy shit, should have outputted that
20:26.25chujito file
20:26.33fileto me?
20:26.43darozlol
20:29.23chujiwell, my register looks groovy
20:29.35chujiI'm using srv records in my register
20:29.44chujiwonder if that is whacking anything
20:30.01chujiarsing '/etc/asterisk/sip.conf': Found
20:30.01chuji<PROTECTED>
20:30.07chujiThe first time
20:30.18chuji== Parsing '/etc/asterisk/sip.conf': Found
20:30.18chuji<PROTECTED>
20:30.21chujithe second
20:30.35chujidca the first, and lax the second?
20:30.57darozI get the same... That's semi -normal
20:31.17pfndca is broken, yes?
20:31.50darozNews to me...?
20:31.56chujiwell, I'm working right now
20:32.02chujiAfter a couple sip reloads
20:32.06chujiand I'm on lax
20:32.22chujiAnd the To: string looks correct
20:32.55chujiI'll never get my wife to let me drop the pots line
20:33.04chujihaha
20:34.01chujiThanks for your help daroz!
20:34.15chuji|Vulture|: Any luck finding that address? Didn't realize you were lurking
20:34.31daroznp chu...
20:38.30kiso79any mysql geeks in here?
20:38.39kiso79or sql for that sake
20:39.25chujisomewhat kiso79, what's up?
20:40.05darozyeah kiso...
20:40.14chujiIf you want real help though, try /join #mysql
20:41.33*** join/#asterisk e_zsolti (~goatr78@212.200.134.86)
20:41.49kiso79did that
20:41.52e_zsoltihi everyone
20:41.57kiso79no geeks available
20:42.14DaminMmmmmm.. College Hotties..
20:42.32e_zsoltiIs someone available for a little discussion ?
20:43.06kiso79daroz: can I get you to look at bit that doesn't work?
20:43.37kiso79http://www.pastebin.com/88199
20:44.05e_zsoltican anyone help me with ParkAndAnnounce ?
20:45.28e_zsoltiplease
20:46.41e_zsoltianyone with a little spare time ??
20:46.43darozkiso: What version of MySql?
20:46.57daroze_z: Haven't used it.
20:47.08e_zsoltiI really need some help, cause I work on some project and ....
20:47.42daroze_z: Have you tried the Wiki?
20:48.05e_zsoltiyes but I havent found answers to my questions there
20:48.05kiso79daroz 5.0 i guess
20:48.16kiso79darow: installed it yesterday
20:48.23kiso79with apt
20:48.27darozkiso: Well if it's not 5.0 (I haven't used it) I know you can't do nexted queries...
20:48.30^ClaSsYhas anyone worked on the wakeup call script?
20:48.31darozer nexted.
20:48.34^ClaSsYand got it working?
20:48.37darozkiso: Are you getting an error?
20:48.53daroze_z: Nothing I can do for ya man -- that's about all I got. :(
20:48.53kiso79jep same one whateverI do
20:49.00darozkiso: What error?
20:49.01kiso79error 1064
20:49.52kiso79daroz: did you see the pastebin
20:49.54darozHave you tried breaking the query apart and using a transaction? (And what is 1064 anyway?)
20:50.08darozkiso: Looking at it now.
20:54.18kiso79daroz: any luck?
20:54.47*** join/#asterisk RoyK (~roy@153.80-202-161.nextgentel.com)
20:56.16*** join/#asterisk dan2 (dan@beta3.registered)
21:09.54*** part/#asterisk Agent_sx (Chilling_c@12-220-162-81.client.insightBB.com)
21:25.17*** part/#asterisk dedd (~wsuff@pcp0010079306pcs.eatntn01.nj.comcast.net)
21:25.24*** join/#asterisk dedd (~wsuff@pcp0010079306pcs.eatntn01.nj.comcast.net)
21:42.46*** join/#asterisk carnt (~carnt@179232138.rjo.virtua.com.br)
21:45.10carntHi people, It's a fast question. I'm using a * box into my office and want bypass it with my two grandstream. So i read in VOIP-info.org that i just need put canreinvite=yes and asterisk will act just like a brige. Thats correct ??
21:48.46PatrickDKnope
21:49.05PatrickDKcanreinvite=yes, means asterisk isn't even a bridge, it drops out completely
21:49.17PatrickDKhow do you mean bypass?
21:50.26carntI want * accept calls but don't use his own internet connection.
21:51.08carntI need * connect the two points to talk each others using they own internet connection.
21:51.56Dave``yes, what you described can be achieved with canreinvite=yes, carnt
21:52.07Dave``it would do a native bridge between the two sip clients
21:52.20PatrickDKya, that is careinvite
21:52.34PatrickDKbut the phones have to support it correctly
21:52.41PatrickDKor you will have problems
21:52.48carntBut i see rtp packets going on and out from my * box ? Why ??
21:53.03carnti mean in and out
21:53.08Dave``http://lists.digium.com/pipermail/asterisk-users/2004-March/039678.html
21:53.11Dave``that looks simialr to what you're asking
21:53.53carntyep
21:57.01carntSo you mean that with canreinvite=yes * will just receive the sip (ack) packets but all rtp traffic will be done using the coonection of the end points ? With canreinvite=no  * receive the RTP from both and use his internet connection for then .
21:57.49Dave``I think that's the way it should work
21:59.13carntI see.  with tcpdump then i think i see the sip packets not the rtp .. Did anyone know a good way to look rtp packets and see if they are going in and out from my box ? ;) thanks
21:59.44*** join/#asterisk aroedl (~aroedl@p3EE292DC.dip0.t-ipconnect.de)
21:59.48aroedlHello
22:07.02*** join/#asterisk Sijiero (~AA@asy85.as253176.sol.superonline.com)
22:08.02wasimfyi: farfon rev4 seems to be stable ... we should take orders for end-august delivery
22:13.44gambolputtyCan a Sipura ATA register against another one?
22:16.06gambolputtyin a peer to peer style connection?
22:16.13*** join/#asterisk prutser (Prutser@bitbucket.capcave.com)
22:16.20prutserAny ZapHFC guru's around?
22:17.28prutserGot a bunch of HFC-S ISDN cards, they accept calls just fine, but I can't seem to get them to dial out...
22:20.10*** join/#asterisk Sijiero (~AA@212.253.181.112)
22:20.18*** join/#asterisk zotz (~zotz@24.231.36.159)
22:21.09wasimhiya astmaster ... we should have a deployable iax2 phone out by end-august
22:21.51^ClaSsYwasim: how much will it cost in Pakistan?
22:22.13^ClaSsY?
22:22.20^ClaSsYor anywhere else?
22:22.34wasim^ClaSsY: initially $125
22:22.54^ClaSsYcan i buy a testing / demo version or something? i am in islamabad
22:23.25wasim^ClaSsY: of course, send a mail to sales@farfon.com
22:23.46dan2kram: ping
22:24.17^ClaSsYwasim : is it like made in Pakistan?
22:24.18visik7!seen cursor
22:24.22visik7no seen bot
22:24.24visik7:/
22:24.53wasim~seen cursor
22:24.54jbotcursor <~kevin@andromeda.office.cursor.biz> was last seen on IRC in channel #asterisk, 7h 16m 22s ago, saying: 'Don't work too hard'.
22:24.57dan2visik7: /msg seenserv seen cursor?
22:24.59kramdan2: hrm?
22:25.18wasim^ClaSsY: completely developed in islamabad
22:25.40^ClaSsYwasim : cool , i'd like to meet you some day , probably can hook up *s
22:25.42dan2kram: would it be possible to get a 120w psu to work with dual fxs module tdm400p, slim 40g hdd, and a cdrom all in a via cl1000
22:25.45dan2cl10000 even
22:26.25kramdan2: i don't know, i'd think so, but i'd be more concerned about the 40G and the cdrom than the tdm400p
22:27.07dan2kram: slimline drives are made for notebooks, low power, low heat
22:27.34kramk
22:27.45dan2kram: expecting that the cdrom drive is used only for install purposes and probably taken out when distributed, I don't think it would be a problem
22:27.57kramsounds ok
22:29.14carnt[Leaving] Reason:[auto away after 30 minutes of inactivity] ·•Polaris CTCP 2001 1.01•·
22:29.17wasim^ClaSsY: wife forced me to move to lahore
22:29.36^ClaSsYwasim : :) why o why?
22:29.42carntWhere i find a example to configure in * groups ? Like I want make a group of channels group A 100 to 200 and another group, group B 300 to 400 . Both groups can't talk with each other. If i call from group A to B i can't reach. Did anyone know where find a doc to explain this ? Thanks alot.
22:29.52^ClaSsYwasim : are you related to convergence?
22:30.09wasim^ClaSsY: yes
22:30.19^ClaSsYoh .. cool
22:30.36wasimcarnt: basic dialplan/contexts should get you there ...
22:30.41wasim~docs
22:30.42jbotDocumentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or #asterisk-doc, or http://www.voip-info.org/wiki-Asterisk
22:31.23^ClaSsYwasim : so .. i'd like to learn / work .. is there any chance your organization gives ?
22:31.26^ClaSsYi can even pay
22:31.45blitzragewasim: how can we change that so it includes http://www.asteriskdocs.org ?
22:31.50wasimcarnt: in your channel definitions, assign the channels to a particular context and then control their dial access that way
22:32.27dan2kram: as soon as I get the new mini-itx box and board, I'll send you the specs
22:32.34wasimblitzrage: something (i think) like ~docs is Documentation can be found at blah, blah, blah (but you need to be in the bots edit list)
22:32.48blitzragedoh  :)
22:33.32^ClaSsYhmms ?
22:34.14dan2kram: I wrote a few patches to lower memory usage and increase performance of asterisk, they are still experimental but look good atm
22:36.09carntthanks alot
22:38.56Damindan2: Submit them to the bugtracker..
22:39.20dan2Damin: will do soon, just trying to test
22:41.07kramdan2: will also need a disclaimer
22:41.17kramdan2: you can still place them if they're experimental
22:41.22kramdan2: just label them as such
22:42.22*** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net)
22:42.24dan2kram: I'll disclaim these
22:42.53dan2kram: frankly, Digium can have them if you guys want it, I was going to give it away under public domain :P
22:42.53Daminkram: Do you mind I post a "RoadMap" page to the Wiki summarizing the discussion we had last night?
22:43.39kramdamin: we should fine tune the language first
22:43.52*** join/#asterisk twilson (~terry@spidey.nuvio.com)
22:44.00Daminkram: That's the point of a Wiki! ;)
22:44.08Daminkram: I'll post it, you edit it. :)
22:45.07krami haven't done anything on the wiki yet
22:45.11krammaybe you can walk me through the process
22:45.18Daminkram: I just want something to be able to direct people to when they start telling me how Asterisk has nodirection! ;)
22:45.31kramdamin: suodns like a plan
22:45.33twilsonI have two asterisk boxes.  One is a registrar box and the other a PSTN gateway.  They talk sip between each other.  Is there any way that I can have the registrar box send a 302 redirect instead of bridging the call in the middle?
22:45.51Daminkram: Oh yeah.. sure.. If you can hop onto one of the iax2 conferences later on tonight I'll be happy to.
22:46.03Daminkram: Wiki editing is just like text editing. :)
22:46.12Daminkram: Easy easy easy..
22:46.17kramcool
22:47.30ddougghey kram, did i share my fantasies with you?
22:47.39kramddougg: can't say you have
22:47.42twilsonI've read what I could find about promiscredir, but it looks like it is just for asterisk receiving a 302... is this correct?
22:47.43visik7can I do something like this ? exten => s/059799542,1,Answer
22:47.52ddougghttp://aurora.telerama.com/asterisk-fantasy-p.txt
22:48.22ddouggalso, i'm looking for someone to contract to set up my home phones: http://aurora.telerama.com/as-setup.txt
22:49.00Daminddougg: #2 is easy if you have a phone that supports "auto-dial"
22:49.14kramddoug: are you using a zap?
22:49.21krami mean a phone connected by zap?
22:49.28dan2kram: have you thought about using stacks for memory allocation?
22:49.31ddouggi don't have a zap...should i get one?
22:49.46Daminddougg: Basically, when you pick up the handset, it will take you to an IVR context that will read you a menu and then let you dial out or whatever..
22:49.54ddouggi'm offering $$ to set me up..
22:49.59kramdan2: yah, probably something post-1.0, but i don't like external dependencies so i'd want something i could have built in
22:50.04ddouggyeah, that sounds rockin
22:50.15kramddougg: with a zap phone (mgcp maybe, too?) fantasy #2 is already a reality
22:50.16Daminddougg: How much you offerin? ;)
22:50.19kram#1 is more challenging
22:50.35dan2kram: Timo (Author of Dovecot) created a Public Domain, lightweight stack implementation
22:51.08dan2kram: very minimal, the other alternative is obstack which is part of libiberty
22:51.25JohnWayne#1 is just like call waiting deluxe from the phone company
22:51.26dan2kram: which is integrated into gcc iirc
22:51.31brc_kram, I think you could also do it with a 7960
22:51.53brc_ddougg, sorry I haven't had time to email you yet
22:52.44kramasterisk really doesn't do much dynamic allocation though
22:52.50kramso i don't know how big a win it would be
22:53.08dan2kram: I notice there were a bunch of functions that are just pointers to other functions really
22:53.21dan2kram: save function calls, make them inline
22:53.59kramdan2: it's only worthwhile if they're called with great frequency
22:54.12kramdan2: otherwise it's really not worth spending much time on, especially if it makes it harder to debug etc
22:54.15dan2kram: yes this is true, I'll go check
22:54.25krami'm all about inline functions, where performance matters
22:55.22dan2kram: what about doxygen, it could be handy
22:55.51brc_make progdocs
22:57.25dan2ahh cool
22:59.27visik7can someone give me his sip.conf and relative context of a working enviroment ?
22:59.37brc_huh?
23:05.22*** join/#asterisk sauber (~ask@Gf7dc.g.pppool.de)
23:05.58jerowhat do SS7 and E1 R2 do on the line ?
23:06.04dan2visik7: to connect to what?
23:06.20jeroa client is asking me if my pstn <-> voip gateway can manage ss7 and/or e1r2
23:06.37izojero : nope
23:06.53izoss7 is not yet supportted as well as r2
23:06.56jerowhat are they used for ?
23:07.20visik7dan2 to route a call through a i4l
23:07.22visik7card
23:07.45dan2not a clue
23:08.02jeroizo, any idean what ss 7& r2 are used for ?
23:08.03izojero: this are special kind of signalling like ISDN but ss7 is much much more complicated where r2 is much simplier
23:08.08dan2visik7: register foo@foo.com/<exten number>
23:08.12jerooh ok
23:08.16dan2visik7: then create the necessary extensions
23:08.44jeroizo: are they used by non-operator people ?
23:08.46twilsonI have two asterisk boxes.  One is a registrar box and the other a PSTN gateway.  They talk sip between each other.  Is there any way that I can have the registrar box send a 302 redirect instead of bridging the call in the middle?
23:09.11wasimwhy on gods earth would you want two * boxes to talk SIP between each other?
23:09.13izojero: no
23:09.21izowasim : whaza :-)
23:09.26wasimhiya izo :)
23:10.28aroedlcu tomorrow, bye
23:10.30*** part/#asterisk aroedl (~aroedl@p3EE292DC.dip0.t-ipconnect.de)
23:10.39^ClaSsYtake care everyone
23:10.44^ClaSsYg'bye
23:10.56twilsonwasim: it's a little complicated, but if you have SIP endpoints and need to handle redundancy (i.e. if an asterisk box dies, the call doesn't drop) having things like redirects and having signalling and media seperated is a Good Thing(TM).
23:11.01visik7dan2 where I have to register it ?
23:11.59twilsonwasim: short answer: my asterisk boxes that are talking via IAX deadlock about every 5 days or so and I'm looking to mitigate the consequences... :-)
23:12.29dan2visik7: dunno
23:12.32twilsonhaving 50 in progress calls drop makes customers very not happy...
23:12.39visik7dan2 ?!
23:12.39wasimtwilson: ah, but they shouldn't deadlock in the first place
23:12.39jeroWhat are the common commercial alternatives to asterisk ? I mean for bridging an existing VoIP network over the pstn ?
23:12.51wasimjero: quintum, cisco, verso
23:12.57jerothanks wasim
23:13.08wasimjero: all shaking in their boots
23:13.18*** join/#asterisk mag`` (~mag__@c-24-18-138-47.client.comcast.net)
23:13.29jeroshaking in their boots? lol
23:13.59jeroin france we would say 'pooing in their trousers'
23:14.08carnt[Leaving] Reason:[auto away after 30 minutes of inactivity] ·•Polaris CTCP 2001 1.01•·
23:15.25PilotPTK-Homemy asterisk box (with 1 x100p, 1 t400p and a bunch of sip phones hanging on it) has uptime of about 4 months.
23:15.39jerohow about the asterisk daemon PilotPTK-Home ?
23:15.42PilotPTK-Homethe only time it ever stops is when i do a cvs update.
23:15.51PilotPTK-Homewhat about the daemon?
23:16.00jeroyou just answered :)
23:16.11jerothe process lifetime
23:16.33PilotPTK-HomeI use the 'safe_asterisk' script to start asterisk, and to my knowledge, it's never had to catch a crash
23:17.11jerothis is nice, how many lines are you managing ?
23:17.26PilotPTK-Homewe have 3 incoming PRI's.  all 23 channels on all of them.
23:17.38visik7the numeration of Zap or Modem resources start from 0 or from 1 ?
23:17.44PilotPTK-Homenormal usage is about 60 channels
23:17.47PilotPTK-HomeZap/1 is the first.
23:18.10jero60 channels is nice
23:18.46PilotPTK-Homeyep.  it's actually time to add another PRI.  getting too close for my comfort
23:19.05deddPilotPTK-Home: geez
23:19.09jeroare you providing pstn to sip clients  in the back PilotPTK-Home ?
23:19.18brc_Hi pilot!
23:19.23PilotPTK-Homehey brc.
23:19.25PilotPTK-Homehow are ya?
23:19.32brc_good
23:20.57DrRighteousanyone selling any VOIP equip or routers?
23:21.23deddDrRighteous: always
23:21.37deddwhat are u looking for so i can find it and pad the cost and profit
23:21.38DrRighteousdedd: what ya selling?
23:21.45dedd^^^
23:25.56jerowhat are the disadvantages of using SER as SIP proxy in front of asterisk on my ~1000 sip clients network ?
23:26.09jeroor advantages
23:31.05tessier_jero: Depends on who you ask. From I've heard there really aren't any unless you want RADIUS billing or something like that. And * may even do that now, not sure
23:31.18tessier_I'm pretty sure google will turn up some hits on that subject
23:31.23tessier_I seem to have looked into the same thing at some point
23:31.39tessier_I don't recall the exact results but I decided to go all * and forget SER
23:33.03visik7I have an isdn line with 2 number and a i4l card connected to it why when I try to Dial(Modem/ttyI1,20,${MYCELLPHONE}) it says  Requested device 'ttyI1' does not exist
23:33.04visik7?
23:34.03*** join/#asterisk neon (~neon@201.129.51.174)
23:37.55jerothanks for the info tessier
23:38.53jeroi thought SER may have been providing simpler user management interfaces ?
23:39.14brc_bkw is evil
23:40.13*** join/#asterisk FasterRaster (~Yeahyeah@220-253-62-175.VIC.netspace.net.au)
23:40.33deddbrc_: if u say so =)
23:46.56jerowhat is the best way to transfer a call to another phone ?
23:47.10wasimjero: Dial
23:47.26jerodial # then the destination phone number ?
23:47.49wasimjero: oh you mean on the handset ... either flash-hook or # transfer
23:48.04jeroflash-hook ??
23:48.11wasimdepending on what the channel supports
23:48.35jerookay, so #transfer is better for analog phones plugged on fxs ports
23:48.51wasimjero: nah, use flash for that ...
23:48.57jerowhat is flash ?
23:49.28jerooh ok the flash key on standard phones
23:49.45jeroon some of them
23:50.03wasimelse you can just tap the onhook/offhook lever
23:50.16*** join/#asterisk litnimax (~max@212.0.199.202)
23:50.41jerodifficult on a cordless phone
23:51.19jerowasim, does it work if the analog phones are plugged onto a FXS bank ?
23:51.25jerolike mediatrix does
23:51.30wasimdepends on the bank, i'd venture
23:51.36jeroof cours
23:51.36jeroe
23:52.09jerocan I do auto-redial on a busy number ?
23:52.51wasimyeah
23:53.04litnimaxi think yes, easy. After the number is busy, n+101 extension
23:53.23litnimaxthere you place cheking varibale NUMBER_OF_REDIALS
23:53.23jero?
23:53.24wasimyeah, put some logic and put it in a loop
23:53.30jerookay
23:53.35litnimaxgotoif
23:53.44litnimaxsee asterisk wiki
23:53.52jerohow about a ldap directory ?
23:54.02litnimaxhttp://www.voip-info.org/wiki-Asterisk
23:54.23jerothanks :)
23:54.28litnimaxURW :)
23:55.12litnimaxi think ldap is not supported
23:55.31litnimaxsee here http://www.voip-info.org/wiki-Asterisk+Wishlist
23:55.36jeroyep
23:56.06litnimaxsee in commnents :)
23:56.27litnimaxby Anonymous on Friday 16 of July, 2004 [21:13:18 UTC] [Score:0.00]
23:56.27litnimaxi wish there were jabber-voip support.
23:56.33litnimax;)
23:56.36jerocan I do a kind of public address function using some sip phones ?
23:56.45wasimyou can with farfon!
23:56.46jerooften used in companies
23:57.24litnimaxwhat do you mean by public address function (my english is not so good) ?
23:57.43jerowell for example a commonly used function is
23:58.10jerothe girl at the buildings etrance presses PA on the phone and says "mr john doe is being looked for"
23:58.32jeroand every phone in the building says that through the speaker
23:58.36*** join/#asterisk evilbuny (~evilbuny@192-172-93-202.dsl.nbdsl.net)
23:58.53FasterRasterjero ICM/Page to all zones?
23:59.09jeroi have no idea of the name :)
23:59.11litnimaxwasim: farfon is hardware phone?
23:59.18wasimlitnimax: oui
23:59.30jerooui lol
23:59.30wasimwe call it intercom/page function
23:59.41FasterRasterICM = intercom, that is correct.
23:59.48jerookay

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