irclog2html for #asterisk on 20050404

00:02.21harryvvSomone was telling me if you move a voip phone from one wifi router to another though a vpn that it will not drop the call is that true?
00:02.45iosahdfh3x do you have any 561
00:02.51tzangerGO FISH!
00:03.28harryvvgo fish :)
00:03.41Qwellharryvv: not likely
00:03.41nDuffharryvv, your VPN should mask the underlying network changes, generally.
00:03.54harryvvyea I had a hard time believing that.
00:03.59h3xiosahdf: what city
00:04.10iosahdfboca raton
00:04.32antifuchsoh, hi nDuff. interesting that we meet again here (:
00:04.50nDuffantifuchs, howdy.
00:05.21harryvvI dont think there is any handoff wifi voip technoligy at the moment. I wonder what cell site equipment use.
00:05.22h3xiosahdf: yep definately, multiple choices
00:05.48nesysthere' s no one that could help me to understand my sip debug? I've problems with call-forward from ccme to * on sip trunk ...
00:06.10h3xiosahdf: just go on www.carrierone.net/dids and fill out the for more info form, and specify if you want to port your number or not
00:06.11iosahdfwhat's your url going to be again?
00:06.12antifuchsharryvv: ISTR a zyxel handheld sip phone thing that does 801.11b or g.
00:06.14iosahdfah
00:06.16iosahdfthanks
00:06.22*** join/#asterisk stustu (~stustu@fluffy.fatburen.org)
00:06.34stustuAny zaptel developers here?
00:06.35antifuchsthe battery lasts for a  whole 30 minutes or so
00:07.15h3xI count 5 options for underlying carriers in boca raton
00:07.36Veryhotknow a good place to get some IAXy?
00:07.44h3xThe cool thing is im going to allow customers to port numbers to another underlying carrier in case the voice quality sucks or anything.
00:08.23h3xwe're also going to implement caller id name+number
00:08.35h3xmost of the providers out there dont have that
00:08.37h3xit will be an option
00:08.56h3xanderiv, outgoing caller id name with a CNAM database record
00:09.03h3xdamn nick complete.  thats supposed to be "and"
00:09.12nDuffharryvv, if your VPN is able to adapt to the underlying network changes, packets traversing it don't know or care if the VPN just had to deal with a server handoff, routing differences, etc, so your connections should be unintterrupted. Now, if there's a change *after* the packets left the VPN, that's a different story; likewise if your VPN can't handle the handoff cleanly. (OpenVPN can, if properly configured).
00:09.23harryvvantifuchs thats cool
00:09.56antifuchsharryvv: yeah, but I don't think it's very useful in practical use (:
00:10.04brc_h3x
00:10.06brc_daaaamn
00:10.07brc_h3x,
00:10.08h3xbee arr cee
00:10.09harryvvyea I see. there is probebly timmer options that also could be configured.
00:10.10Godseyanyone here use broadvoice? :)
00:10.13brc_duuuude
00:10.15h3xDUDE
00:10.27h3xbrc_: I'm going to have phoenix DIDs soon too btw hehehe
00:10.35h3x*turrets*fuck ELI*turrets*
00:13.06*** join/#asterisk greg_work (~greg@d221-73-198.commercial.cgocable.net)
00:13.50greg_workwhat am I doing wrong here?  GotoIf($[${CALLERIDNAME:0:${LEN(RGPREFIX)}} != ${RGPREFIX}]?3:2)   I'm trying to check if ${CALLERIDNAME} starts with ${RGPREFIX}
00:14.05*** join/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com)
00:15.18*** join/#asterisk feklee (feklee@genba.ffii.org)
00:15.55fekleeI just transfered a setup from one machine to another.  However, on the new machine my Asterisk configuration doesn't work anymore: When I call, I always get the busy signal.
00:16.00fekleeHow do I find out more?
00:16.11patdklogs
00:16.19*** join/#asterisk djvoodz (~oli@host217-44-21-66.range217-44.btcentralplus.com)
00:16.24fekleeCan't find anything interesting in them.
00:16.40*** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
00:16.49djvoodzhi guys
00:16.54djvoodzgot a few problems with an x100p card
00:16.57fekleeIt says something about a broken pipe, but that also happened with the original setup, IIRC.
00:16.59djvoodzanyone able to help me?
00:17.17patdkdjvoodz, they don't make x100p anymore
00:17.27fekleeThe thing is, I want the incoming call to be streamed using IceS.
00:17.51djvoodzi believe its actually an x100p clone
00:18.03djvoodzi cant seem to get asterisk to pick up the line
00:18.03patdkgood luck
00:18.09djvoodzor even notice there is any form of call coming in
00:18.27*** join/#asterisk Dr-Linux (~sshah@202.163.69.3)
00:18.30*** join/#asterisk Zipper_32 (~none@d209-121-36-44.bchsia.telus.net)
00:18.38Dr-Linuxanybody can tell me please
00:18.50Dr-Linuxabout SIPPS softphone ?
00:19.07Dr-Linuxi m using SIPPS softphone
00:19.29Dr-Linuxhow i can check its registerd or not ?
00:19.35Veryhotdjvoodz: who the provider?
00:20.39djvoodzwe dont have a voip provider yet - it is plugged into an analogue pstn
00:21.09Veryhotdj: which X100p clone?
00:21.27djvoodzambient md3200
00:21.34Zipper_32I have a relatively general question; I've been put in the position of implimenting a PBX in a new warehouse/12 person office, the cement is being poured right now and I have until mid may to setup a system. I have just installed asterisk on a Redhat8 system and I am hoping somebody could explain what i need to emulate this office environment that I am going to setup.
00:22.02Zipper_32... Or point me in a useful direction.
00:22.12Veryhotzipper: check out Asterisk@home
00:22.38Dr-Linuxanybody know about SIPPS softclient ?
00:23.26bjohnsonDr-Linux: sip show registry
00:23.29bjohnsonor sip show peers
00:23.34bjohnsonor sip show users
00:23.40tzangerI'm creating a patch for asterisk
00:23.43tzangerIAX2 GIMME BEER
00:24.15djvoodzwhen plugging the phone line into the x100p there is a fuzzy echo
00:24.30bjohnsonZipper_32: make a lan and set up some softphones on them to start configuring your system.  sound quality will be better with hardware phone but will get you started
00:24.43tzangerdjvoodz: shave it?
00:25.27djvoodzhaha - cud the card be faulty or is this normal behaviour?
00:25.32bjohnsonZipper_32: decide what kind of phones you want .. for 12 people in a warehouse you might want some fxs+regular cordless phones, some voip phones, and maybe even a wifi sip phone (but they are not highly regarded)
00:26.39*** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net)
00:27.13bjohnsonZipper_32: for 12 people you probably don't want enough lines for PRI or enough extensions for channel bank .. so you will likely want small numbers of fxo and fxs devices or pci cards (or a mix)
00:27.54bjohnsonDr-Linux: I don't have time to troubleshoot your problems .. I'm just answering your question
00:28.12tzangerbjohnson: actually 8-10 lines is where I start suggesting a channel bank or PRI
00:28.41DEEZEDIs it ok to have the extensions.conf include another config which is full of includes for each one of my customers that will have there own config files. Such as asterisk.conf -> clients.conf -> account#.conf
00:28.46Dr-Linuxbjohnson: SIPPS << is softphone like x-lite
00:29.14DEEZEDthis will result in numerous account#.conf files. will this slow down or hinder asterisk by having to read each file?
00:29.47Dr-Linuxbjohnson: i'm using SIPPS softphone trail version, look for regisration option over it ..
00:30.59JerJer[interop]DEEZED:  why would you need a whole file for each account?
00:31.23tzafrir_laptopI'm trying to build asterisk-addons. I get the following error from mkdep (in app_addon_sql_mysql.c)
00:31.27tzafrir_laptop/usr/include/sys/types.h:158:20: missing binary operator before token "("
00:31.33DEEZEDto make it easy for a php script to modify each account instead of creating one big file
00:31.39tzafrir_laptopAny idea what am I missing?
00:32.01JerJer[interop]a (
00:33.28tzafrir_laptopDEEZED, I don't supposed it is much of an overhead. Anyway, you can use globbing in #include
00:33.45Dr-Linuxanybody know about SIPSS softclient ?
00:35.16*** join/#asterisk advorak (~advorak@12-220-96-185.client.insightBB.com)
00:36.48*** join/#asterisk MikeJ[Laptop] (~icechat5@pcp02795302pcs.roylok01.mi.comcast.net)
00:37.00*** part/#asterisk Darwin[laptop] (~darwin-la@24.3.226.147)
00:38.46DEEZEDthanks tzafrir_laptop
00:40.34*** join/#asterisk [shodan] (~shodan@216.113.99.160)
00:43.25*** join/#asterisk Newbie___ (~me@218.208.232.24)
00:43.37Newbie___hi, anyone familiar with perl ?
00:43.55iq~perl
00:43.56jbotextra, extra, read all about it, perl is at http://www.handhelds.org/z/wiki/Perl or at http://www.perl.com, or a knitting stitch, or the Pathologically Eclectic rubbish Lister, or that other "P" language
00:46.08tzafrir_laptopNewbie___, ask your question, anyway
00:47.43Newbie___tzafrir_laptop: all my calls are now channel to only one source, i would like * to channel to other provider if it matches the area code
00:47.52Newbie___i think is written in perl
00:48.12DEEZEDleast cost routing?
00:49.30Newbie___yes, a LCR  http://pastebin.ca/8754
00:49.40Newbie___thats my original source
00:50.04Newbie___right now , everything goes to span2
00:50.42Newbie___i like * to go to span1 ie if area code is 416
00:51.04*** join/#asterisk mariop (~mariop@201.133.224.253)
00:51.09mariophi
00:51.45DumbDudehi mariop
00:52.28mariophi
00:52.39mariopi'n newbie with asterisk
00:52.42DumbDudewhats up mariop
00:53.00mariopexist some gui to administrate the asterisk
00:53.21iqmariop, yes there are few available
00:53.22*** join/#asterisk tugalone (~tugalong@pcp0010303951pcs.avenel01.nj.comcast.net)
00:53.47JerJer[interop]interop question; does the message waitng indication on SIP need a valid type=peer?
00:53.54mariopwhat you recomend me i need one simple and very very easy to use
00:53.59iqmariop, http://www.voip-info.org/wiki-Asterisk+GUI
00:54.10JerJer[interop]i blieve so, just want to verify
00:54.35tugalonedo sip phones work with asterisk?
00:54.44JerJer[interop]tugalone: um yes
00:54.45Newbie___of course, tugalone
00:55.06iqtugalone, they need to be plugged in and properly configured ;)
00:55.10*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
00:55.13tugalonehow about across nat's?
00:55.18Newbie___hi, neopher
00:56.03iqtugalone, I never experienced any problem. But some people say that they do.
00:56.30Newbie___grrr, GSM channel bank is still not available on ebay
00:57.03JerJer[interop]sip and nat can be made to cooperate but is not perfectly friendly
00:57.36tugaloneiq: hmm okay. where do i find technical information on asterisk? i'm looking for - interfacing with the PSTN network, the protocol that's used for signaling and media and the likes?
00:58.03iqtugalone, http://www.asterisk.org
00:59.36mariopi have a project i need to know if posible withe asterisk i need to coneectr the pstn to the pbx create with asterisk and the extensions management via ip is posible?
01:00.02h3xmariop: only 100 people have done that already :)
01:00.15iqmariop, where are you located?
01:00.52mariopmexico
01:01.08JerJer[interop]better get a license then
01:01.24mariopin this case my PSTN need e1 card
01:01.26iqmariop, technically it is possible. Check to make sure it is not illeagle
01:01.28marioplicense
01:01.28h3xs/license/beer, drugs, hookers for telmex/
01:01.52mariopnop
01:02.25iqmariop, then start reading :)
01:02.26mariopno is ilegal meantime i'm not make me a carrier of telephony
01:02.31marioplike telmex
01:02.36mariopyep
01:02.59mariopthe e1 i see the only company sell is diginum
01:04.35*** join/#asterisk Rick_Hunter (~rhunter@03-111.008.popsite.net)
01:07.09*** join/#asterisk AmaDEE0_ (~foo@209.52.90.78)
01:07.28AmaDEE0_Does anyone have AMP working with Postgresql?
01:07.42h3xamp?
01:07.58shmaltzanybody want to take a look at this:
01:08.00shmaltzhttp://lists.digium.com/pipermail/asterisk-biz/2005-April/004017.html
01:08.09Zipper_32apache mysql and perl?...
01:08.30Newbie___anyone help me on LCR with perl ?
01:08.41h3xleast cost routing?
01:08.50Newbie___h3x: yes
01:08.54h3xinternational?
01:09.12Newbie___internation and domestic
01:09.22h3xim working on a solution for US Domestic
01:09.33h3xyou can't do a NPA/NXX table because of ported numbers
01:09.50h3xwell you can, but 20% of numbers are ported now and you'll misroute calls
01:10.21Newbie___i have not try anything yet, right now all my calls are going to one provider
01:10.43tzangerh3x: there's got to be a solution because it's already in use
01:10.49tzangerh3x: I'm guessing it's ss7 access
01:10.52h3xoh so you are going to send all domestic calls to one place
01:11.09Newbie___for now, all domestic and internation going to one place
01:11.13h3xtzafrir: correct, but nobodys written a module for open source software like asterisk or ser yet
01:11.39Newbie___need to sent few calls based on area code / country code to other provider
01:11.53tzangerwould it not be possible, at least in interim to just dump the NANPA to a table every week or so and run off that
01:12.00h3xnewl: aren't your domestic rates based on LATA by OCN Class?
01:12.05h3xsilly nick completion
01:12.20h3xtzanger: that dosent help with ported numbers.
01:12.41h3xYou can port wireless -> wireline, wireline -> wireline, wireless -> wireless, ilec to clec, etc.
01:12.49tzangerh3x: sure it would (maybe I have my terminology wrong) -- I'm looking at a table that has number mappings and who they should route to
01:12.59h3xif you misroute a call you could be paying $.03/Min instead of like <$0.01/minute
01:13.05tzangerh3x: I understand
01:13.14h3xtzafrir: you have to send a 10 digit query to a ss7 database provider
01:13.26h3xand it responds with the OCN of the place you are calling
01:13.33tzangerh3x: right but do they not publish "as of" lists of the numbers?
01:13.37h3xyou take that OCN and figure out the LATA from the area code
01:13.49h3xtzanger: its not a block of numbers, every number could be ported
01:13.52tzangerI know
01:14.04tzangerbut at the present there are only a small percentage of ported numbers
01:14.07tzangercompared to all of NANPA
01:14.10h3xtake for instance people that port their number from uhm... bellsouth over to vonage, well vonage could be using XO as their underlying carrier for the DID
01:14.18h3xNo way dude, its 20%
01:14.21h3xits terrible
01:14.23tzangerh3x: 20% is still only 1/5
01:14.24tzafrir_laptopmy errors were gone when I removed -I/usr/include/asterisk from cc's command-line
01:14.39h3xthat 20% error could cost you tens of thousands of $ as a carrier
01:14.46tzangerh3x: I'm not saying ignore it
01:14.55tzangerI'm saying you have the "traditional" NANPA mappings to the oCN
01:15.03h3xThe bigger problem is when you are billing your wholesale customers
01:15.04tzangerand then this table of the 1/5 of all NANPA that is "weird"
01:15.06h3xyou could bill the wrong rate
01:15.19tzangernow you do a dump on Monday
01:15.26tzangerand one of those numbers changes on Tuesday
01:15.30tzangeryou'll misbill for  aweek
01:15.39h3xnanpa dosent give you ported number information
01:15.50h3xthey just tell you thousands or ten thousands blocks of who can assign new numbers in those prefixes
01:15.52tzangerh3x: and it's not possible to get that information in an "as of this date" format?
01:15.55h3xNEW numbers
01:16.16h3xthats all NANPAs information means now
01:16.21h3xwho can assign new numbers
01:16.23tzangerh3x: it's not possible to get a list of all ported numbers and their correct OCN in "as of this date" format?
01:16.44h3xSure you can but the SS7 providers charge a setup fee of $2 Million for that information
01:16.45h3xseriously
01:16.45h3xheh
01:16.50tzangerok
01:16.52h3xthats what SNET told me
01:16.56tzangerso that is certainly out of the question
01:17.01h3xSentito or whatever
01:17.03ManxPowerMere mortals can't get that information
01:17.10h3xits a big ass database with millions of records
01:17.10tzangerhow much trouble would you get into for "wardialing" the SS7 database and creating your own list?  :-)
01:17.11h3xLIDB
01:17.22ManxPowerA "mere mortal" is someone with less than $2 million in available money, of course.
01:17.23h3xUh... you have to pay by the query
01:17.31tzangerh3x: I'm kidding
01:17.41tzangerif you can query the SS7 database you may as well just do it right and query it
01:17.58h3xyes, so what im doing is putting together a module for asterisk, ser, etc that will query
01:18.06h3xit queries by going through my ss7 gateways
01:18.17h3xim going to give it away for whatever my cost is so it drives my volume up and brings my cost down
01:18.33tzangerh3x: so what are you complaining about then if you're building it?  :-)
01:18.33h3xthe per query cost that is
01:19.05h3xI'm not complaining about anything, it just sucks that there isnt a solution out there for reasonable software yet
01:19.07*** join/#asterisk santiago (~santiago@63.245.86.85)
01:20.49tzangerh3x: perhaps but you are carving out a business opportunity from that lack of solution
01:20.56h3xof course thats probably because until recently there weren't any IP interfaces to LIDB
01:21.18h3xwell its a matter of aggregating to lower my costs than making money in this case :)
01:22.31bkw_anyone here thats not in the US or Canada sms me?
01:22.37bkw_I wanna test something
01:24.15*** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au)
01:24.59mithrohowdy people, anyone here know how FXS work electrically?
01:26.02ManxPowermithro: how much info do you want?
01:26.13mithroi'm intrested in finding out how the FXS cards provide the required 48V DC and 120V AC 25Hz ringer
01:26.35ManxPoweractually it's 90VAC ringer
01:27.06ManxPowermithro: look at the datasheet for the TigerJet FXS chips
01:27.07mithroyeah, how do they do that from the 12V rail in the computer?
01:27.31ManxPowermithro: they prolly drop the amps and up the voltage.
01:27.42ManxPowerI don't know much, but I do know that's pretty trivial
01:28.02mithroyeah, i know how to do everything apart from that in designing a FXS
01:30.18*** join/#asterisk |nix (~inix@202.148.164.48)
01:31.01*** join/#asterisk PBXtech (~nik@70-58-41-173.slkc.qwest.net)
01:31.03*** join/#asterisk d00gster (~doughant@Toronto-HSE-ppp3661779.sympatico.ca)
01:33.18|nixhey PTG1234, you're around?
01:36.09florzmithro: I dunno how that chip does it or any other FXSes either, but at least some cheaper analog PBXes often simply use 24 V at 50 Hz or something (in europe, that is) simply taken from the grid using some transformer
01:36.27JerJer[interop]uses ohms law
01:36.48*** join/#asterisk zhier (~nick@61.144.20.3)
01:36.53JerJer[interop]takes a lot of current with a transformer to create a lot of voltage
01:36.58JerJer[interop]with little current
01:38.05harryvvJerJer learned all of that in electronics collage. :)
01:39.08zhierhow to register a user to my sever? that is say how to configure my conf file to receive the registratiom.
01:39.12JerJer[interop]more like high-school
01:39.15florzJerJer[interop]: Though, neither is 90 V a lot of voltage nor does it take a lot of current to make a telephone ring =:-)
01:39.19JerJer[interop]then self taught the rest
01:39.21harryvvmithro, stepping up 12 volts though a class a amplifier or transformer.
01:39.37JerJer[interop]florz:  hence why there is a molex power connector on the TDM400P chassis
01:39.42mishehuipx hehe
01:39.59mishehudeprecated protocol.
01:40.06JerJer[interop]and also why one cannot run a small power supply and expect all four phones to ring at the same time
01:40.43tzangerthe bigger problem is that the 12V line on the PCI connector is not very beefy
01:40.53ManxPowermishehu: We are trying to track down and kill open wireless networks in our building
01:40.59tzangerthin traces and big current spikes don't mix
01:41.05harryvvwhats the maxium current the pci will take
01:41.13tzangerthat and those current surges are coupling to all the other lines nearby
01:41.19tzangerharryvv: you'd have to check the spec
01:41.23harryvvyea
01:41.27tzangerbut even motherboards that are PCI2.2 compliant aren't
01:41.32harryvvprobebly a few miliamps
01:42.05tzangerI'd be willing to suggest high dozens to maybe a hundred or so
01:42.09tzangerjust from intuition
01:43.01harryvvyea probebly ;)
01:43.37hermieUSB is ~500 milliamps
01:44.24mishehuManxPower: your building meaning your residence or your employment?
01:44.47tzangerhermie: yeah but that's 500mA at 5V or 3.3V, not 12V
01:45.16harryvvmanx why are you trying to kill it? unauthorized wireless routers being put on your network?
01:45.18harryvvbrb
01:45.34ManxPowermishehu: the office building of my largest customer
01:45.34harryvv5 volt ttl
01:45.40tzangerbut the entire point is that you can't rely on it, which is why the molex connector's there.  hard drives and tape backups are very power hungry devices on the 12V line, especially when starting up
01:45.46tzangerharryvv: USB is not TTL
01:46.00harryvvi see
01:46.19harryvvbut mabey in the ps ?
01:46.53tzangerTTL is a logic level specification, it has nothing to do with power
01:46.54tzangerwell
01:46.59tzangerokay it has something to do with power
01:47.00ManxPowerharryvv: Once I figure it out I'll use arp spoofing and a web page to basically say "Your wireless network is not secured.  Turn on the security features of your wireless devices.  Someone sitting in the parking lot with a laptop and wireless card could be capturing all your data!
01:47.03tzangerbut not how you're using it.  :-)
01:47.44harryvv:)
01:48.08ManxPowerharryvv: We are finding that our users with laptops and wireless cards are trying to send corporate data over other companies wireless networks and we are tired of it.
01:48.23tzangerManxPower: bitchslap them
01:48.30ManxPowerThere is NO reason to have an unsecured wireless network in an office building.
01:48.36tzangerManxPower: we do it
01:48.38tzangerbut
01:48.46tzangerthe wireless network attaches OUTSIDE the firewall, not inside
01:48.50ManxPowertzanger: The coffee shop is going to be pretty pissed if we implement this at other locations.
01:48.56harryvvnot good. alot of that info is considered sensitive? btw dinner. I will tell you of a defence contractor that almost put my freind in prison for what one of the company employees almost did.
01:49.01tzangerso if you're on the wireless network you're also on the VPN to get in to the network
01:49.09ManxPowertzanger: You know that XP will BRIDGE wireless and wireline networks, right?
01:49.16tzangerManxPower: we don't run XP
01:49.40ManxPowertzanger: The company does not own the computers.
01:49.45ManxPowerThe users own their computers.
01:49.52tzangerManxPower: oh yeah I remember you told me that
01:50.01tzangerso they are ACTIVELY trying to send corporate data around?
01:50.40ManxPowertzanger: Whenever I refer to "my largest customer" just imagine someone that does a lot of LSD writing up how a company should be run and you'll have a pretty good idea...
01:50.43ariel_anyone want to spend  $ 2800.00 on a domain name?  www.vitaphone.com  hummmm expensive.
01:50.46ManxPowertzanger: Not that we know of......
01:51.04tzangerManxPower: so why are they hooking up both wireless and wired connections?
01:51.14ManxPowertzanger: because they are idiots
01:51.17tzangerok
01:51.22tzangerthat works. :-)
01:51.38DEEZEDanyone running asterisk on adsl?
01:51.46ManxPowertzanger: The person that generates the most revenue is someone that sells most of the houses to people she meets at the country club.
01:51.47ariel_DEEZED, yes
01:51.59DEEZEDhow many calls can you handle at once?
01:52.10ManxPowerThe male version meets clients while playing golf.
01:52.22hermiegoogle thinks 12v@50mA
01:52.29ariel_DEEZED, it depends on codec. But if I use gsm about 4
01:52.45DEEZED=/
01:52.48hermiewait, i've got it!
01:52.53ManxPowerDEEZED: We have the same questions as the last 300 people that asked the same thing: What is the upload and download bandwidth, what codec will you be using, will you be using trunking?
01:53.01tzanger12V@50mA for what
01:53.13hermie500mA@12V,100mA@-12V for PCI 2.3
01:53.21DEEZED3000/384
01:53.26DEEZEDgsm
01:53.28hermiethat's per slot max
01:53.29tzanger500mA wow they claim that??
01:53.30DEEZEDiax trunk
01:53.37ManxPowerhermie: As Digium discovered a lot of motherboards don't let you draw that much current
01:53.49tzangerI call bullshit, I don't think any motherboard can handle a half an amp on the 12V rail PER SLOT
01:53.51hermietzanger: that's the max allowed
01:54.02tzangermaybe that's what the spec says but I'll bet there isn't a commodity mobo that handles it
01:54.18hermietzanger: most can't handle >120 on the 12v, but that's what the spec says
01:54.23tzanger:-)
01:54.30ManxPowerDEEZED: GSM codec is about 12k + IP overhead of about 18k
01:54.51ManxPowerso 30 or 31.
01:54.59*** join/#asterisk JerJer[mobile] (~nonyobizn@RtrHSTF-FC.hstf.interop.net)
01:55.00*** join/#asterisk robl^ (~robl@dsl093-025-118.hou1.dsl.speakeasy.net)
01:55.13*** join/#asterisk syslod (~yurplsl@65.114.0.198)
01:55.27ManxPowerDEEZED: so about 12 calls if you live in a PERFECT universe and you have no other traffic.
01:55.42ManxPowerIn the universe I live in expect about 6 calls
01:57.04syslodHEAD broken.  /usr/lib/asterisk/modules/chan_iax2.so: undefined symbo
01:57.04syslodl: ast_memcpy_byteswap
01:57.46ariel_GSM is 32 plus 16 to 18 for the ethernet overhead. g729 is about 12 plus.
01:57.58DEEZEDManxPower.. what do you use your pbx for?
01:58.14ManxPowerDEEZED: phone calls
01:58.23tzangerhahaha
01:58.26tzangerI was just about to say that
01:58.54robl^DEEZED: he also runs one of those pay per minute phone sex lines . :)
01:59.07ManxPowerhttp://www.packetizer.com/voip/diagnostics/bandcalc.html
01:59.29ManxPowerrobl^ is my..er...biggest customer!
01:59.51ManxPowerDEEZED: I personally use Asterisk to get free phone calls to my lovers.
01:59.57robl^ManxPower: that's right!  "Press 4 for Teltetubbies in heat."
02:00.11ManxPowerI professionally use Asterisk for free phone calls to my customers and I use it as a PBX for customers
02:00.50ManxPowerariel_: please step away from the alcohol.
02:00.55ManxPowerit's IP overhead, not ethernet overhead
02:01.10DEEZEDic.
02:02.24harryvvarial want to create a voip domain?
02:03.13*** join/#asterisk tessier (~treed@222.253.72.192)
02:05.52*** join/#asterisk |nix (~inix@202.148.164.48)
02:06.22harryvvlets not get layers confused :)
02:06.59zackis it impossible to use chan_bluetooth without rebuilding all of asterisk?
02:08.47ManxPowerMy cat is pretty nice to me when his supper dish is empty.
02:09.17ManxPowerzack: I wasn't aware that chan_bluetooth even worked at all
02:09.30JonR800it doesn't.
02:09.36zackManxPower: hmm, i might care less, then.
02:10.42*** join/#asterisk Slainte (~Slainte@66.55.112.85.ppp.northrock.bm)
02:12.18SlainteAnyone have a working example of SetAccount, in their extensions.conf?   I want anyone who enters a long distance call to enter a four digit code, so we can back charge the client.
02:13.07mishehuiax2 trunking works for all codecs, no?
02:13.11SlainteI dont need to authenticate the code, it just needs to be entered.  (validation to come later)
02:13.25mishehusomebody had posted on the mailing list back in february that he thought that you can't trunk g711ulaw
02:13.33syslodAnyone testing HEAD?
02:13.38Zipper_32For a home setup, do I need a card for use of softphones over a DSL connection?
02:14.08*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
02:14.13*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
02:14.31ManxPowerI'll be in #asterisk-stable if people have questions about 1.0.x Asterisk
02:14.35*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
02:15.21h3xreading scrollback....
02:17.36syslod:( cvs update -D yesterday
02:18.11Newbie___exit
02:18.18h3xZipper_32: Im more pissed off about a customer i could have had that spent $5 Million on an asterisk powered call center%!#
02:18.30Zipper_32sweet jebus...
02:18.37h3xno shit mang
02:18.42Zipper_32That's nuts..
02:18.50Zipper_32How the... wow... just wow.
02:18.58syslod$5 Million???
02:18.59h3xim sure they paid less for the digium commercial license than they did on dinner to take the people out , wine and dine them to buy it
02:19.11h3xYeah it was some non profit orginzation in the midwest
02:19.12jakepdevh3x - I feel your pain
02:19.27Zipper_32<3 for h3x
02:19.42h3xi dont have the <3 to charge
02:19.44h3x$5 mil
02:19.44h3xwell
02:19.57Zipper_32hehe, I"m sure you could... =)
02:20.02h3xI hope they gave all the 300 agents a color cisco phone for that price
02:20.03h3xdamn
02:20.16Zipper_32lol,
02:20.20*** join/#asterisk tessier (~treed@222.253.72.192)
02:20.31h3xjesus, why dont i ever get leads like that before people buy stuff
02:20.56*** join/#asterisk iq[laptop] (~iq@207-224-100-90.omah.qwest.net)
02:21.01DEEZEDnoob question.. what does 5555 mean in "210 => 5555,John Smith,jsmith@yourdomain.com"
02:21.20PBXtechyour saying someone built a $5m * call center?
02:21.39h3xi didnt personally see the system but my counterpart paid them a visit and noticed it had te410 cards in it..
02:21.55h3xPBXtech: yeah supposedly, well im sure it had other things in it too but they got screwed
02:22.01PBXtechfor $5m?
02:22.03PBXtechdamn
02:22.16PBXtechthats like 4.5 profit huh
02:22.18PBXtech:)
02:22.24h3xit aint the first multi million $ call center raping ive heard of though
02:22.33h3xi saw another one a couple years ago for century 21 here in vegas
02:22.48h3xthey put in a whole 60 cisco 7960 phones
02:22.50PBXtechcome up to SLC call center central here
02:22.57h3xran that into cisco catalysts, and then cisco
02:23.03h3xer.. as5300s
02:23.06DEEZEDI use to work for convergys in SLC
02:23.11h3xand then from the 5300s into shitty ISA dialogic based dialer
02:23.22h3xthat system sold for $2.5M and the guys never got it working completely
02:23.25PBXtechive walked into convergsys :)
02:23.27h3xand they all moved to hawaii
02:23.32h3xand disappeared
02:23.37|nixanyone using PT1234's SIP code?
02:23.39h3xthe customer cant find em, and went under
02:24.13h3xyou wouldnt believe the sales pitch they used to sell it having VoIP capability
02:24.13PBXtechyoud think for 2.5m you'd do your homework
02:24.27h3xthe reason they decided to buy it is because "you can transfer both voice AND data!%#!!#%  to another party in another office!"
02:24.30h3xwell no shit you can do that with tdm too
02:24.37Zipper_32How much would you guys charge for a 9-line asterisk system with support for 20-25 phones?
02:24.38h3xbut it dosent matter coz they never got it to work
02:24.39PBXtechi lose $100k deals cause i dont have enough employees (which is BS)
02:24.50h3xZipper_32:  Uhmm.. how does $1.9 million sound? hahah
02:25.10h3xPBXtech: just hire some india call center for $2 an hour
02:25.17Zipper_32h3x: .... How about no? =)
02:25.32PBXtechHaHaHa staff of 100 then huh
02:25.33h3xDamn I always lose to higher bidders!!!
02:25.38Zipper_32hehe
02:26.04Zipper_32I'm trying to learn how to set this stuff up, I just finished installing asterisk today.
02:26.17h3xwhere are you at
02:26.43syslodNext time someone sees a $5 million bid I'll be happy to send you all the staff you need.
02:26.44harryvvh3x on ebay? what you loose at the last second?
02:26.48DEEZEDsorry to ask again. but what does 5555 mean in "210 => 5555,John Smith,jsmith@yourdomain.com". Is it the phone extension? If so, then can i just put any number if i don't want it linked to an actual phone?
02:27.09PBXtechDEEZED thats the password
02:27.10SlainteDEEZ  password for voicemail
02:27.16DEEZEDthanks guys
02:27.41Zipper_32Well, so far h3x, I've setup asterisk on a redhat 8 box, and that's it.   I am going to order some cards, but I'm reading up and trying to figure out what I need/.
02:28.01h3xoh telus. .. hmm
02:28.11h3xare you gonna use e1s?
02:28.59harryvvtelus your complaints :)
02:29.25h3xshit, bitchx would never port to atari 2600
02:29.39h3xthey had enough problems getting it to bsd
02:29.51harryvvsomone shot at a telus worker because we were suffenly hit by very poor service in this area about a year ago.
02:30.03harryvvsuddenly that is :)
02:30.06h3xdaamn
02:30.33h3xdo you got decent aussie did providers down there
02:30.36harryvvyea it was affecting all of vancouver. Poor service month or more to get a phone line everything.
02:30.43h3xer canada
02:30.43h3xdoh
02:30.50h3xsmoking too much crack
02:30.52Zipper_32hehe
02:30.58Zipper_32I don't know what we're going to use yet
02:31.06h3xwhy dont you use primus did's
02:31.11syslodThey sell guns in CA? :)
02:31.15h3xthen you dont need phone lines
02:31.37PBXtechare there any providers that will carry 100+ dids for a comercial co?
02:31.49h3xPBXtech: where
02:32.00PBXtechhuh i said are there any..
02:32.06h3x(i should have just said 'yes')
02:32.06h3xheh
02:32.18h3xYes, but its usually by the minute
02:32.32h3xsince thats the only good way to measure the usage without physical lines
02:32.37h3xits cheap though
02:32.44PBXtechhow cheep/ who
02:33.12syslodPBXtech.  Any provider will do that.
02:33.12h3xunfortunately most of the telcos that actually do it with VoIP only do it under wholesale to resellers like us
02:33.30h3xand all of their coverage areas suck
02:33.38*** part/#asterisk santiago (~santiago@63.245.86.85)
02:33.39h3xso you have to use them all if you want really good coverage
02:34.37PBXtechhow do you find out who is providing local DIDs to these carrier? that possible
02:34.51syslodYou want to find what?
02:35.12h3xthe largest ones are level3, mci, and maybe xo
02:35.14PBXtechthe local company who is providing DID blocks to like broadvoice
02:35.30PBXtechof IC
02:35.33syslodDo you have a NPA/NXX in particluar you want to find a carrier for?
02:35.48h3xyou look up the NANPA assignment of the npa/nxx
02:36.14syslodLERG has a better listing.
02:36.32h3xyeah
02:36.34*** part/#asterisk zack (~zack@sebastian.redhat.com)
02:36.55PBXtechwhat LERG
02:37.08syslodWhat are the resellers doing about 911 and CALEA
02:37.18syslodLocal Excahnge Routing Guide.
02:37.31h3xit dosent really matter that much if you find out who it is anyway, many of the providers require a $25k a month committment to even get a DID product from them
02:38.09syslodLevel 3 is almost impossible to deal with unless you are a larger carrier.
02:38.15h3xsyslod: CALEA isn't required for VoIP yet
02:38.42h3xlevel3's prices suck anyway, i am going to use them but i found several other ones combined together that equals their footprint
02:38.55PBXtechh3x you new to this channel?
02:38.57harryvvh3x what to high?
02:39.04h3xnah i was just gone for quite a while
02:39.09syslodWell kinda.  See if you purchase DID blocks and digital service you still are technically a CALEA product.
02:39.23harryvvsyslod thanks for saying that about level 3 was going to give them a call.
02:39.36h3xsyslod: the underlying carrier that converts to TDM provides CALEA
02:40.02syslodYea but its all screwed up in the database.  It has the wrong end user information.
02:40.34h3xharryvv: theres a couple problems with level3, they have the worst volume committment (you need to do $50k in business a month with them to get decent rates) and they have a proprietary voip gateway platform called Viper
02:40.41h3xit has some issues interfacing to asterisk
02:40.42syslodharryvv: What are you looking for? Nationwide or select markets?
02:40.49h3xone of my guys spent 3 months patching asterisk to make it work
02:41.32PBXtechyou resell LD and * box's?
02:42.09h3xwho me?
02:42.12h3xwww.carrierone.net is my company
02:42.14PBXtechya
02:42.28h3xive been using asterisk for a couple years
02:42.40h3xbut for the DIDs and core, we're deploying SER
02:44.07h3xamong our market niches is giving away free voip gateways to customers doing a lot of traffic on tdm t1's
02:44.31h3xcarrying trunked voip over private line
02:45.00h3xwe might be switching to a DSP based product soon for CPE though
02:45.37PBXtechthere are good non * gateways out there that are good price
02:45.43shmaltzanybody want to take a look at this:
02:45.44shmaltzhttp://lists.digium.com/pipermail/asterisk-biz/2005-April/004017.html
02:45.47syslodh3x: Compressed voice?
02:45.55h3xyes, compressed and trunked
02:46.16h3xwe've got some deals with various IXCs that give us flat rate private line to anywhere in the US
02:46.23h3xits actually cheaper than dedicated internet most of the time
02:46.24BoRiSwhat about canada?
02:46.28syslodshmaltz: What about it?
02:46.37PBXtechh3x can you PM me your email ide like to talk about this further when im at work
02:46.43shmaltzI'm looking to pay for this
02:46.47shmaltzsyslod
02:46.54h3xyes, we'll have canadian DIDs in the near future from a carrier out in one wilshire
02:46.56syslodshmaltz: We have this.
02:47.04shmaltzhmmmmmmmm, pm please
02:47.24h3xwe've got our own datacenter here in vegas with private fiber cross connects to our carriers
02:48.09*** join/#asterisk munchausen (~oihsafd@68.71.213-37.atlsfl.adelphia.net)
02:48.10*** join/#asterisk esandeen (~sandeen@sandeen.net)
02:49.33PBXtechide to biz with you just to have to come to las vegas for a tour :)
02:49.44munchausennewbie question: is it possible to "forward" a connected audio stream to a new destination so that the packets can go directly to the new destination instead of bouncing through the old?
02:50.06h3xhahaha
02:50.10h3xdid i pick a good location or what.
02:51.30BoRiShmm
02:52.20bjohnsonmunchausen: yes.  reinvite in sip
02:52.43*** join/#asterisk malverian (~malverian@adsl-065-005-207-210.sip.gnv.bellsouth.net)
02:52.50munchausenthanks
02:52.51malverianHmm..
02:52.55Dr-Linuxanybody know about SIPPS softclient ?
02:53.03malverianI'm having weird lockups when I use my sk98 with my w100p card.
02:53.40malverianI get lock-ups when I enable the wcfxo and my sk98lin modules together. I saw 2 posts on the mailing list regarding this (unanswered)
02:53.57*** join/#asterisk rumba (~ropawa@cpe-68-201-148-205.sw.res.rr.com)
02:54.14*** join/#asterisk Ferrari (~IPlexbyVe@216.196.255.42)
02:54.24*** part/#asterisk Ferrari (~IPlexbyVe@216.196.255.42)
03:00.02bjohnsonh3x: ny hints at what the costs are for your services?
03:00.08bjohnsonnothing on the web site
03:00.26h3xit all depends on the location really
03:00.35h3xtheres so many underlying carriers its difficult to determine fixed costs
03:00.40h3xwe're mostly wholesale though
03:01.03munchausenbjohnson: do people successfully use reinvite to loop a call out through the same provider it came in on?
03:01.06h3xI don't think I'll really ever have a public DID service, but ill let you guys test it as markets become available though
03:01.24PBXtechso h3x for 1M minutes we comp'd to "splash"  :)
03:01.43*** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-93.modem.logical.net)
03:01.50h3xhaha, i think theres some better shows than that ;)
03:02.00h3xmy friend's wife is the lead dancer in Jubilee...
03:02.06Carp1how do you retrieve voicemail passwords?
03:02.10PBXtechnice
03:02.20PBXtech[Carp1]: voicemail.conf
03:02.24h3xThe dumb thing is i still haven't seen that show
03:02.39PBXtechhah its your friends wife
03:02.40Carp1Sorry, I havnt been able to use asterisk in like 4 months
03:02.42opus_does anyone know about verisign ss7?
03:02.43Carp1system went down
03:02.46Carp1just got it back up
03:02.46PBXtechgot to support that :)
03:02.51h3xwell
03:03.01PBXtech:/
03:03.17syslodverisign ss7 sucks
03:03.22shmaltzanybody want to take a look at this:
03:03.24shmaltzhttp://lists.digium.com/pipermail/asterisk-biz/2005-April/004017.html
03:03.38PBXtechshmaltz you on repeat mode?
03:03.55opus_syslod - how does it work? do they have a price online, perhaps you might know it?
03:03.55shmaltzPBXtech, almost
03:03.57bjohnsonmunchausen: no
03:04.09shmaltzI'm a bit late to get this up and running
03:04.09PBXtechheh
03:04.13bjohnsonmunchausen: only works between sip clients you control
03:04.29syslodopus_: Its almost impossible to get in touch with them.  Once you do they'll miss the first 6 months of FOC and you'll give up.
03:04.32h3xyeah well, verisign for ss7 is like using h323 for voip
03:04.56PBXtechif * had an interface that nice, then we wouldnt have anything to do in here
03:04.58syslodopus_: What are you looking for specfically from verisign?  And why them?
03:05.17opus_syslod - I'm trying to not buy a FXS card
03:05.18h3xI met a systems engineer with verisign that knew his shit
03:05.18*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr)
03:05.22h3xi have his card somewhere
03:05.26syslodHuh???
03:05.44syslodIlliumnet
03:05.49h3xtheir solutions are too complicated though
03:06.00opus_i mean, never mind.  Thats what I am trying to do.  I just came across verisign and wanted to know about it, completely seperate
03:06.03syslodopus_: Explain?
03:06.25syslodopus_: You trying to set cid?
03:06.44bjohnsonopus_: trying not to buy a fxo?  just use a normal voip provider
03:06.56opus_No, I'm just looking for a cheap FXS card. sorry for the confusion
03:07.21h3xwhats verisign ss7 have to do with fxs cards
03:07.31opus_Nothing, sorry for the confusion.
03:07.43bjohnsonopus_: spa 2000 is cheapest per port next to linksys pap2-na (if you can find them)
03:07.47syslodh3x: Are you a carrier or reseller?
03:07.52opus_thanks
03:07.52bjohnsonboth are 2 port fxs
03:08.06opus_why would the linksys pap2-na be hard to find?
03:08.12h3xsyslod: Whats the difference? :)
03:08.18PBXtechunlocked
03:08.25bjohnsonopus_: time for you to go shopping
03:08.37h3xafter the incest i've seen in telcos here im not sure who exactly really carries the traffic anyway
03:08.38opus_here is an off the wall question.. do all FXS cards support rotary?
03:08.42syslodh3x: You have to file a 499 or you don't
03:08.56bjohnsonopus_: I don't know if ANY support rotary
03:09.08h3xyou have to file a 499A exemption if you are an ESP :)
03:09.23h3xanyway, yes we're an enhanced service provider
03:09.31h3xwe dont have any carrier licenses, we don't want any
03:09.42syslodOwn any last mile?
03:10.21h3xnoley, we use the rediculiously cheap wholesale prices of all the IXCs we're directly cross connected to
03:10.25h3xs/noley/no/
03:10.53h3xthats a terrible business to be in
03:11.06PBXtechjust ask XO :)
03:11.07h3xwhen I can get a T1 private line from vegas to new york with the local loop for under $500
03:11.20h3xwhy the hell would i want to spend millions building a network
03:11.21syslodThat doesn't work everywhere.
03:11.23opus_nobody has been able to hack the linksys yet?
03:11.28h3xit works damn near everywhere
03:11.32*** join/#asterisk Newbie___ (~me@218.111.224.175)
03:11.41*** join/#asterisk CoffeeIV (rgr@cpe-70-112-100-20.austin.res.rr.com)
03:11.50Newbie___hi, anyone help me on LCR with perl please ?
03:11.52PBXtech[opus_]: openWRT
03:11.56h3xi have had to get fiber constructed for customers before that are in BFE
03:11.57opus_this sucks: **You must be buying at least 5 of these units to continue with the checkout process**
03:12.09h3xordering fiber construction from a railroad company or something, but they had to pay for it
03:12.11harryvvh3x thats nice
03:12.12harryvv:)
03:12.20bkw_grrrrrrrrrrrrrrreat
03:12.32PBXtechits tony the tiger
03:12.33syslodh3x: Most rural areas that are run by independants etc you'll pay $500 to get to customer from the local tandem.
03:12.36bkw_wish I could solve this stupid issue with usb modems in linux
03:12.36h3xsometimes you can buy distressed fiber from a bankrupt company and pay to get it spliced
03:12.43harryvvh3x when you first started out how did you obtain the money to start the biz
03:12.51h3xsyslod: I usually find a way to bypass the ILEC
03:13.05syslodYea.  Me too.
03:13.06h3xyes even in rural areas
03:13.16h3xbut it really dosent matter that much with compressed voice anyway
03:13.30h3xif somebody need 120 channels of VoIP, if a loop costs $1000 a month who cares
03:13.33h3xit really isnt that bad
03:14.05CoffeeIVwhat is a softphone equivalent to XLite that I can use on linux ?  will ohphone work with asterisk ?
03:14.22h3xharryvv: well we originally developed CTI software with aculab prosody cards on solaris, then we acted as a master agency selling telecom services for the past almost 5 years
03:14.24harryvv120 channels for 1000? nice. I was quoted 23 for 600 bucks here in town.
03:14.32h3xcommission checks from that -> carrier one facility
03:14.41syslodbkw_: Is your valetpark still supported/available?
03:15.26*** join/#asterisk PTG1234 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
03:15.32PBXtechexcept $1000 for 5/T1 equivelant isnt much savings.
03:16.05h3xYeah but that $1000 isnt shit to a customer thats spending $10ka  month in long distance
03:16.31harryvv600 dollars for 1 T-1 here via thinktel/alstream
03:17.27PBXtechyou got to be sub 2 cents
03:17.30BoRiS$650-$800 for a T1 via grouptelecom
03:17.40file[laptop]yay BoRiS!
03:17.54h3xyou wouldn't believe it but many call centers etc. are still paying over .02/Min
03:17.56h3xfor TDM
03:18.04syslodYour talking about different things.  A T1 loop is cheaper usually than end to end service.
03:18.08h3xplus the taxes
03:18.16PBXtechplus the tax is KEY
03:18.26PBXtechF-in 14% ish
03:18.28harryvvh3x are these like call centers linking india with the states?
03:18.28h3xThe point i was making is that long haul + local loop is usually cheaper than internet access on both sides plus a t1 loop
03:18.44h3xharryvv: no, india call centers want to pay like .000002/Min
03:18.48harryvvhahaha
03:18.53PBXtechh3x true
03:18.53h3xare you kidding
03:18.56harryvvyour joking right?
03:18.56h3xthey make $2 an hour
03:18.59h3xYes
03:19.00PTG1234did you guys see what happened with those soyo ip gateways?
03:19.04harryvv:)
03:19.18*** join/#asterisk eaperezh (~eaperezh@200.75.242.202)
03:19.21h3xthey probably look at voipjet's .013/Min advertised price and email to ask for .007/Min flat
03:19.28h3xwith a 50/50 rboc ratio
03:19.34file[laptop]HA
03:19.37file[laptop]a 50/50 rboc ratio
03:19.40h3xyeah hahahaha
03:19.47opus_PTG - yeah
03:19.49Slainteanyone in here happen to know postfix?
03:19.51opus_PTG -- what the fuck??
03:19.56harryvvh3z i see voipjets lag at 2000 ms
03:19.59harryvvalot of times.
03:20.03h3xindians have to call BFE rural areas to sell widgets
03:20.09PTG1234they freaken told me they had the wrong specs up.. did they with you opus?
03:20.12h3xharryvv: holy cow
03:20.20PBXtechi have a 30ms ping to their LA side
03:20.22opus_PTG -- yup, shentech.com ??
03:20.23PBXtechfast
03:20.35h3x30ms?
03:20.42PTG1234opus_: yep.. i said your bait and switching me which is illegal in the US :P)
03:20.42h3xI have 3ms time to Qwest's voip gateway
03:20.45h3xfrom my DS3 on Wiltel
03:20.52h3xer wait, i think its like 6ms
03:21.02opus_PTG -- i'm having my friends in china locate the factory.
03:21.04PBXtechthats from ELI T1
03:21.09PBXtechin SLC
03:21.11h3xew, eli.. hmm
03:21.15PBXtechheh
03:21.20PTG1234opus_: let me know what you find out
03:21.20h3xthey used to be the bomb in data
03:21.21opus_PTG -- I still want one, and I still want that price
03:21.25PTG1234opus_: no one seems to sell them here
03:21.33PTG1234opus_: i am just curious what the real price of them is
03:21.39h3xvegas is the only market they dont have voice in which sucks
03:21.40opus_$399 i think
03:21.45PBXtechi got a killer rate on local PRI. $450 with FX from 3 rate areas
03:21.48h3xthey are cheap as hell on voice, wholesale anyway
03:21.58PTG1234opus_: thats insane
03:22.02harryvvPBX who
03:22.04PBXtechELI
03:22.16harryvvthats about as cheap as it gets for t1 right?
03:22.17PBXtecherr $550
03:22.18h3xtype II loop?
03:22.22opus_PTG - what I want to know is how 3 different vendors all had the same problem
03:22.24PBXtechyea
03:22.34harryvv600 here for one company thats CDN rates
03:22.36h3xnoley, xspedius sells type II looped vegas T1s here for as low as like $300
03:22.38PTG1234opus_: told you they are all resellers of same place :)
03:22.40PBXtech400 base then 150 for the extra rate centers
03:22.40h3xdamn nick complete
03:22.44opus_whats a Type II loop PRI?
03:22.46h3xvoice
03:22.47h3xyes
03:22.56h3xtype II means they have to use the ILEC to deliver your circuit
03:23.04PBXtechELI uses Qwest to deliver
03:23.07h3xtype I means you are on-net
03:23.17harryvvI see
03:23.18*** join/#asterisk scubasteve (~steve@cpe-024-088-248-113.nc.res.rr.com)
03:23.27*** join/#asterisk mezzmor (~mezzmor@adsl-068-209-180-119.sip.mco.bellsouth.net)
03:23.51h3xmy building has xspedius fiber in it, i have a DS3 cross connect to their OC-3 ADM
03:23.51opus_h3x that means they are going through some other company?
03:23.51PBXtechnever heard of xspedius
03:23.52harryvvh3x data in for voip pstn out and that is called type II also?
03:23.53h3xopus_: yes which they have to do 99% of the time because they dont have fiber everywhere
03:24.12mezzmorI am having crazy problems with voicemail. Anyone else having weird problems?
03:24.16opus_if I purchase a PRI, how do I update the DNIS?
03:24.25PBXtechfrom the carrier
03:24.46opus_hmmm. nobody will let me update myself, say like, at 2am on sunday? :)
03:24.48harryvvmezzmor, let me guess it sound like the comedian mail female voice does not finish speaking one word before she speaks the next one?
03:24.50h3xupdate ?
03:25.01mezzmorWhen MWI works, I cant retrieve messages. When I can retrieve messages, MWI doesnt work.
03:25.09mezzmorIts crazy.
03:25.31harryvvmmm
03:25.35harryvvnot seen that one.
03:25.57opus_hmmm. it looks like all nufone does is that they have a PRI in michigan, and thats it... does anybody use them? is that the case?
03:26.15PBXtechprobably for inbound DID
03:26.28opus_then, they spoof caller id
03:26.39file[laptop]they don't spoof caller id
03:26.46PBXtechdont see why they would
03:26.47h3xopus_: thats probably because they get paid for inbound calls in michigan from being a CLEC, why encourage anything else? :D
03:27.45Carp1I dont really know linux, I downloaded a tar.gz file...how to I uncompress and install?
03:27.53PBXtechHaHaHa
03:27.58opus_If I just bought a PRI, sold SIP/IAX accounts, and let everyone spoof caller id .. I think thats their business, am I wrong?
03:28.00h3xhonestly though, jerjer insists on TDM handoffs from everybody, so its really cost prohibitive to have DIDs all over the place
03:28.02harryvvohh boy
03:28.13esandeencarp1: tar xvzf <file>... and then start reading
03:28.26h3xyoud need colo space, private line, etc etc.
03:28.27PBXtechless INSTALL
03:28.29PBXtech:)
03:28.37opus_oh yeah, INSTALL
03:28.39mishehutzanger: you around?
03:28.58h3xI was going to do all that, and even do all VoIP to TDM transitions for my major IXCs
03:29.17esandeenhey I have a slightly less newbie question... is there any sort of asterisk howto, or a "what you can do with asterisk" kind of page....
03:29.17opus_well, it seems like an obvious business plan..
03:29.20h3xbut then ive found out that the telcos out there got some spendy $$$$$$$$$ gateways that do a hell of a lot better job than asterisk
03:29.28h3xfrom doing interop tests
03:29.34h3xits a pain in the ass to set up initially
03:29.53PBXtechesandeen read the wiki
03:29.56h3xbut i think we're so much better off relaying voip traffic over private line to the carrier's SBC
03:30.14h3xFor instance, I can G.711 fax at 14.4K to Qwest over the public internet
03:30.24h3xwhereas going to my own asterisk gateway over a LAN it dosent work
03:30.28h3xeven with everything set up right
03:30.33opus_why is that?
03:30.37esandeenPBXtech, found it, thanks
03:30.38h3xwell it does work but it renegotiates to 9600
03:30.52|nixPTG1234: you're around?
03:31.08*** join/#asterisk blitzrage (~blitzrage@blitzrage.documenter.extraordinaire.of.asterisk)
03:31.17*** join/#asterisk Dovid (~hirisk@pool-151-198-15-126.mad.east.verizon.net)
03:31.18blitzragehttp://pastebin.com/266765
03:31.26smurfixAnybody with zaphfc experience around?
03:31.27h3xI'm not quite sure, i think maybe the other load from doing many channels screws it up
03:31.38blitzragefor some reason, I have a phone that won't register. The proxy is sending a 401 back...
03:31.51blitzrageI have checked the settings in sip.conf, and the phone, and I'm stumped as to where to look next
03:32.03h3xFurthermore, T.38 fax will eventually work once the carriers turn it on and theres a decent ATA that supports it
03:32.07opus_h3x - if asterisk was just doing that call, and only that call, would it wowkr?
03:32.21harryvvwhat phone
03:32.30blitzragefax will work with spandsp at about 9600 and ulaw
03:32.33h3xopus_: Come to think of it i did do it in the middle of the night once with no calls
03:32.41h3xblitzrage: I meant sending a fax from an ATA
03:32.43PTG1234nix: yah i am whats up
03:32.45h3xto an asterisk box
03:32.45PTG1234msg me
03:32.47blitzrageh3x: ah :)
03:32.49h3xthat goes to TDM
03:32.54h3xwith a zaptel device
03:32.58h3xlike a t1 card
03:33.07harryvvI need to test my spa for fax capability
03:33.08h3xperhaps i need to adjust some things like the um
03:33.09blitzrageI have a SIP device that is getting rejected, what can I do to debug why?
03:33.16BoRiSh3x: I emailed sipura and received a response from them (suprisingly quickly) and they said they expect T.38 support for their spa-2100 in Q2 of 2005. (atleast thats what they said)
03:33.27h3xharryvv: my spa-2100 is what i was testing fax with ulaw, bt it does not have T.38 yet
03:33.27*** join/#asterisk ubergoober (~ubergoobe@c-24-16-110-117.hsd1.ca.comcast.net)
03:33.31h3xBoRiS: yeah i know
03:33.32PTG1234blitzrage sip debug
03:33.34opus_if ( fax) then use_dedicate_p4_asterisk_just_for_fax() ;  :)
03:33.45h3xIf I run the call through asterisk it will probably not support T.38
03:33.46harryvv2100?
03:33.58harryvvI know the 1k and 2k are almost identical
03:34.06BoRiS2100=2k + router
03:34.10harryvvahh
03:34.16h3xyeah it can do nat to the PC port
03:34.19h3xand QoS
03:34.29h3xwhat i dont know is why didnt they put a ethernet switch in it too
03:35.07*** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net)
03:35.21blitzragePTG1234: I'm doing that: http://pastebin.com/266765 - I just see 401 Unauthorized and I don't know why its not authorized. I can't see the authorization credentials
03:35.36blitzrageso I'm stuck as to where to debug next
03:35.53h3xby the way in my test scenario the 14.4k fax on ulaw was going spa2100 -> same asterisk box -> qwest voip
03:35.56harryvvbtw, I still need somone to test a remote softphone client with my system. all rtp stun and sip ports open but end user could not hear my voice. Could see him accessing vm but he could not hear anything. I suspect his xp firewall may be blocking it by default?
03:36.10h3xon sip
03:36.15PTG1234blitzrage: sounds like password or login is incorrect
03:36.16harryvvyea
03:36.23h3xThe IAXy is HORRID with fax
03:36.26h3xit wont even work on a lan
03:36.35blitzragePTG1234: yah, thats what I thought... but I've checked the username and password.... very simple...
03:36.41harryvvh3x thanks for that info :)
03:36.43blitzrageI even re-entered them in the phone to verify
03:36.47h3xi think maybe the jitter buffer needs to be adjusted or something
03:36.53BoRiSI could do faxing with G711 within a lan using my SPA2100 no problems...Going out to the internet...died 95% of the time
03:37.18h3xBoRiS: I can do 14.4k fax over the sipura to asterisk to voip providers, as long as its not on a cablemodem
03:37.20h3xDSL works just fine
03:37.21harryvvboris ever get that resolved?
03:37.41h3xI bet maybe my problem coulda been fixed with gain settings
03:37.43blitzrageand the context for the friend exists
03:37.44ubergooberCan * randomize music on hold?
03:37.47file[laptop]which it then went to LA to TDM
03:37.56h3xfile[laptop]: what, spandsp?
03:38.02file[laptop]nope I used a fax machine
03:38.06file[laptop]I hooked it up to my PAP2-NA
03:38.06h3xoh ok
03:38.25file[laptop]I had to fax in some forms for the box that I just got in San Jose ironically
03:38.26h3xOne guy I have coloed at my DC wanted fax, and he spent forever messing with spandsp
03:38.32h3xhes using asterisk obviously
03:38.49h3xinstead, after trial and many errrors (mostly problems with compatbility with fax machines on the other end)
03:38.59h3xi talked him into putting a modem in his box and using a pots line off my channelbank :)
03:39.03h3xhylafax rules
03:39.07|nixhey PTG1234, i've messaged you
03:39.20opus_spandsp has compatability issues? I kind of thought so... what kind of problems did he run into?
03:39.39h3xcertain fax machines it wont negotiate with and of course theres that streaking problem with HP fax machines
03:40.00opus_weird.
03:40.21opus_isn't there only one guy who works on spandsp? The orig coder himself..?
03:40.31opus_perhaps if you ordered him a pizza and sent him the fax he'd do it :)
03:40.34h3xYeah its steve, hes a very talented guy
03:40.35h3xcoppice
03:40.49h3xi think hes probably just busy with other things
03:40.55h3xfax takes a hell of a lot of work to perfect
03:41.00PBXtechspandsp pre11 solved a lot of my problems but still a tiny amount of problems. but MUCH better than previous
03:41.34h3xwell im sure its fine if you use it with your own fax machines, but somebody trying to blast out faxes for like say, a food.com clone it dosent work too well
03:42.02PBXtechblast with spandsp  hmmm :)
03:42.14h3xI wish there was a good t.38 soft modem
03:42.18harryvvhylafax can work with windows bases fax clients?
03:42.20h3xbesides t38modem in openh323
03:42.24h3xyes
03:42.26h3xtheres lots of them
03:42.29harryvvokay
03:42.30harryvv:)
03:42.39PBXtechhylafax rocks
03:42.52harryvvwhat modems does it work with
03:42.56h3xas sad as it is, i am really thinking about setting up a dozen modems on a channelbank and selling hylafax ports
03:42.56h3xhehe
03:42.56harryvvi have a number of modems
03:43.01PBXtechi got a NMS board with 64 DSP's that would be a good blaster box :)
03:43.12h3xheh
03:43.31h3xtheres a lot of modems it works with even some $4 winmodems
03:43.38harryvvokay thats cool
03:43.39h3xim prob gonna use like USRs
03:43.50bkw_PBXtech, duh
03:44.10bkw_I also luv rxfax
03:44.22h3xwe all love bkw
03:44.27PBXtechi love rxfax just not 100% solid
03:44.58opus_dude, there is a $4k bounty on t.38
03:45.17h3xshit, it needs a $40k bounty to put up with that crap
03:45.19h3xehehhe
03:46.45opus_how does hylafax work
03:46.47opus_?
03:46.52PBXtechhylafax.org
03:47.26*** join/#asterisk finejava (~abc@218.208.119.98)
03:47.32opus_can you receieve a fax with it?
03:47.36finejavahi guys
03:47.41PBXtechyup
03:47.45finejavaany1 out there can give me a hand
03:47.52PTG1234just use app_rfax
03:48.10Newbie___hi, anyone help me on LCR with perl please ?
03:48.10finejavahow can i check how many user which r logon to the asterisk PBX?
03:48.16opus_app_rfax is a hylafax module? I thought it was spandsp
03:48.24PTG1234no asterisk module
03:48.29PTG1234no need to use hylafax really
03:48.35PBXtechlogin to what console?
03:49.14Zipper_32One quick question, how does one use their home DSL connection with asterisk to place calls? (Isn't this similar to an office using an E1?)
03:49.21finejavai need a count of how many user is logon...anyway or command which i can use???
03:49.32PBXtech[Zipper_32]: pick a VoIP provider
03:49.51PBXtechlogin to what? SIP? CLI?
03:49.59finejavaCLI
03:50.08finejavai mean logon to SIP
03:50.15finejavabut how i can i check from CLI
03:50.16PBXtechsip show peers
03:50.24finejavathen will show all the user
03:50.28PBXtechyup
03:50.32opus_pbxtech what is a nms board?
03:50.49finejavai just wan to know how many user is logon with the status 'OK'
03:50.50PBXtechexpensive T1 boards
03:50.55h3xwell, the max tnt 11.0 TAOS supports T.38 on SIP
03:51.03h3xthat is encouraging
03:51.03opus_hmmm.
03:51.13PBXtech[finejava]: show sip peers
03:51.18h3xat least i could set up a endpoint to test
03:51.26opus_pbxtech -- somebody should hack up a t1 board/fpga and put it on opencores.org
03:51.53opus_is the circuitry used in a FXS really that complicated?
03:52.02h3xits called the zapata T400P/E400P :P
03:52.03finejavashow sip peers????
03:52.03malverianHey guys.. if I'm using a W100P card with the analog line plugged into the card and a line from the card to a normal phone.
03:52.05opus_Isn't it just an FXO with a ringer/ circuit?
03:52.09finejavais there such command
03:52.15malverianIs there any way to make an extension ring through to the phone?
03:52.18MajestikI've got some codec weirdness, It doesn't seem to behave with the priority I'm trying to set..
03:52.19PBXtechfinejava type that in the CLI
03:52.30opus_I thought the T400P was FXO
03:52.37finejavayeah...i did...no such command
03:52.38opus_nvmind
03:52.39PBXtech[malverian]: whatcha mean?
03:52.44h3xT400P is quad T1
03:52.45finejava'show sip peers'
03:53.05PBXtechman this irc client i wrote forever ago has lots of colors in it. annoying. have to fix it
03:53.13malverianPBXtech: I want asterisk to pick up the line after 1 ring (done), and if they press for example... "1234" I want it to ring at the line that is plugged into the phone part of the zaptel modem.
03:53.38malverianI want to basically just ring the telephone line like would happen if asterisk hadn't picked up at all.
03:53.43PBXtechthats basic
03:53.48malverianYay :)
03:53.49PTG1234opus: you know a guy was selling them on ebay cheap
03:53.51malverianI'm glad it's easy.
03:54.03PTG1234two people were
03:54.07opus_ptg - yeah..
03:54.23opus_wonder what happened with that
03:55.04malverianI tried making it dial(Phone/phone0) but it doesn't seem to work!
03:55.09PBXtechmalverian you just have the incoming FXO line answer play a background message and have 1234 dial the FXS ZAP circuit
03:55.30PTG1234opus: i am emailing them
03:56.20opus_PTG -- i think shentech.com is short for shenzhen china..
03:56.31malverianPBXtech: I'm having it play a background message, but what is the correct command to have it go to the zap circuit?
03:56.35Zipper_32PBXtech: This is my first day with Asterisk, and I completely understood that message.
03:56.47Zipper_32I'm so excited...
03:56.49opus_haha
03:56.50Zipper_32=)
03:56.58opus_what was your question?
03:57.22PBXtechexten => 1234,1,Dial(Zap/2)
03:57.29malverianAh...
03:57.30malverianThanks man :)
03:57.37PBXtechassuming FXS is #2
03:58.01shmaltzhttp://bugs.digium.com/bug_view_page.php?bug_id=0002905
03:58.03shmaltzin the above bug will revision 5 have everything from the previous ones as well?
03:58.11PBXtechive been into asterisk for a year now and this still excites me :)
03:58.20Zipper_32If anyone wants to help me though, I'm trying to figure out what kind of equipment I'll need for a 4 line office with 12-15 extensions.
03:58.27opus_Hmmm.. I  think I'm going to buy one of these linksys pap's and see if they have a JTAG port
03:58.35h3xZipper_32: your # of lines is going down .. heh
03:58.39PBXtech[Zipper_32]: quad FXO card
03:58.53h3xopus_: haha what do you need jtag for on that
03:58.57Zipper_32h3x: The main location is 9 lines.
03:59.17opus_h3x - i hear if you upload the XML to it, it reboots, it goes back to vonage
03:59.19Zipper_32h3x: this is a new site. The cement is being poured for the building right now.
03:59.38h3xOh that thing
03:59.39opus_h3x -- file[] just turned me on to this nice piece of hardware
03:59.55opus_<PROTECTED>
04:00.02opus_i'm just going to /s/vonage/opus_/g
04:00.04opus_:)
04:00.10opus_put on a new sticker, tada
04:00.11PBXtech[opus_]: read this http://lestblood.imagodirt.net/archives/83-Asterisk-on-OpenWRT.html#extended
04:00.20h3xhah
04:00.25h3xand get sued for millions
04:00.46opus_oh yeah, wait its cisco
04:00.46h3xPBXtech: one person i talked to running asterisk on openwrt said it crashed a lot
04:01.00Carp1Why is Postgres better than MySQL?
04:01.07PBXtechinteresting i thought it was stable
04:01.17h3xuhoh, here comes the religious RDBMS floodgate from hell
04:01.17opus_PBXtech  -- I am talking about the linksys with the FXS port : http://www.voip-info.org/wiki-Linksys?page=Linksys&comments_threshold=0&comments_offset=0&comments_sort_mode=commentDate_desc&comments_maxComments=10&comments_parentId=861#threadId937
04:01.44terrapenwhy the hell would you want to run Asterisk under OpenWRT
04:01.49terrapenOpenWRT runs on junk hardware
04:01.52opus_h3x - wait, you can reverse engineer hardware to be compatible with it.. should be legal
04:01.58h3xterrapen: I would but the problem is theres not enough flash for voicemail etc
04:02.12terrapenthere are many more problems than that
04:02.19h3xits a 300mhz processor
04:02.20terrapenit's woefully underpowered
04:02.26h3xits actually overpowered
04:02.39h3xit just dosent have enough ram and flash
04:02.39terrapennot for anything useful
04:02.53opus_h3x - for a VPN gateway it is is underpowered.. 4mb/s
04:02.54h3xwell its not like it has a pci slot to stick a t1 card in or anything
04:03.14opus_h3x - the older ones have minipci!!!
04:03.16h3xhah, whos got more than that on their cable or dsl anyway
04:03.20h3xthey do?
04:03.25opus_yup
04:03.42opus_the real old ones have pcmcia
04:03.46h3xmaybe atacomm should make their dsp iVolution card in a minipci format for that thing
04:03.47h3xheheehhe
04:04.10opus_i think you could build a minipci to pci adapter with something from opencores.org
04:04.26h3xum i think its just a different socket, same bus
04:04.40malverianPBXtech: Hmm.. it appears that's not the correct channel, what's the best way to figure out which channel it is?
04:04.43opus_however, its out of my r&d budget:) and time. would be cool thou
04:04.47PBXtechi just want to see linux put on my PSP
04:04.52finejavaas i understand...'sip show peers' will list all the user regardless whether they r logon or not
04:04.57PBXtechin the CLI type zap show channels
04:04.58h3xactually
04:05.01malverianPBXtech: Also, I can't find anywhere in the configuration that defines how many rings to wait before picking up.
04:05.11PBXtechyou should have already configured the channels
04:05.28finejavabut all i need to to list the user with the status 'OK'
04:05.35finejavaanyway we can do that
04:05.38PTG1234i couldn't deface my psp with linux
04:05.40malverianPBXtech: I only have "pseudo" and "1"
04:06.10opus_once you load linux on it I'll deface it for you
04:06.18PBXtech[finejava]: did sip show peers not work?
04:06.46finejavasip show peers works...but it shows every single peer regardless it's offline or online...
04:07.14PBXtechsip show registry
04:07.34finejavabut i just need to view the user who is logon...with the fiels status 'OK'
04:07.36terrapeni would like a box with a >= 667MHz P3-comparable CPU, 512M RAM, totally fanless
04:07.43terrapenthat takes a 12VDC power input
04:07.49Qwellterrapen: picky
04:07.53terrapenand comes in a very small box
04:08.06opus_terrapen - laptop on ebay with broken screen... $30-$50
04:08.08PBXtechmalverian is that 1 channel what your trying to send the call to?  guess you dont have a FXO and a FXS port eh
04:08.22harryvvopus what laptop
04:08.26Qwellopus_: Thats not a bad idea actually
04:08.29terrapenperhaps
04:08.33malverianIt's a digium wildcard w100p
04:08.34terrapenno, not a bad idea
04:08.44terrapenexcept that most laptops were not made to run 24/7/365
04:08.48terrapenheat can be an issue
04:08.50terrapenand they have fans
04:08.51opus_i also offer professional servies:)
04:08.58finejavasip show registry is to show the peers which uses the sytanx register =>
04:09.30PBXtechif you dont have them register then sip show peers is all that i can think of
04:09.31harryvvI need a laptop and if its a dell I can repair it.
04:09.42finejavai just wan a list of user which r successgully logon...and their status fiels is "OK"
04:09.45malverianPBXtech: I used the instructions from digium website on setting up the zaptel configuration, and it only told me to use one channel (1) with fxsks
04:10.06finejavais there any other like using manager API or something
04:10.13PBXtechmalverian if you only have 1 card for an incoming line where are you trying to send the call?
04:10.16PBXtechip phone?
04:10.45PBXtech[finejava]: write your own app and use the manager API sure
04:11.34PBXtechif you just looking for something like an operator console deal, that stuff is out and about
04:12.24malverianPBXtech: It's a single card, but there are two ports (one from wall, one to normal analog telephone). I was just trying to figure out if it was possible to pass the call on to the phone after asterisk has already answered it.
04:12.46PBXtechmight have 2 physical port but its not for that purpose
04:12.49finejavabut i can't seems to find anything in the manager API which can give me the counter
04:12.55malverianIf I don't have asterisk answer the line, it rings on the normal phone by default.
04:13.10malverianPBXtech: Ah, so this isn't a possibility then? That's fine, I just figured I'd ask in case.
04:13.41PBXtechyou can have it answer after 2 rings of what not, but if you want it send to a phone you have to have a different card or IP phone to do that
04:14.32PBXtechfinejava manager API is just a CTI style feed you would have to build your own counter.
04:15.26PBXtechmalverian killer answering machine?
04:15.27PBXtechheh
04:15.39WilliamKblaaaaaaaaaaaah
04:15.40malverianPBXtech: Gotcha, and how do I set the number of rings before pickup?
04:15.54PBXtechdo a Wait(3) then Answer
04:16.01malverianAh.
04:16.15malverianI notice it won't pick up at any less than 2 rings.
04:16.31PBXtechwaiting for CallerID info eh
04:16.51robl^caller ID comes in between first and second ring
04:17.05malverianPBXtech: Will it still do that if I disable caller id?
04:17.23shido6back
04:17.27shido6whats up
04:17.31malverianPBXtech: I'm basically just messing around on my home analog line because I'm going to be creating a new phone server at work.. it's sort of my playground.
04:17.35PBXtechshould be able to answer if upon ring detection
04:17.49PBXtechgood place to start :)
04:18.18PBXtechhi shido6
04:18.35PBXtechyou still at nufone?
04:19.56PBXtechwell im out tata
04:20.02finejavathx PBXtexh
04:20.09finejavaappreciate ur help
04:20.37*** join/#asterisk Rick_Hunter (~rhunter@03-111.008.popsite.net)
04:21.49*** join/#asterisk ericw (~eric@pcp04966776pcs.benslm01.pa.comcast.net)
04:21.51ericwhello
04:22.59ericwtrouble with connecting to sipphone - I initially received response 480, "you are not who you say you are".. although I'm now getting circuit-busy errors instead
04:24.09mishehuok.  lovely.  I don't think we have nearly enough information to even attempt to assist you.
04:24.41greg_workCan anyone see why this:  exten => s,1,Noop(${MACRO_CONTEXT})    exten => s,2,GotoIf($[${MACRO_CONTEXT}=macro-rg-group]?5:3)          does this:        -- Executing NoOp("SIP/219-8925", "macro-rg-group") in new stack    -- Executing GotoIf("SIP/219-8925", "macro-rg-group=macro-rg-group?5:3") in new stack    -- Goto (macro-dial,s,3)      ?
04:24.50ericwmishehu, well, what info is needed?  I have my settings, but I'd prefer to avoid flooding the channel :)
04:24.51Carp1I havnt updated asterisk in like 6 months
04:24.58Carp1and i'm on 56k
04:25.07Carp1and its been updating for over an hour :-\
04:26.27ericwsip.conf:  type=peer, secret= is set appropriately, fromdomain=proxy01.sipphone.com, callerid= my number, host=proxy01.sipphone.com, quality=no, nat=yes, dtmfmode=inband (also tried rfc2833).  reinvite=no, canreinvite=no, context=default, disallow=all, allow=ilbc, allow=gsm, allow=ulaw, allow=alaw, insecure=very
04:26.52*** join/#asterisk jdg (~jdg@CA03F89B.adsl.mana.pf)
04:27.06ericwexternip is set to my internet IP.  localnet is set to 10.0.0.0/255.0.0.0
04:28.09ericwthe machine's IP is 10.0.0.16 and is behind the firewall 10.0.0.1 and is configured on the DMZ port.  The WAN IP of that (NAT) firewall is 10.1.0.2 and has an upstream router of 10.1.0.1 and is connected on the DMZ port of that router.
04:28.19ericwthe WAN port of that router is my internet IP.
04:28.42*** join/#asterisk carlosh (~carlosh@203-96-159-89.paradise.net.nz)
04:31.02mishehujbot: pastebin
04:31.03jbot[pastebin] a place to paste your stuff without flooding the channel - try http://pastebin.ca
04:31.27mishehuericw: that's a good place to dump the stuff you'd like to dump.  I use it fairly often.
04:31.57mishehuquality=no ???  is that a direct copy & paste?
04:32.12*** join/#asterisk ericw (~eric@pcp04966776pcs.benslm01.pa.comcast.net)
04:32.19ericwgot disconnected
04:32.41mishehuericw: generally speaking, http://pastebin.ca is a good place to put stuff like you wanted to
04:32.46mishehuquality=no ???  is that a direct copy & paste?
04:32.59*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
04:33.22ericwmishehu, I was copying and pasting from examples I found online regarding sipphone
04:33.35ericweveryone I saw had it
04:34.30*** join/#asterisk techie (gus@asterisk.horizonte.us)
04:35.04malverianAre there any gnome based sip phones?
04:35.13malverian(gtk based, rather)
04:35.19robl^gnophone?
04:35.27malverianI'd install kphone, but I dread installing qt and the kde libs.
04:35.28ericwmalcolmd, gaimphone, if you can get it working.
04:35.37*** join/#asterisk brc_ (~brian@brc.base.supporter.pdpc)
04:35.37malverianThat works :)
04:35.56mishehuericw: you sure they didn't mean "qualify=yes" ?
04:36.00mishehuerr
04:36.06mishehu"qualify=no" I mean
04:36.31malverianOoh.. linphone
04:36.36carloshhello every one. Could someone tell how many calls would I be able to connect to a meetme dedicated server?  many thanks..
04:36.56ericwmishehu, know what.....
04:37.00mishehucarlosh: however many you had bandwidth and cpu to handle.
04:37.03ericw*I* have quality=no
04:37.09ericw*They* have qualify=no
04:37.15shmaltzhow do I patch app dial with this patch:
04:37.17shmaltzhttp://bugs.digium.com/bug_view_page.php?bug_id=0002905
04:37.18shmaltzHow do I apply the .diff file?
04:37.23mishehuericw: well, I guess you didn't want quality ;-)
04:37.41ericwmishehu, is that even a valid directive? :P
04:37.42carloshwishehu : say, 120 simultaneous calls on a beefy server ?
04:37.51mishehushmaltz: try patch -p0 --dry-run < diff.file
04:38.01mishehuthat will do everything but actually patch it
04:38.10*** join/#asterisk alphaque (~Alphaque@218.111.60.60)
04:38.14ericwmishehu, changing that didn't fix it, though
04:38.15mishehuyou might need to use 1 or higher instead of 0 to patch.
04:38.50mishehuericw: you probably have a different problem.  if you could paste the relevant code to pastebin.ca, it'd make it a bit easier to read.
04:40.45shmaltzmishehu, gives me a msg that 7 out of 7 hunks FAILED
04:41.41shmaltzit was telling me can't find file blah blah blah, so I gave it the path
04:42.00shmaltzbut it still failed
04:42.15QwellI hate seeing the word "hunks" when I patch something...they really should make it "chunks"
04:42.29ericwhttp://pastebin.ca/8758
04:42.30shmaltzQwell any idea?
04:42.36Qwellnope
04:42.37Corydon76-homeQwell: blowing chunks?
04:42.37malverianIs there some kind of asterisk command reference?
04:42.43QwellCorydon76-home: far better
04:42.53Qwell"7 out of 7 of your files blew chunks
04:42.54Qwell"
04:42.55shmaltzmalverian, try the wiki
04:43.01Corydon76-homeQwell: I like blowing hunks better.
04:43.11robl^OH MY!
04:43.12shmaltzQwell, so what should I do?
04:43.13QwellCorydon76-home: tmi
04:43.31shmaltzis it something wrong with the diif file? or the way I'm applying it?
04:43.38shmaltz~docs
04:43.39jbotDocumentation 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, or http://www.asteriskdocs.org
04:43.40mishehushmaltz: man patch
04:43.42Corydon76-homerobl^: I'm good at it, too
04:43.44Qwellshmaltz: The way you're applying it probably
04:44.02shmaltzmishehu, didn't help for me I tried
04:44.05shmaltzwill try again
04:44.25mishehushmaltz: does it prompt for files or does it run thru seemingly automatically?
04:44.26iceypyone here use voipjet?
04:44.35iceypanyone*
04:44.58terrapenis it me or is Sourceforge's download server always...down?
04:45.06shmaltzmishehu, iit prompts me for files
04:45.08JerJer[interop]terrapen: there are a ton of them
04:45.09ericwsip uses udp, right?
04:45.18terrapenno, i mean, the selector server
04:45.24terrapenprdownloads.sourceforge.net
04:45.25shmaltzam I suppose to be in /usr/src/asterisk or in /usr/src/asterisk/apps?
04:45.30terrapenhttp://prdownloads.sourceforge.net/vnc-tight/tightvnc-1.2.9_unixsrc.tar.gz?download
04:45.32shmaltzand where should teh .diff file be?
04:45.35terrapenthat's not loading for me
04:45.53shmaltzand how do I run the patch command?
04:46.01Sedoroxhttp://cogent.dl.sourceforge.net/sourceforge/vnc-tight/tightvnc-1.2.9_unixsrc.tar.gz
04:46.13terrapenthx
04:46.16Sedoroxyup
04:46.25ericwnevermind.. I'm forwarding both
04:46.26terrapenis prdownloads broken?
04:46.30terrapenor is it just me?
04:46.36Qwellterrapen: just you I'd say
04:46.38bkw_ok who wishes to fund the res_sms.c ?
04:46.39ericwso I should be good there
04:46.50Qwellbkw_: I've got $2.50
04:46.54bkw_hehe
04:47.01SedoroxI've got $0.04
04:47.04Qwellsorry, misplaced decimal
04:47.07bkw_well i'm actually gonna get my hands on a GSM/GPRS modem soon
04:47.08Qwell$.250
04:47.11Sedoroxterrapen: worked for me
04:47.16bkw_and whip out res_sms
04:47.28ericwmishehu, any idea?
04:47.29terrapenhow WIERD
04:47.30Sedoroxhmmm
04:47.35terrapenit works on my win32 machine
04:47.39bkw_and write some ast_sms* api's to queue sms's in and out of the machine
04:47.45terrapenbut Firefox on my MacOSX box fails to load it
04:48.03terrapenjust hangs on "Waiting for prdownloads.sourceforge.net..."
04:48.04Qwellterrapen: Is it on a roundrobin DNS or something maybe?
04:48.11mishehubkw_: with iax2, when the user context has trunk=yes defined, any idea why delayreject would cause it to return INVAL ?
04:48.15terrapendunno lemme check that
04:48.31Qwellterrapen: Just see if it resolves to something different on each machine
04:48.43terrapennope
04:48.48mishehuericw: It's not exactly how I'd do it (I'd not use the macro) but I don't see anything that sticks out at me as a problem immediately.  what ver are you using?
04:48.53terrapenonly 1 A record
04:49.03Qwellterrapen: weird
04:49.08ericwmishehu, cvs from a week, maybe 2 weeks ago.
04:49.21Carp1Why is Postgres better than MySQL?
04:49.26mishehuericw: cvs stable or cvs head?
04:49.31QwellCarp1: Who says it is?
04:49.32ericwI had the problem with the cvs from 2 months ago too.. been struggling with this for a while
04:49.37mishehuCarp1: because it isn't.
04:49.50iceypne here use voipjet and find problems with them? my calls die after 22 seconds
04:50.01ericw*default release=cvs
04:50.08ericw^-- asterisk.sup
04:50.10Carp1It seems to me as if everyone writing apps for Asterisk are either switching from MySQL to it or starting with it completely
04:50.21SedoroxWhat would I search for if I wanted to run my own "cell site" like.. if I had a few gsm cell phones, with sims.. and I could register the sims with it..?
04:50.25mishehuiceyp: I've not made many calls this past week, but I have no problems.
04:50.35ericwI suppose that is head?
04:50.44terrapenUGH
04:50.47terrapentightvnc uses imake
04:50.51mishehuericw: show version  will tell you
04:51.06iceypmishehu i;ve made 5 test calls and using all servers, westcoast is better for me
04:51.08ericwAsterisk  built by root@warhol on a i686 running Linux
04:51.09ericw....
04:51.19iceypbut after 22 seconds of the call its dead
04:51.27iceypall their email addresses bounce too
04:51.43mishehuiceyp: fastsupport@voipjet.com bounces?
04:51.51iceypyep
04:51.54iceypsame with carriers
04:52.15*** join/#asterisk Inv_arp (junya@adsl-3-237-168.mia.bellsouth.net)
04:52.29ericwmishehu, btw.. I added a paste of the error I'm getting now in there.  I was getting another before.
04:52.31iceypanyone here using ser @ asterisk?
04:52.41iceypi cant get ser to accept calls from asterisk
04:53.47Sedoroxhmmm
04:53.49Sedoroxguess no
04:53.50Sedoroxnot*
04:54.01mishehuiceyp: I'd have to say the problem is either with westcoast, as I use eastcoast, or it's with you.  I just tested, I called more than 22 seconds.
04:54.25iceyphmmm, i tried primary east coast server and westcoast server
04:54.33iceypmy calls are fine with nufone ;/
04:54.40iceypmaybe its calls to new zealand
04:54.48mishehuiceyp: where are you calling to?
04:54.51iceypNZ
04:54.56iceypmy home phone
04:55.17mishehuiceyp: and do you have the 22 second problem if you call another country?
04:55.24iceyphavent tried
04:55.32iceypu got a number i can dial
04:55.33*** join/#asterisk quickmoney (~jfu2808@CPE00a0c5e1b8b3-CM013010000950.cpe.net.cable.rogers.com)
04:55.40mishehuericw: sec
04:55.59iceypu dont have to talk just leave it off the hook when i dial u
04:56.11Qwelliceyp: Want me to msg you a number?
04:56.40greg_workfeatures.conf mods require a complete restart?
04:57.21mishehuericw: I don't see the update.   you sure it posted to the same pastebin?
04:57.29ericwhttp://pastebin.ca/8759
04:57.37Qwellmishehu: it updates the number I believe
04:57.40ericw"--- error being given (currently) ---"
04:58.05iceypi seem to have dialed u fine
04:58.09iceypargh
04:58.37shmaltzhow do I patch app dial with this patch:
04:58.39shmaltzhttp://bugs.digium.com/bug_view_page.php?bug_id=0002905
04:58.40shmaltzHow do I apply the .diff file?
04:58.52shmaltzshould I apply just the last revison?
04:58.59shmaltzor all previous ones as well?
05:01.02wildcard0ok...am i just missing it or is there go 'and.gsm' sound file from allison smith?
05:01.23Qwell/usr/cvsroot/asterisk-sounds/sounds/and.gsm
05:01.36wildcard0hmm.  why don't i have that?
05:01.39QwellThere's also vm-and.gsm
05:03.11ericwmishehu, with a different number (a long distance number, which I would need to pay for), I'm getting - Got SIP response 400 "Bad Request" back from 198.65.166.131
05:03.33mishehuericw: have you tried tcpdumping?
05:03.37mishehumight be useful.
05:06.32*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || Who wants SMS in asterisk? TRUE SMS? /msg bkw_
05:09.30mishehualright, time for sleep.
05:09.42ericwmishehu, ok. night.  thank you
05:10.02ericwbtw, I got that tcpdump
05:10.12ericwhttp://eric.bwbohh.net/sipphone.log
05:10.33mishehuericw: sorry, wife is getting mad
05:10.34mishehuheh
05:10.42ericwmishehu, heh.. mine too!
05:10.55ericwnight
05:13.46*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || res_sms anyone?
05:13.47*** join/#asterisk veesmooth (~veesmooth@bgp505102bgs.verona01.nj.comcast.net)
05:14.05veesmoothhey guys
05:14.12*** part/#asterisk esandeen (~sandeen@sandeen.net)
05:14.17veesmoothanyone here
05:15.17`SauronNobody here, move along...
05:15.37veesmoothhe he, thanks for tryin to chase me away
05:15.47veesmoothgot a question if anyone willing to answer
05:16.23shido6shoot
05:17.05marloweDon't ask to ask.. Just ask.
05:17.38Sedoroxanyone know what keyword/companies I should look for if I want to register sim cards and have my own "Cell site" ?
05:17.50shido6ATEUS
05:17.51marloweSedorox: Why do you want to?
05:17.57marloweOut of curiosity.. :)
05:18.11*** join/#asterisk rowter (~Drake@201.133.210.80)
05:19.32Sedoroxwell I have some old cell phones.. and was kinda hoping to do my own little setup with * to have my own personal cell service going, basicly just to mess around with
05:20.34marlowethats cool
05:20.40marlowethats beyond me
05:20.51Sedoroxhehe
05:20.55QwellThats an expensive "messing around" :p
05:21.20*** join/#asterisk sunil (~sunil@202.54.37.181)
05:21.21marloweIll buy service... Whats the reception area? Around your house?
05:21.53Sedoroxlol
05:21.54Sedoroxprobably
05:22.01Sedoroxsince It would run into fcc problems
05:22.13Sedoroxand Qwell.. yes.. most of my 'messing arounds' seem to be
05:22.35Sedoroxshido6: thanks... have you used it before?
05:23.13shido6yes
05:23.17shido6in aussie land
05:23.20shido6and an E1
05:23.27shido6buts its gsm
05:23.32Sedoroxyea
05:23.34Sedoroxcool
05:23.39Sedoroxwork well?
05:24.02shido6absolutely
05:24.10Sedoroxneat
05:24.13shido6what are you looking to do?
05:24.43*** join/#asterisk yxa (~void@203.118.40.42)
05:24.54nDuffWhat's the Right Way to dump a bunch of data into the Asterisk DB? (I've got a script w/ "database put" statements, but piping it to "asterisk -r" doesn't seem to Do The Right Thing).
05:26.05Sedoroxjust use some old cell phone and make it where I can dial into asterisk and from asterisk... kinda a longer-range cordless phone, just with stuff I have (minus the gateway thingy)
05:26.42nDuffI suppose "asterisk -rx <command>" would work, but that would mean a separate invocation for every put, right?
05:26.52*** join/#asterisk Fddayan (~fddayan@c-67-191-7-6.hsd1.fl.comcast.net)
05:27.08greg_worknDuff: use the asterisk manager api
05:27.20Fddayansombody knows how to emulate a call with a WAV file in asterisk ?
05:28.12wildcard0Fddayan, you want to make a call and then play a wav file to it?
05:28.52marloweKeep in mind the asterisk datbase is just version 1 of the berkley db
05:29.05Fddayanyes !
05:29.13Silik0n*yawn*
05:29.18Fddayanthere is any way ?
05:29.20marloweTats my hint
05:29.22marloweFddayan: Of course
05:29.22wildcard0Fddayan, create a call file and have it use an extension that has a Playback command in it
05:29.44marloweumm
05:30.29Sedoroxhmmm
05:30.44marloweLook up the cmd Dial
05:30.51marloweIt's right there
05:31.10marloweA(x): Play an announcement (x.gsm) to the called party.
05:31.15marloweCan't be much easier.
05:31.31SedoroxOk... heading in
05:31.35SedoroxNight
05:31.41sunilhello, can any one help me in configuring asterisk on mfcr2 signalling
05:31.56*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
05:36.46*** join/#asterisk wiseguy_ (~chivilis@vadyba.vtu.lt)
05:36.48wiseguy_helow
05:37.08wiseguy_anybody with cisco routers?
05:37.09wiseguy_:)
05:37.16wildcard0im with 2 of them
05:37.19wildcard0we're close
05:37.30*** part/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net)
05:37.41wiseguy_wildcard0
05:37.49wiseguy_can show me part of sip.conf
05:37.57wiseguy_where cisco peer is described
05:38.14wiseguy_?
05:38.24wildcard0oh i dun have asterisk talking to them directly.  they're IP routers
05:38.45wiseguy_:/
05:39.43DEEZEDhow do you get asterisk to say a number? does it need a gsm file, or does it have it built it to say numbers?
05:39.55wildcard0SayNumber
05:40.11wildcard0it uses the numbers in /var/lib/asterisk/sounds/digits
05:40.31robl^DEEZED: yes.  SayNumber will find the sound files it needs and say the the number for you
05:40.59tessierCan anyone tell me what Vovida does that asterisk doesn't?
05:41.22h3xtessier: 0 results returned
05:41.24h3xactually
05:41.28DEEZEDthanks... so SayNumber(1)
05:41.32DEEZEDwould say 1>
05:41.33DEEZED?
05:41.40robl^yes
05:41.44DEEZEDthanks
05:41.46h3xVovida what application?
05:41.49robl^and 201 would say "two hundred one"
05:41.59DEEZEDsweetness
05:42.05QwellSayNumber(304) says "Hello Bob, how is your day?"
05:42.07Qwelletc
05:42.13robl^SayDigits(201) would say "two zero one"
05:43.51h3xtessier: what are you talking about, VOCAL?
05:43.55wiseguy_anyone with cisco 1760 + asterisk?
05:44.32veesmoothanyone good with programming in astersisk?
05:44.42nDuffveesmooth, I'm sure someone is.
05:44.43tessierh3x: Yes, vocal.
05:44.47veesmoothmight need some assitant
05:44.49veesmoothlol
05:44.51*** join/#asterisk TekGecko (~univ@dhcp-96878f9f.rescomp.arizona.edu)
05:44.57h3xwell i bet it does a better job of h.323 -> sip than asterisk :P
05:44.59veesmoothwith what to do with asterisk as a masters project
05:45.17veesmoothtrying to get some ideas
05:45.41wildcard0what area are you studying?
05:45.46veesmoothcomputer science
05:45.49wiseguy_when i write sip show peers, status is unmonitored
05:45.56wiseguy_how to change into monitored?
05:45.56wiseguy_:)
05:45.56tessierLots of people say they are using ser, asterisk, and vocal.
05:46.10wildcard0veesmooth, yes i gathered that...i ment what concentration
05:46.12tessierI don't see what vocal does that asterisk doesn't.
05:46.22nDuffveesmooth, maybe you could try for a better echo cancellation algorithm? That's a good computer-sciency subject.
05:46.35wildcard0wiseguy_, add qualify=yes
05:46.47wildcard0nDuff, that'd be REALLY useful
05:46.56veesmoothwell im not really concentrationg on anything
05:47.03veesmoothmore general with software and networking
05:47.07h3xvocal and ser have a provisioning gui
05:47.25wildcard0maybe work on the more general jitterbuffer?
05:47.56veesmoothecho cancellation algorithm huh
05:47.57veesmoothhmmmm
05:48.04veesmootha better one
05:48.19veesmoothis that something you can find more info on the net or anything?
05:48.55wildcard0http://cnx.rice.edu/content/m11909/latest/
05:49.05wildcard0http://www.embeddedstar.com/articles/2003/7/article20030720-1.html
05:49.14wildcard0http://lcavwww.epfl.ch/~prandoni/dsp/echo/echo.html
05:49.22wildcard0and generally
05:49.27veesmoothall for me wild card?
05:49.29wildcard0http://www.google.ca/search?q=echo+cancellation
05:49.31wildcard0yes
05:50.59veesmoothhold on trying to copy these links now
05:51.32veesmoothbut that does sound good for  a project right
05:51.46wildcard0i think it sounds good, but i'd run it by my advisor :)
05:51.54Zipper_32How does one dial into a phone (from a PSTN) which is behind asterisk via an E1 connection? Where is the actual number being routed?
05:52.04wildcard0you might choose 2 or 3 and run them all by your advisor
05:52.32veesmoothso there are more then one then huh
05:52.40wildcard0Zipper_32, so you want to go   E1 --> asterisk --> ata --> phone?
05:53.00wildcard0veesmooth, i meant 2-3 possible topics.  but it's your thesis
05:53.16veesmoothyeah thats what i want to do , im a def put that as one of my topics
05:53.49wildcard0veesmooth, it'd be cool to do some research on extending the max latency for a voip call also
05:53.50Zipper_32wildcard0: [trying to figure out what 'ata' means... wanna give me a hand?
05:54.24Zipper_32wildcard0: I'm just wondering how someone from behind the PSTN can call me if I'm behind an E1 line running asterisk with an IP phone.
05:54.24wildcard0Zipper_32, analog telephone adapter.  like http://www.sipura.com/products/spa2000.htm for example
05:55.21wildcard0they dial the number to the E1 you have plugged into asterisk.    in extensions.conf you have something like 'exten => 9995551212,1,Dial(SIP/1)'
05:55.44wildcard0where 9995551212 is your DID and SIP/1 is the path to your IP phone
05:56.20Zipper_32So the E1 has a number... gotcha. [I didn't know that part, I assumed it was just a broadband channel]
05:56.35*** join/#asterisk Kumbang (~ecvs@167.205.24.4)
05:56.48wildcard0oh it CAN be a broadband channel.  or it can be a channelized voice connection
05:56.52wildcard0it depends how you have it set up
05:56.57*** join/#asterisk SuperMMan (~graphic@edtntnt2-port-216.dial.telus.net)
05:57.32Zipper_32Alright, say I have 4 lines, I'm assuming that I want a channelized voice connection to a VOIP provider, right?
05:57.46wildcard0yes
05:57.48Zipper_32By the way, forgive my ignorance, it's my first day.
05:57.57wildcard0lemme find a url
05:58.07Zipper_32OoOo, goodie, Thanks.
05:58.13Zipper_32I've been reading ever since morning.
05:58.45veesmoothso to do these research u think it will take a lot to figure these things out
05:58.51veesmoothespecially since im new to astrisk
05:59.32wildcard0Zipper_32, this is a good starting place.   a t1 is roughly the same as an E1.  just an E1 has more bandwidth/channels
05:59.33wildcard0http://en.wikipedia.org/wiki/DS1
06:01.11Zipper_32Alright, now I read earlier that you can run 4 simultaneous lines off of a regular DSL connection, so is this really all I would need?
06:01.13wildcard0Zipper_32, if you're really starting from scratch to be a voice providor, consider hiring a consultant
06:01.47Zipper_32I'm starting from scratch to find out what to setup for a new office location. And if possible, set it up myself.
06:01.53wildcard0ah
06:02.01wildcard0oh you -need- a voice provider
06:02.04wildcard0where are you located?
06:02.11Zipper_32Coquitlam BC
06:02.18Zipper_3220min east of Vancouver
06:02.21wildcard0oh.  hmm.  that would be me actually
06:02.22wildcard0hehe
06:02.25Zipper_32lol!
06:02.27wildcard0<-- downtown van
06:02.36Zipper_32I'm in White Rock right now
06:02.44wildcard0"surrey" :)
06:02.49Zipper_32Shh!
06:02.53wildcard0hehe
06:02.54Zipper_32Don't say that too loud...
06:03.10Zipper_32What kind of work do you do?
06:03.32wildcard0termination services for small businesses, hotels and commercial highrises
06:03.39wildcard0http://www.mxunetworks.com
06:03.41Zipper_32That's me.
06:03.50Zipper_32me = small business
06:10.31*** join/#asterisk jwitte (~jwitte_su@port-212-202-101-206.static.qsc.de)
06:12.52veesmooththis asterisk stuff is pretty interesting though
06:13.01veesmoothjust installed it and getting use to it on my linux computer
06:15.45*** join/#asterisk l00p (~l00p@c-67-171-201-105.hsd1.or.comcast.net)
06:16.26l00pAnyone awake?
06:16.49veesmoothi am
06:16.59shepherdNO!
06:17.01shepherd:D
06:17.12shepherddangit
06:17.17shepherdDay light savings time
06:17.18shepherdni ni
06:17.23veesmoothyup
06:17.23l00pI've been doing a lot of reading on asterisk and have a question.
06:17.26veesmoothmake it closet to work
06:17.32l00pCan I use DIDs instead of extensions?
06:18.11veesmoothim a asterisk newbie so im tring to figure out stuff myself
06:18.18shepherdDID's like voip lines from a provider?
06:18.26*** join/#asterisk JerJer[mobile] (~nonyobizn@RtrHSTF-FC.hstf.interop.net)
06:19.19l00pWe use a Meridian phone system that has direct numbers (555-5555 not 555-5555 x1234)
06:19.30Qwelll00p: sure
06:19.40l00pWe are talking about going to asterisk, but don't want to change everyone's number
06:19.51l00pAnywhere I can read about this?
06:19.52shepherdyeah.. you can do that :)
06:20.11shepherdwell..
06:20.12shepherdbasically
06:20.22shepherdyou have to associate a DID with an extension
06:20.27carloshhello every one. Could someone tell how many calls would I be able to connect to a meetme dedicated server?  many thanks..
06:20.36shepherdbut it works the same
06:20.47l00pWhere do you do the association?
06:20.53Qwelll00p: extensions.conf
06:21.07shepherd<PROTECTED>
06:21.10shepherdyeah.. there
06:21.16l00pK, that's what I thought.
06:21.20l00pThanks.
06:21.36l00pYou guys here often?
06:21.40shepherdl00p: read the handbook, it's the old way of doing it, but it works nonetheless
06:21.48Qwelll00p: 24 hours a day
06:22.06l00pI read the handbook(mostly)
06:22.18l00pIt is good.
06:22.18shepherdk
06:22.31l00pKnow anyone in Oregon who has a system up and running?
06:22.32*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
06:23.37l00pI made my first call about two hours ago using kphone. It was pretty cool.
06:25.44munchausensomeone should set up a simple service which calls any two phone numbers with any specified callid
06:26.12munchausenwhy? because
06:26.34luke-jr_l00p: exten => _555555XXXX,1,Dial(SIP/${EXTEN:6}) if you're lucky ;)
06:27.05shepherdneat.. a phone that supports iax
06:27.05luke-jr_munchausen: you mean dial 2 #s and connect em?
06:27.06shepherdhttp://www.virbiage.com/products/lanphones.php
06:27.30luke-jr_shepherd: too bad everyone's moving to the new XIAX, eh? =p
06:27.37l00pluke-jr: thanks
06:27.37shepherdyeah..
06:27.42shepherdxiax is the shit
06:27.44shepherd:)
06:28.19shepherdlord.. if xiax was a reality, i would shoot myself
06:28.21shepherdheh
06:28.33luke-jr_why?
06:28.38shepherdokay.. not really
06:28.42luke-jr_lol
06:28.42shepherdbut it's a bad idea
06:28.45shepherdhehe
06:29.01luke-jr_which part?
06:29.16shepherdoh just the overhead
06:29.17luke-jr_native SSL support would be nice
06:29.32shepherdthe cpu required to parse it
06:29.39luke-jr_better than when I have to tunnel VoIP over PPP over SSH at least
06:29.51shepherdyeah
06:29.54*** join/#asterisk veesmooth (~veesmooth@bgp505102bgs.verona01.nj.comcast.net)
06:29.56shepherdiax3 needs ssl :)
06:29.57luke-jr_(when I don't trust a network)
06:30.15shepherdyou could vpn
06:30.15luke-jr_does IAX use TCP?
06:30.18*** join/#asterisk PoWeRKiLL (~PoWeRKiLL@l01m-20-133.d4.club-internet.fr)
06:30.20luke-jr_that's what it is
06:30.20shepherdno.. udp
06:30.25luke-jr_PPP over SSH VPN
06:30.29shepherdoh, lol
06:30.46shepherdso you see the server as a local ip?
06:30.51luke-jr_yes
06:31.09shepherdi wonder how much compression it would do on ulaw
06:31.23luke-jr_SSL compresses? O.o
06:31.39luke-jr_I thought it was merely encryption
06:31.49shepherdyeah.. yeah.. openssl does compression too, i think
06:31.58shepherdjust depends on the setting
06:32.06luke-jr_SSH does... dunno about SSL
06:32.52shepherdssl is the protocol layer of ssh :)
06:32.52l00pI don't think ssl normalyl does compression, but I could be wrong.
06:33.00shepherdi'm pretty sure
06:33.12luke-jr_for the encryption, yes
06:33.26luke-jr_I would expect SSH to use something else like zlib for compression tho
06:33.27shepherdlike.. sometimes
06:33.31shepherdwhen i use sftp
06:33.39shepherdit downloads websites superfast
06:33.44shepherdbecause it's several files
06:33.50luke-jr_sftp is SSH, not SSL
06:33.53shepherdthat have a lot of waist
06:33.57shepherdand are uncompressed
06:34.21luke-jr_SSL may be used in SSH, but SSL still != SSH ;)
06:34.24shepherdso i just always assumed it was compressing all the files together and sending them at once
06:34.37luke-jr_I doubt that
06:34.52luke-jr_if it did that, it'd need to be at sftp layer
06:34.54luke-jr_not even SSH
06:35.00shepherdnonetheless, i can pull a website down faster with sftp than ftp :)
06:35.18luke-jr_i'm lazy when I do stuff like that
06:35.26l00pThanks for the help. I'm out.
06:35.33luke-jr_I usually do something like tar cjvp files | nc
06:35.43shepherdyeah..
06:35.47shepherdbut when you don't have console access
06:35.54shepherdgot to get it somehow, heh
06:35.58luke-jr_...
06:36.04luke-jr_sftp requires shell access
06:36.08shepherdi have a provider that supports sftp
06:36.17shepherdbut no login
06:36.21luke-jr_interesting
06:36.25shepherdit's a windows server
06:36.27shepherdbah :(
06:36.29luke-jr_evil
06:39.10*** join/#asterisk _vic (~riccardo@gw-fi.esaote.com)
06:39.33veesmoothso guys
06:39.36veesmoothim a newbie here
06:39.42veesmoothcan i ask some newbie questions lol
06:39.47shepherdno
06:39.49shepherdhehe
06:39.55shepherdsarcasim
06:39.57shepherd:(
06:40.02veesmoothok, i take that as a yes he he he
06:40.05veesmoothhopefully
06:40.12veesmoothi just install asterisk on my system
06:40.14veesmoothand it work great
06:40.20veesmoothwell with the sample install that is
06:40.32veesmoothbut im wondering if there is something else i can do with it
06:40.42luke-jr_sample install I got was useless, IIRC
06:40.56veesmoothi know most likly i cant dial regular phone lines
06:41.02luke-jr_why not?
06:41.08veesmoothjust asumming
06:41.13shepherdyeah you can
06:41.13veesmoothcan you?
06:41.16luke-jr_yea
06:41.17veesmoothhmm
06:41.21veesmooththats wild
06:41.25shepherddo you have any hardware?
06:41.25veesmoothso through high speed
06:41.25nDuffshepherd, no, SSH doesn't use SSL/TLS.
06:41.25luke-jr_I suggest http://www.voipjet.com/
06:41.27drumkillanot on the sample install ...
06:41.34veesmoothno hardware
06:41.45veesmoothdoing this through highspeed
06:41.46shepherdvoicepulse!
06:41.52shepherdnduff: okay.. heh
06:41.54luke-jr_tho VoipJet is meant for big groups
06:42.10veesmoothso i need a special hard ware to call regular phones then
06:42.15luke-jr_veesmooth: no
06:42.15veesmoothcant do it through highspeed right
06:42.32luke-jr_veesmooth: Just need someone providing VoIP termination
06:43.03shepherdnope
06:43.03veesmooth<PROTECTED>
06:43.03luke-jr_lots of companies do it
06:43.10shepherdyou can do a total voip solution :)
06:43.33luke-jr_veesmooth: VoIP services are generally either termination or origination
06:43.41luke-jr_termination is for outgoing POTS calls
06:43.49shepherdi use voicepulse for my phone number coming in, into a TDM411
06:43.51luke-jr_origination is for incoming POTS calls
06:43.58shepherdbut my fxo isn't hookup
06:44.10malverianHmm, I've noticed my Asterisk server doesn't catch keypad presses when I dial from a cellular phone a lot.
06:44.17malverianIs there anything I can do to tweak this?
06:44.27luke-jr_shepherd: heh... my interest in VoIP originally stemmed from the absense of a non-SBC landline solution
06:44.53luke-jr_malverian: use a different codec maybe
06:45.31shepherdmalverian: isn't by any chance a sony cell phone?
06:46.09luke-jr_veesmooth: Termination is usually charged in cents/minute; origination is usually a flat monthly fee
06:46.24veesmoothgot ya
06:46.44luke-jr_For example, I pay $5/mo origination and 1.3c/min termination
06:47.03veesmoothok, so lets say i dont have any of those providers and i just have the software
06:47.08veesmoothis there anything i can do with it
06:47.10veesmoothjust as it is
06:47.20luke-jr_you need to configure it to do something
06:47.35veesmoothhmm, ok
06:47.35luke-jr_but you should be able to call anyone else using VoIP in theory
06:47.47luke-jr_via some enum service or dundi
06:47.55luke-jr_I haven't got those setup myself yet tho
06:48.06veesmoothok, so if someone had a voip then i can dial their number
06:48.12veesmoothi know the sample software let me dial their company
06:48.17luke-jr_yes
06:48.18veesmoothand fool around with them
06:48.24veesmoothok cool
06:48.33veesmoothso what do u do with asterisk
06:48.40luke-jr_me?
06:48.48veesmoothyes, or anyone else
06:48.49veesmoothjust curious
06:48.50luke-jr_It's my phone :)
06:49.02luke-jr_well, controls the phones in my appt
06:49.02veesmoothoh
06:49.12luke-jr_and handles calls
06:49.20*** join/#asterisk tafazzi (~Dario@eporroj0-1.customer.nettuno.it)
06:49.24veesmoothhanles call in a smart way
06:49.25malverianshepherd: It is.. an ericsson
06:49.30*** join/#asterisk Rick_Hunter (~rhunter@03-111.008.popsite.net)
06:49.39luke-jr_not really... I just have it doing speeddial mostly now ;)
06:49.41shepherdKevin Mitnick uses asterisk to spoof callerids :)
06:49.47luke-jr_shepherd: LOL
06:49.50shepherdmalver: intresting
06:50.10shepherdi had the same problem yesterday, but wasn't sure it was the phone
06:50.16shepherdor something else
06:50.41veesmoothreason im asking cause im suppose to do a project with the software for school
06:50.48veesmoothtrying to get some good ideas with this
06:50.52shepherdi set mine to 555 555 4444
06:50.56shepherdcoffee talk :)
06:51.13veesmoothgot two decent subjects so far
06:51.15luke-jr_veesmooth: could setup something for your dorm to use... assign em extensions and stuff
06:51.44shepherdyou could use asterisk to control your firewall :)
06:51.51shepherdmeaning..
06:51.53luke-jr_shepherd: wtf?
06:51.58shepherdfirewall everything out
06:52.03shepherdthen when you want access to your sever
06:52.08shepherdcall up your asterisk box
06:52.16luke-jr_lol
06:52.22shepherdkey in your ip address and update iptables :)
06:52.42luke-jr_shepherd: is there actually apps to do that?
06:52.46veesmooththink the prof wants me to do something software wise
06:52.47shepherdno
06:52.53shepherdbut you could do it!
06:52.55shepherdhehe
06:52.58veesmoothis that something simple?
06:53.03veesmoothsounds kinda tricky
06:53.06luke-jr_shepherd: sounds like it would be fun if I had time
06:53.12shepherdwell.. it's like writting a cgi
06:53.16luke-jr_veesmooth: you'd need to code it
06:53.23shepherdonly in asterisk its an agi
06:53.29veesmoothcan u code that in c?
06:53.33shepherdyes
06:53.37shepherdc, java, perl
06:53.40shepherdpython
06:53.44shepherdphp
06:53.45*** join/#asterisk ckruetze (~nospam@131.8.dsl3.ip.foni.net)
06:53.53shepherdc# even
06:53.58Qwelleww
06:54.11QwellDoes it use mono or something?
06:54.15shepherdyeah
06:54.29shepherdi think the support for c# agi is kinda sketchy though
06:54.31veesmoothis that something complicated to do
06:54.32shepherdi wouldn't trust it
06:54.50shepherdvees: not really.. i learned agi before i learned extensions.conf ;0
06:54.53shepherd;)
06:54.56luke-jr_maybe write a voice bulletin board ;)
06:55.33veesmoothmaybe
06:55.42shepherdthat might be harder to do though
06:55.45veesmoothim writting down all these ideas
06:55.51veesmoothso i can look into it
06:55.57veesmoothok
06:55.58luke-jr_the hard part of a BB would be the UI design
06:55.58shepherdthe firewall idea would be really simple project
06:56.05shepherdas far as agis go
06:56.12veesmoothbut i think what the prof wants me to do
06:56.16veesmoothis take the asterisk code
06:56.17veesmoothitself
06:56.23veesmoothwhich i believe is written in c
06:56.25shepherdyup
06:56.28veesmoothand do something with it
06:56.48veesmoothi have to refresh my brain on c
06:56.59veesmoothlol but i love programming
06:57.03shepherdwell.. asterisk is modular, so you can do a lot without changing the core of asterisk :)
06:57.22veesmoothoh
06:57.30veesmoothso in other words i can just add on with no problem
06:57.36shepherdyeah
06:57.38veesmoothwith out changing the basics though of the program
06:57.44shepherdif it doesn't already do it
06:58.04shepherdmore than likely, it can, or someone has a program out there that will do whatever you want it to do
06:58.15veesmoothoh
06:58.19veesmoothso i can download programs
06:58.21veesmoothand just add it on
06:58.30veesmoothjust like legos
06:58.31veesmoothlol
06:58.34veesmoothadd and take away
06:59.01veesmoothis that what u mean?
07:00.08veesmoothok shepherd and luke
07:00.08shepherdnot as simple ;)
07:00.19veesmoothtell me if this sound like something
07:00.24veesmooththat is do able
07:00.30veesmoothi got two subjects so far though
07:00.36veesmooththat was sugested by some other folks
07:00.50veesmoothone: better echo cancellation algorithm
07:00.53*** part/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net)
07:01.04veesmoothtwo: extending the max latency for a voip
07:01.14veesmoothdo you think that sounds like something?
07:01.36shepherdheh
07:01.46shepherdi would stay away from the echo cancellation
07:01.59shepherdmark said that was the hardest thing he ever had to do in asterisk ever
07:01.59veesmooththat sound kinda complicated
07:02.05veesmoothhmm
07:02.06veesmoothok
07:02.09veesmoothi keep that in mind
07:02.23veesmooth<PROTECTED>
07:02.24shepherdbut the max latency sounds neat
07:02.28veesmoothoh yeah
07:02.42shepherdbut, most of that would just be configuration changes
07:02.47veesmooththink that sounds kinda doable
07:03.06shepherdand you can do the jitter buffer and just talk really slow
07:03.14*** join/#asterisk RestLessGemini (~umairbari@202.142.189.86)
07:03.24veesmoothhmm, have u work with c and asterisk before
07:03.31shepherdno
07:03.36shepherdi can't really code in c
07:03.41veesmoothhmm
07:03.42shepherdmost of what i've use was perl and php
07:04.00veesmoothhmm cause im wondering if it would take a lot to tweak the program
07:04.03veesmoothusing c
07:04.14veesmoothsince i never really went into a program and change stuff
07:04.22veesmoothi usually create things from scratch if anything
07:04.29shepherdyeah
07:04.40*** join/#asterisk Himeko (~himeko@S01060040ca128fc3.ed.shawcable.net)
07:04.41shepherdnow i have tweaked stuff in c
07:04.49veesmooththink this sounds like a tough project?
07:04.51shepherdlike max threads
07:04.54veesmoothor something thats doable
07:05.00*** join/#asterisk tuxinator_linux (~tuxinator@ip68-109-146-168.ph.ph.cox.net)
07:05.02*** join/#asterisk tartar (~tartar@CPE0004e27b716e-CM014370001917.cpe.net.cable.rogers.com)
07:05.39shepherdyou could speed up iax initiations by doing away with linked lists ;)
07:05.54veesmoothhmm, ok
07:06.10veesmoothbut this project does sound do able though right
07:06.24shepherdyeah
07:06.33shepherdbut
07:06.43*** join/#asterisk |nix (~inix@202.148.164.48)
07:06.45shepherdyou might want to look over the code first
07:06.55shepherdand focus on one part
07:07.01veesmoothyeah true
07:07.07veesmooththe comments should tell u what it does right
07:07.18shepherdbecause, i'm not sure how about pieces affect latency, etc
07:07.36shepherdhow many pieces
07:07.40veesmoothyeah, well he said im a be doing a lot of hours
07:07.42veesmoothand research
07:07.48veesmoothas long as i have an idea
07:07.52veesmoothand willing to put in some work
07:07.54veesmoothand time
07:07.56opus_does anyone know how to simulate latency?
07:08.03veesmooththis is mostly in the fall
07:08.05veesmoothjust got to prepare now
07:08.31veesmoothso this site a lot of people come on and help right
07:08.41veesmoothso far i got some decent help here
07:09.02shepherdopus: you could probably throttle back the bandwidth and ping flood yourself :)
07:09.05opus_whats up man
07:09.14opus_shepherd- not a bad idea
07:10.39Qwellwtf...
07:10.57*** join/#asterisk pif (ldm@zenon.apartia.fr)
07:10.58shepherd??
07:11.11Qwellmy brother is telling me about a hotel he and my dad did a telecom install for...  They asked them to give the bathroom in each room a seperate extension
07:11.28Qwellmain room would be 1011, bathroom would be 2011
07:12.05Himekothat would be handy
07:12.11Qwellseems retarded
07:12.22shepherdwhy would anyone want that
07:12.33*** join/#asterisk dg1nsw (~schulte@gate.sympat.de)
07:12.34Himekoyou could call the bathroom from the room then
07:12.41Himekoand visa versa
07:12.52shepherdor! talk through the door
07:12.54shepherdhehe
07:12.56Qwellheh
07:13.16Himekowhy, when you got separate extentions
07:14.55Himekohey slePP, you still up
07:15.22QwellHow much could a hotel with 2,000 rooms save by going from a normal PBX to *?
07:15.43Qwellor, rather...
07:15.48Qwellgoing to * instead of a normal PBX
07:16.00QwellI didn't really think my question out the first time :p
07:16.26slePPHimeko: yeh
07:16.29shepherdthe question should be
07:16.34shepherdcan asterisk do 2,000 lines
07:16.43Qwellshepherd: I'm sure it could, with a few boxes
07:16.55HimekoslePP i don't think the * box knows it is DST
07:17.13slePPthe box does.. asterisk apparently isn't aware, though
07:17.20Himekoweird
07:17.37slePPergasio root # date
07:17.37slePPMon Apr  4 01:17:04 MDT 2005
07:18.09Himekoit must checking the rtc itself and doing the the tz or something
07:18.33slePPno.. asterisk tends to maintain its own clock
07:18.37slePPactually
07:19.15slePPwhat are you figuring the time is wrong from?
07:19.15slePPcaller id?
07:19.15Himekoya
07:19.15slePPthat's the pap2
07:19.25Himekoit has it's own clock eh
07:19.27slePPit lacks a concept of DST
07:19.50*** join/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
07:19.50slePPthe pap2 uses ntp to keep in sync, but it doesn't adjust itself
07:19.50*** join/#asterisk eivindtr (~eivind@062016241059.customer.alfanett.no)
07:19.52slePPon the next profile update it should clear itself up
07:19.59Himekoi just figured it passed the cid though not generate it from itself
07:20.20slePPit generates the local CID info from the SIP
07:20.43slePPbut it doesn't take the time from it
07:21.08Himekoi thought it was set to check the config every 5 min
07:21.29Himekoor not anymore
07:21.29slePPew, no... they have this nasty habit of rebooting when they don't need to.
07:21.35slePPit checks configs every hour or two or 24
07:22.01slePPmind you, you're not on the new provisioning system..
07:24.53*** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se)
07:33.44carloshhello every one. Could someone tell how many calls would I be able to connect to an server dedicated to meetme conferences ?  many thanks..
07:34.08carlosh..an * server...
07:35.25Qwellcarlosh: There are alot of variables
07:35.36Qwellhardware used, codecs, amount of transcoding, etc
07:35.45*** join/#asterisk mbaron (~mbaron@AVelizy-154-1-28-204.w82-124.abo.wanadoo.fr)
07:35.45Qwellcould be 10, could be 1,000
07:35.53*** part/#asterisk mbaron (~mbaron@AVelizy-154-1-28-204.w82-124.abo.wanadoo.fr)
07:36.53carloshQwell: say good bandwidth, 2.4GHZ processor or better, 1GBRAM... hyperthreading CPu, the works...
07:37.06Qwellcarlosh: There are still alot of variables involved
07:37.28carloshQwell: how would you measure the usage per 10 users or 100 ?
07:37.45carloshQwell: please ellaborate.. what variables.. ?
07:37.52Qwellcodecs, amount of transcoding, etc
07:37.53*** join/#asterisk zoa (~zoa@pirus.securax.be)
07:38.14Qwellcarlosh: Thats like me asking "How fast can my car go if I push the pedal all the way down?"
07:38.21Qwellwithout me saying what type of car I have
07:38.49QwellIts something you'll have to test yourself, for your specific settings
07:39.25PTG1234Qwell: pretty fast in my case :)
07:39.32QwellPTG1234: Nobody asked you :p
07:39.56QwellHow's your coding going?
07:40.00carloshQwell: so, there are not yet benchmarks done i take it..
07:40.13zoayes, where is that new chan_sip ?
07:40.13Qwellcarlosh: I'm sure there are plenty done, but it doesn't work that way
07:40.18PTG1234Um well i got distracted this weekend :)
07:40.20zoayour deadline was last friday wasnt it ?
07:40.21zoa:)
07:40.23PTG1234but should have it done by tommorow
07:40.29Qwellisn't the chan_sip stuff done?
07:40.33PTG1234i need the dnc list as well, working on gettingt hat
07:40.36oejTell me more about the chan_osip
07:40.39PTG1234well a good chunk
07:40.51carloshQwell: R U one of the programmers?
07:40.53QwellPTG1234: Yeah, I was wondering if you were gonna remember the dns, heh
07:40.54Qwellcarlosh: no
07:41.02PTG1234the dnc you mean? :)
07:41.08Qwelldnc, yeah
07:41.09PTG1234technically we are exempt
07:41.09Qwelltypo ;p
07:41.17PTG1234so i am 50/50 rather i should use it :)
07:41.20PTG1234well maybe 75/25
07:41.37QwellI would use it, honestly.  I can explain my reasoning in a pm if you'd like to hear it
07:41.42PTG1234sure
07:42.53carloshQwell: U using * to handle lots of calls ?
07:43.11Qwellcarlosh: 1 simultaneous call
07:43.26*** join/#asterisk Verliba (~Miranda@250.1-14-84.ripe.coltfrance.com)
07:43.40carloshQwell: hmm... don't know what to say.. I initially thought (from youranswers) that you had the experience..
07:43.54zoaPTG1234, could you discuss your chan_sip changes with oej ?
07:44.02carloshI handle more than that.. ;o)  no offense..
07:44.03Qwellcarlosh: I watch alot.
07:44.19Qwellcarlosh: I've simply seen your question asked countless times.
07:44.20PTG1234discus how so? :) when i am done there won't be any original code in there :)
07:44.21PTG1234more or less
07:46.35carloshQwell: I'd say there should be by now an idea of the footprint (processor time/memory) per call... per one call using a particular protocol and codec..
07:46.46*** join/#asterisk kensuke (~bryan@rrba-146-111-08.telkomadsl.co.za)
07:46.53Qwellcarlosh: to be fair, you still haven't specified any details about the calls
07:47.02kensukemorning all ... can anyone reccomend a console based sip phone ?
07:47.10carloshSIP/ilbc no transcoding...
07:47.34*** join/#asterisk three55ml (~none@cpe-66-68-98-68.austin.res.rr.com)
07:47.35Qwellsee, thats a bit better
07:47.43PTG1234why would you use asterisk to handle alot of calls if you don't need transcoding
07:47.45carlosh:o)
07:47.58PTG1234thats whate ser is for, till ir elease my chan_sip :)
07:47.59three55mlAnyone with a SIP ATA interested in doing some beta testing?  I'll provide a free DID for temporary use.
07:48.20three55mlOr SIP phones.
07:48.24carloshcos you might need it only as an Open source quicker replacement to SER..  maybe..
07:48.45PTG1234ser is open source
07:48.51carloshi know
07:50.43*** join/#asterisk jonas (jonas@jonas.probe-networks.host.freenode)
07:50.53*** join/#asterisk Gunnar (~gunnar@34.80-203-209.nextgentel.com)
07:51.56wildcard0PTG1234, are you updating chan_sip to behave more like a traditional proxy?
07:52.10carloshI think that answers my question (thanks PTG1234)
07:52.56carloshHowever, if I want people to use the meetme functionality.. SER would not apply..  :(
07:54.55*** join/#asterisk ariel_ (~Ariel@dsl-20-177.cofs.net)
07:55.08*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
07:55.12smurfixAnybody knowledgeable WRT zaptel.conf? I want to declare spans for the third+fourth card (zaphfc) only, but ztcfg doesn't appear to let me do that
07:57.50*** join/#asterisk ckruetze (~nospam@131.8.dsl3.ip.foni.net)
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08:08.22*** join/#asterisk Mazda-MX5 (~root@220-130-142-43.HINET-IP.hinet.net)
08:08.32Mazda-MX5hi ~ every body .
08:08.47three55mlHey
08:10.40timecophm
08:11.14Mazda-MX5I have question , what mean the "callid" for sip.conf?
08:11.19Mazda-MX5thank you
08:11.22*** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
08:11.23Qwellcallerid?
08:11.31Mazda-MX5yes , is callerid
08:11.51Qwellthe name and number of the person calling you, or the name and number the person you are calling sees
08:12.16QwellThats by far the most interesting question I've seen.
08:12.23*** join/#asterisk oej (~oej@40.186.204.213.sol.worldonline.se)
08:12.59Mazda-MX5thank you , I see. callerid can same the sip context name ?
08:13.00*** join/#asterisk Aze` (~aze@host229-162.pool80105.interbusiness.it)
08:13.20Qwellhuh?
08:13.43three55mlMazda-MX5: The callerid can be set to anything you want.  It will show up to other SIP clients as whatever you set.
08:14.03three55mlMazda-MX5: So yes, it can be the same as the context if you so desire.
08:14.03Qwellthree55ml: That made sense?
08:14.11PTG123needs to be a # not a name
08:14.35*** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it)
08:14.39Mazda-MX5thank you ! three55ml , and Qwell , thank you very much
08:14.49three55mlIdeally, yes, it should be the extension :)
08:15.12Mazda-MX5I can use ${CALLERIDNUM} to know callerid number , right ?
08:15.21QwellMazda-MX5: yes
08:15.26three55mlYes
08:15.43smurfixIs there a way to get yor password reset in the bugtracker? Want to submit a patch and my old login seems to be ... old.
08:15.49three55mlBut keep in mind that you can easily change the caller ID information based upon what type of channel you're pulling it from.
08:16.21Qwellsmurfix: gotta ask an admin I think
08:16.25nesyshi folks ... there's someone with * and ccme? I've a call-forward problem from ccme to * via sip trunk
08:22.47oejsmurfix: The asterisk-bugs channel is where you find bug marshals that handle mantis. I'll check if you can change. What's your id?
08:23.25smurfixoej: Thanks -- found the old email with my password in an unlikely place. :-/
08:23.36oejOk, see you in the bug tracker
08:23.41smurfixoej: (Depending on cookies for your brain isn't a good idea sometimes ;)
08:23.43Qwell"unlikely place", email inbox?
08:24.51smurfixQwell: no, personal email in box from friends (i.e.. grossly misfiled)
08:24.57Qwellahh
08:24.59oejsmurfix: You need to disclaim all patches, regardless if they're trivial
08:25.59smurfixoej: Gah. With real physical paper, or does Digium accept GPG signatures?
08:26.03*** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au)
08:27.04oejReal physical paper on fax as far as I know. Sometimes a scan in mail to mark
08:29.52oejsmurfix: Why two bug reports?
08:30.39smurfixoej: Because my browser crashed sending the first one and I didn't think to check whether it succeeded
08:31.03*** join/#asterisk miller7 (~none@adsl49-static-gw1.access.acn.gr)
08:31.07oejWhich one do I close?
08:32.15smurfixoej: the one whose explanation sounds less clear to you
08:32.29smurfixthe patch is the same
08:32.40oejsmurfix: Closed the first one
08:32.45smurfixthanks
08:34.34miller7Any European here interested in USA numbers?
08:34.47miller7I have a new * box so if anyone's interested msg me
08:36.01*** join/#asterisk zilas (~zilas@c-24-30-75-206.hsd1.ga.comcast.net)
08:36.26zilashello
08:36.33zilasanybody still alive?
08:36.43three55mlzilas: A few people
08:37.16zilas:) good. I sent 5 hours trying to configure call parking...no luck
08:37.51three55mlSorry, I haven't played with it much.
08:37.58Zeeekzilas what's wrong?
08:38.05Zeeekwhat's to configure?
08:38.07*** join/#asterisk |Vulture| (~Vulture@152.238.204.68.cfl.res.rr.com)
08:38.09zilasits a simple thing
08:38.16zilashow you place on hold
08:38.21|Vulture|anyone here use nagios?
08:38.32zilasyou should punch #700?
08:38.38Zeeekyes
08:38.47Zeeekwhen you hit # you should hear "transfer"
08:39.01Zeeekif you don't, it's because you didn't include a 't' in the dial command
08:39.34zilasoh this may be the issue
08:39.59Zeeekcould be
08:40.02zilasok how should it look like in extension.conf?
08:40.11Zeeekshow application dial
08:40.26Zeeekthat will give you the options
08:41.03Zeeekthe trouble is that using 't' or 'T' will stop you from using the # key if you call an IVR
08:41.12Zeeekso you have to be careful
08:41.21zilasI dont have IVR
08:41.42ZeeekI mean if you call your bank or something
08:42.02zilasit makes sense
08:42.05*** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net)
08:42.18Zeeekif you used T in the dial and they say "now hit your pin followed by the # key" the poiund key will not be heard. Instead you'll hear "transfer"
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08:43.45*** part/#asterisk Kumbang (~ecvs@167.205.24.4)
08:44.41Zeeekso did you try a dial with the 't' ?
08:44.46zilashow it should look exten => 1,1,Dial???
08:44.55zilasunder [parking]?
08:45.05Zeeekwhat does it look like now?
08:45.19zilasall I have is include parking
08:45.24Zeeekno not under parking, in whatever context you want to put it in
08:47.00zilasfile:/media/cdrom/bad boys blue - euro hits 2000.mp3
08:47.00zilasfile:/media/cdrom/Bad Boys Blue - Gimmie, Gimmie Your Lovin'.mp3
08:47.07zilassorry
08:48.44zilasI have as an example exten => 1,1,Dial(SIP/phone1,20,tr) ????
08:48.54Zeeekthat's good
08:49.09Zeeekthat will work for parking if you RECEIVE a call
08:49.23zilaswhat does here phone1 stand for?
08:49.35Zeeekthe name of your phone
08:49.54zilaslike extension of the phone?
08:50.10Zeeekif you had a phone called 2002 that would be SIP/2002
08:50.27Zeeekit seems to me you should try to read up a little on the dialplan
08:50.46Zeeekhere are a few links:
08:50.48ZeeekStarter tutorial:
08:50.48Zeeekhttp://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
08:50.48Zeeekhttp://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
08:50.48Zeeekhttp://www.automated.it/guidetoasterisk.htm
08:50.48ZeeekTHE reference of the moment:
08:50.49Zeeekhttp://www.asteriskdocs.org
08:51.16zilasI know :) but its so puerly documented for dialplan.... I'll try on those urls
08:51.45Zeeekthe dialplan is explained in detail in the asteriskdocs.org book
08:51.47ZeeekThe dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls.
08:51.47Zeeekhttp://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650
08:51.50Zeeekand here
08:52.23zilasI read google on all call parking and you was the one who put me on a right direction
08:53.15Zeeekcall parking is so simple that it is complicated!
08:54.04zilasI figured that :-)... I am reading this good Guide to Asterisk VOIP man its so goofy
08:54.34ZeeekI think the best thing to read is this: http://www.automated.it/guidetoasterisk.htm
08:54.43Zeeekit has most of the stuff you want to start wiht
08:55.08zilasso what you need to park a dialed call?
08:55.20ZeeekT instead of t
08:55.30Zeeekyou can put both to test rtT
08:56.24ronnhi guys
08:57.06ronni have been trying to use g726 codec on a supura 2000 with asterisk .. but always get incompatible codec error
08:57.15ronnanyone having the same problem?
08:57.57*** join/#asterisk cced (~wangxinta@222.33.36.198)
08:58.41zilaszeek: ok if I have 4 phones I write my example 4 times correct?
08:59.16Zeeekzilas you could do that, yes. Or you could use a macro that does it
08:59.51zilasthats more complicated already I just need to start with simple anyway THANKS A LOT!
09:00.54Zeeekyeah, once you get the four working you can look into macros
09:04.28*** join/#asterisk Delvar (~irc@83.146.53.34)
09:07.44*** join/#asterisk langals (~icechat5@196.7.14.183)
09:12.58langalshi there....wondering if someone could answer a few questions on use of the g729 codec?
09:13.39WilliamKlangals, ask away
09:13.48langalsI am using meetme conferencing. All of my phones will be using g729, so do I need to license g729 on Asterisk?
09:13.58WilliamKyep
09:14.22langalsEven though there is no transcoding being done?
09:14.59WilliamKif you're connecting to the asterisk box using g729 and asterisk is having to speak back on g729 then you gotta license it
09:15.15timecoper
09:15.19*** join/#asterisk kks (~kks@203.115.208.140)
09:15.22timecopthis aserisk flash panel shiz seems to be rather limited
09:15.31timecophow the hell do I tell it to rename a trunk of SIP buttons with their Caller ID?
09:16.01timecopits kinda useless to see 10 open sip channels all pointing into a conference and not being able to know who is on that channel
09:16.05langalsWilliam - ok - another question: Is it possible to use g729 for non-commercial purposes without buying licenses?
09:16.21zigmanlangals njo
09:16.23zigmanno
09:16.26langalsI know there is a non-commercial version, but I believe this is not as good quality?
09:16.33*** join/#asterisk E818 (anonymous@rrcs-24-199-5-190.west.biz.rr.com)
09:16.52langalsSo one would need to buy licenses in order to enable it?
09:17.18timecopright
09:17.52langalsAnd then on the client side - does anyone know anything about licensing g729 on windows?
09:17.59ardAs far as I know, this is a software patent. That means it doesn't matter: you need a license to use it, even for non-commercial purposes.
09:18.42ardLicensing starts at 15k $ :-)
09:18.54langalsard - so it is not like mysql, where it is based on trust that you get a license when you go commercial?
09:19.11ardmysql is GPL, you buy support.
09:19.27ardUnless you need some extra features.
09:20.02ardBut digium made a binary codec, and made that $15k deal, so you can buy from them a license for a single channel for $10 or so.
09:21.32WilliamKlangals, no because it's patented technology, and the owner of the patent wants his/her royalties
09:21.33ardActually you can make the codec yourself (use source from intel). And if you tell nobody, your ok :-)
09:22.34*** join/#asterisk Qorky (~goaway@202.173.160.18)
09:22.54langalsard - where do I get that - don't get me wrong, I do mean to license when going commercial, but I want to try it out first to make sure it works with client
09:24.26WilliamKlangals, get a couple licenses then and build on to it later
09:24.38ardhttp://lists.digium.com/pipermail/asterisk-users/2004-January/035492.html
09:24.49WilliamKthis is interesting
09:24.50WilliamK=)
09:25.00ardthe easy way out is really buying that license from digium
09:25.29ardyou can write the codec yourself, but that means your only save in the EU for now... or china
09:26.21langalsI am in South Africa
09:26.50ardhttp://www.voip-info.org/wiki-Asterisk+G.729+licensing
09:26.55ardHmmmm
09:27.11ardI dunno about south africa
09:29.10Qorkyanyone have some example configs for connecting two asterisk servers with iax ?
09:32.10ZeeekQorky what do you want to achieve specifically?
09:34.07ronnhas anyone managed to use g726 codec on a supura 2000 with asterisk?
09:38.57QorkyZeeek. I have 2 offices. wonna be able to talk between them.
09:39.22Qorkyone office is 10XX numbers. the other is 11XX numbers. ideally i want to leave the option of adding a third later.
09:39.27ZeeekI would say you can just set them up as peers
09:39.54h3xi think digium's codec does g.729b too
09:39.58Zeeekand route the calls by the first two digits as you have planned
09:40.32Qorkyyar. makes sence. any config example or howtos around. maybe on the wiki ?
09:41.39ZeeekI think if you just "pretend it's FWD" it'd work!
09:41.54Zeeekyou know, standard peer entry
09:41.59Zeeekin iax.conf
09:42.15Zeeekthere was also a switch keyword you might want to look up
09:43.59Qorkyah yeah ok. let me see.
09:50.55*** join/#asterisk TheEmperor (~mattn@203.121.47.100)
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10:06.06tainted-anyone have issues where sip channels don't go away after hangup?
10:10.29*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
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10:18.59facek_anyoen alive?
10:19.12RestLessGeminiyup
10:22.34facek_what are the callgroups?
10:24.16|Vulture|facek_: in zap configs?
10:24.53facek_yes
10:25.05facek_and how can i connect groups with peers from iax and sip.conf
10:25.06|Vulture|call groups if you mean in zapata.conf are just groups that are set for a dial out
10:25.11*** join/#asterisk cced (~wangxinta@222.33.36.198)
10:26.06|Vulture|facek_: not following.. but groups are like instead of Dial(Zap1/555) you can issue Dial(Zap/g1/555)
10:26.14|Vulture|something like that... I am kinda tired lol
10:31.22*** join/#asterisk brc-tux (~brc-tux@p54A98D54.dip0.t-ipconnect.de)
10:31.30*** join/#asterisk lters (~lters@mrtc-mm-600046.mis.net)
10:33.20ltershow soon will we see the 1.2 release ?
10:33.46lterscan't wait for the new options :)
10:37.43*** join/#asterisk RoyK (~roy@83.223.171.239)
10:40.25*** join/#asterisk _Crash_ (~me@adsl3p183.access.maltanet.net)
10:40.41_Crash_Can anyone help me with an MOH issue I have?
10:45.16zoa_crash_ what is the issue ?
10:48.29_Crash_Hi zoa...is there some specific setting to save MP3s for MOH?
10:52.57zoayeah
10:53.05zoano vbr for a start
10:53.16zoatry old formats
10:53.17zoano vbr
10:53.20zoa128kbit
10:53.23zoawill probably do fine
10:53.23*** part/#asterisk brc-tux (~brc-tux@p54A98D54.dip0.t-ipconnect.de)
10:53.41_Crash_10x....
10:53.57_Crash_Would you know of there is some bug reporting list for AMP?
10:55.05|Vulture|AMP... has a bug?! *gasp* :P
10:57.49E818help!
10:57.50E818asterisk: relocation error: /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_cust_config_register
10:58.06pifhi, if a phone sends back a 302 redirect to an outside number, how can I have * refuse that?
10:58.10E818is there a problem with the current cvs version?
10:58.56_Crash_What are the viable alternatives to AMP...?
11:08.36three55mlAnyone know how to enable SIP realtime caching?
11:09.27*** join/#asterisk marks__ (~marks__@cpe-70-112-81-84.austin.res.rr.com)
11:09.39|Vulture|_Crash_: for a household user AMP is fine, but for any business AMP should not be used, its just sloppy if you know how to do conf files yourself
11:12.32_Crash_|Vul|: Yes can edit conf directly but is there "the" business AMP equivalent?
11:13.02facek_i am working on that ;]
11:14.57three55ml_Crash_: I'm working on one, to a degree
11:19.10facek_do you have soem interesting dialplan for an example?
11:20.46three55mlI'm doing more of an end-user interface, I'm splitting off the UI I've developed for a service I'm working on to it's own program.  It looks like this: http://www.premierpbx.com/images/screenshots/2.jpg
11:25.30*** join/#asterisk Corydon76-home (twelve@pcp08665860pcs.500ash01.tn.comcast.net)
11:27.52*** join/#asterisk Darwin35 (~Darin@24.3.226.147)
11:27.59*** join/#asterisk gambolputty (~gambolput@cblmdm65-221-51-80.buckeye-express.com)
11:30.54*** join/#asterisk lilo (lilo@levin-pdpc.staff.freenode)
11:32.23*** join/#asterisk mithro (~tim@dsl1-83.gw1.adl1.airnet.com.au)
11:32.47*** join/#asterisk JohnJacob (~JohnJacob@pcp0011542342pcs.mainf01.in.comcast.net)
11:33.09mithrohello, if i could produce an 8 port FXS device for under $100 US, do you think it would be a good deal?
11:33.33InfraRedi'd buy one
11:34.38mithroit would be a USB based system with the host computer software doing most of the work
11:34.45InfraRedheh
11:36.08mithroit would be basically a simple USB PIC with 8 DACs and some DC/Ringing logic
11:37.01mithroso DTMF detection, voice encoding, etc would all be software
11:38.37*** join/#asterisk webman (~adamg@202-44-171-5.nexnet.net.au)
11:38.39h3xTheres 4 port fxs to ethernet already
11:38.48h3xit costs maybe a bit more but hey at least you dont need a computer
11:39.07Wonkamithro: do you intend to put the plans open?
11:39.14mithroyeah but they are all expensive ie > $300 US
11:39.19mithroWonka: most probably
11:39.26Wonkah3x: you need a computer to run * already...
11:39.30h3xnah ive seen one for less than 200
11:39.31Wonkacool
11:39.34h3xWonka: not if its a remote location
11:39.41h3xor service provider model
11:39.42tainted-three55ml that looks a lot like switchvox
11:39.46mithroh3x: where?
11:39.54webmananyone used wcusb with linux 2.6.x ?? I seem to get a segv when running ztcfg ?
11:39.59h3xwell a taiwanese company makes a 4 port
11:40.01h3xummm
11:40.03h3xrepotec.com
11:40.34h3xive never used it but i have some of their ethernet switches
11:40.36mithroif we pushed for mass production we could quickly drop the price
11:40.46h3xand they are of decent quality and cheap
11:41.35tainted-mithro how much would it be to include the other stuff
11:41.42tainted-mithro voice encoding etc
11:41.53mithrotainted-: quite alot
11:42.08tainted-licensing costs or chip costs
11:42.13mithroh3x: where can you buy them
11:42.20mithrotainted-: both
11:42.35h3xive got a chinese importer in walnut california that i buy the other stuff form
11:42.35h3xfrom
11:42.41h3xill have to ask them about that box
11:42.51tainted-h3x are u in socal?
11:43.00h3xno im in vegas
11:43.10tainted-which importer?
11:43.14h3xpi
11:44.13tainted-pi?
11:44.20h3xyes, P.I.  thats what its called
11:44.20tainted-just pi?
11:44.20h3xheh
11:44.26tainted-oh
11:44.59h3xwho knows its probably some mandarin acronym for "stupid roundeye"
11:45.00h3x:)
11:45.11tuxinator_linuxhe he
11:45.13Supaplexlol
11:45.34h3xI've spent 10 grand with those guys
11:45.36mithroi could most probably do a single USB FXS device for under $30 US
11:45.37h3xin the past 8 months or so
11:45.53tainted-mithro but where do u see the need?
11:45.56h3xthey even carry fiber stuff
11:46.03SupaplexI'm courios, how do you do the ringing circuit?
11:46.04tainted-repotec looks pretty cool
11:46.27h3xthats it, im emailing my sales rep now and asking her if they have these things
11:46.40mithroi see the need because currently it costs me ~$100 for 2 FXS ports
11:47.01tainted-why USB
11:47.02h3xbuy Sipuras
11:47.06h3xthey are only $83 for two ports
11:47.11Supaplexyea
11:47.19tainted-i'm waiting for someone to embed asterisk in a soekris box or something
11:47.26tainted-that'd be cool
11:47.32three55mltainted-: It's been done
11:47.46tainted-three55ml your stuff looks like switchvox
11:47.51tainted-three55ml where?
11:48.02three55mlThere's some info about it on the wiki
11:48.36three55mltainted-: I actually wrote it before Switchvox even came out.  I think the UI is infiniately more friendly than Switchvox's as well.  Two different target audiences I think.
11:48.40h3xthe real question is
11:48.43h3x*drumroll*
11:48.48h3xdoes that repotec REALLY support T.38
11:48.56mithroif i could do $100 for 4 FXS ports, it's better then $83 for 2 FXS
11:49.04three55mlhttp://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware%20Soekris
11:49.36tainted-nDuff u can roll your own in a few hours
11:49.36tainted-oh wait.. hardware phone
11:50.05h3xmithro: maybe, but the sipura rules.
11:50.09tainted-no way manu would use openvpn.. they'd prefer something proprietary and sell it as a 'premium feature'
11:50.20h3xit has a dsp in it
11:50.24h3xso the echo can actually works right
11:50.29h3xand it supports a shitload of codecs
11:50.43webmanI think the PA1688 chipset phones support IAX2, and probably could be convinced to support openvpn ... if they can take the CPU encryption penalty
11:53.40cypromisthey could
11:53.42h3xRepotec's RP-ID162 is interesting also
11:53.57cypromisbut they still miss some iax2 features, which is no wonder with the current state of documentation and specification of the protocol
11:54.29h3xRepotec's stuff is actually pretty solid for ethernet anyway, the metal casings are heavy duty, and I opened up an ethernet switch and it was a good quality circuit board and broadcom chipset
11:54.37h3xwith heatsinks and everything
11:55.03h3xI bought a gigE 8 port switch for $80 from california
11:55.09h3xand thats qty 1 of course
11:55.21h3xthey have some rack mount stuff with 802.1q vlans etc
11:55.57tainted-three55ml
11:55.59tainted-http://www.premierpbx.com/products.php
11:56.11tainted-ever heard of 37signals?
11:56.12tainted-http://www.37signals.com/
11:56.35three55mlYep :)  I'm actually redoing the site to match the look of the internal program.
11:56.58tainted-which site will the internal program match?
11:57.31three55mlThe main site will match the look of the control panel - http://www.premierpbx.com/images/screenshots/2.jpg
11:59.38h3xis that supposed to be one of those nortel phones
11:59.52h3xwell
11:59.53h3xaastra
12:00.15tainted-that looks a lot better than http://www.switchvox.com/sv?cmd=screenshots&pic_id=8
12:01.03three55mlI think sticking with the Asterisk terminology and conventions is way too confusing to end-users.
12:01.28tainted-i agree
12:01.31three55mlI'm pondering dropping the term "Call route" all together in lieu of something simpler.
12:01.35tainted-your landing page it top notch
12:01.42tainted-very user centric
12:01.47tainted-s/it/is
12:01.50file[laptop]an hour late!
12:02.01file[laptop]stupid daylight savings
12:02.02three55mlThere's also an admin panel on top of it all that does allow for more of a "raw" interface, but nothing the end-user needs to see.
12:02.18three55mlfile[laptop]: Haha, you're a day off.
12:02.47*** part/#asterisk RazaMetaL (razametal@pc.gsalas.manta.telconet.net)
12:02.52file[laptop]I'll miss my first period, but I can get there for the rest of the day
12:03.02tainted-first period!?
12:03.05tainted-how old are you!
12:03.08file[laptop]18
12:03.15file[laptop]:p
12:03.16tainted-jesus
12:03.26three55mlMan, I haven't even gone to bed yet.
12:03.37three55mlFigured I'd stay up, I have to go downtown and sign some papers at 10 anyways.
12:03.59tainted-you're going to be wasted in a few hours
12:04.04tainted-a walking zombie
12:04.18file[laptop]or I can like, not go to school today
12:04.30file[laptop]but that's not advised
12:04.30three55mlPossibly.  I'm used to it, I usually play poker 1-2AM until 6-7.  Hit up all the drunks.
12:04.39*** join/#asterisk dwmw2_gone (dwmw2@baythorne.infradead.org)
12:04.45*** join/#asterisk SkyTel_DK (~skytel@cpe.atm2-0-1041166.0x50a3c79a.albnxx14.customer.tele.dk)
12:05.33tainted-ok supermen
12:05.50SkyTel_DKcan anyone help me on pri i get an error on my te410 card on one of the spans that say : Apr  4 14:03:23 NOTICE[-1250526288]: chan_zap.c:7379 pri_dchannel: PRI got event
12:05.50SkyTel_DK: HDLC Bad FCS (8) on Primary D-channel of span
12:05.56tainted-i'm off to dream about dialplans and channels in never never land
12:06.24three55mlLater
12:06.26*** join/#asterisk CleanerX (~nix@nat-ph3-wh.rz.uni-karlsruhe.de)
12:06.35webmanSkyTel_DK: AFAIK, this is 'normal' if it happens rarely (ie, once every few hours/days)
12:07.02SkyTel_DKno i get it every 5-10 secs
12:07.48SkyTel_DKcould it be bit errors on the e1 side ???
12:09.26webmanSkyTel_DK: well, I have/had the same problems, though I get it together with other error messages, and frequant dropped calls....
12:09.43SkyTel_DKokey
12:10.01webmancurrently, I've been led to believe that it is due to a faulty 'timer' chip on the board...
12:10.27webmanbut then, I got my board when they were first released, one of the very first ones....
12:11.19*** join/#asterisk HoppaZ (~HoppaZ@xdsl-81-173-148-174.netcologne.de)
12:11.23webmanhopefully tonight I will talk to digium and see if they'll still replace it for me.... (hope so, those things are expensive)
12:11.24SkyTel_DKhmmm just that i have it on 2 te410 cards on the same span IF i move the pri from the public side the the other card'
12:11.25*** join/#asterisk Mother__ (~m@53.Red-217-126-93.pooles.rima-tde.net)
12:11.27Mother__greetings
12:11.33HoppaZhi
12:11.50webmanskytel_dk: are you sure you are taking timing from the right pri/span ?
12:11.50Zeeekhello Mother__
12:12.05Mother__question: just bought 15 g729 licenses off Digium, but they didn't ask for any MAC address, how does it work afterwards?
12:12.07HoppaZgot problems with h323 and asterisk... no sound with gnomemeeting
12:12.08Mother__hiya Zeeek
12:12.11*** join/#asterisk Damin (~damin@nucleus.nacs.net)
12:12.18SkyTel_DKyes i take the timing from the rigth one
12:12.30webmanmother__: just run the register, and it will get the MAC address the first time....
12:13.13florzDo these licenses work with tap interfaces, too, BTW?
12:13.14Zeeekthen just don't change the MAC !
12:13.19webmanskytel_dk: do you get the same problem if you run a crossover on the same card (ie span 1 <=> span 2) with no other PRI connected, and span 1 provides timing to span2 ??
12:13.32Mother__webman: OK, but how does it make sure I'm the licensee? I should stick some reference or serial or something somewhere no?
12:13.48webmanzeek: you can change the MAC address once....
12:14.05SkyTel_DKZeeek if i do a cross over the error folows the e1 kabel
12:14.08webmanmother__: you are emailed a registration key, which you need to give to the register program
12:14.08Mother__yes, I read that you can re-register the MAC address once if you have to, but does it work on good faith?
12:14.12Mother__AH!
12:14.19Mother__webman: OK, that clears it up, thanks
12:14.24SkyTel_DKi have 2 * with te410 in it
12:14.35SkyTel_DKit = them
12:15.02webmanmother__: AFAIK, it works automatically the first two times (with different MAC address) then after that you need to contact digium to discuss it...
12:15.32webmanskytel_dk: what do you mean it follows the cable?
12:16.40SkyTel_DKlets say that cabel 1 is the one with errors on and cabel 2 is error free in the other asterisk with te410 in
12:17.34SkyTel_DKthen if i take the cabel 2 and put it in the 1.st asterisk and the cabel 1 into the 2.
12:17.40*** join/#asterisk viLeR (1000@ip-47-252.telesat.com.co)
12:18.02SkyTel_DKthen the 1.st asterisk is error free but the errors have moved to the 2.nd one
12:18.34webmanskytel_dk: yes, but what I asked is, if you config one asterisk box, so that span 1 is pri_net and span 2 is pri_cpe, and connect them with a crossover cable, do you get this error?
12:19.11SkyTel_DKahhhh
12:19.18SkyTel_DKi have not testet that
12:19.20webmanskytel_dk: anyway, sounds like the error is in the cable, so check your physical cable from the telco to your asterisk, and then ask your telco to test the line
12:19.36SkyTel_DKyeah
12:19.37*** join/#asterisk FireFoxIXI (~FireFox@floyd.gms.lu)
12:19.47SkyTel_DKit can be the cabel
12:22.48*** join/#asterisk shodan (~shodan@216.113.99.220)
12:23.53Mother__when an error follows a piece of hardware around, it's quite possible that the piece of hardware is the problem...
12:24.32Mother__also, if you use a test tool that always reads failures on different used & new equipment, it may be the test tool has failed itself
12:25.00Mother__I once saw a good number of VGA cards go to the crapper because they were testing them on the same faulty motherboard
12:25.05Mother__dumbasses
12:25.30ZeeekMother__ what about when an error follows a user around? :)
12:25.53Mother__oh, that happens a lot, then I get my Rhyno and label the person as a (L)user
12:25.58Mother__:)
12:26.15*** join/#asterisk pjm_uk (~pjm_uk@cpc1-pool3-3-0-cust116.sot3.cable.ntl.com)
12:26.23SkyTel_DKdo any of you have any idea how i can get a dial tone using asterisk and my te410 card i have problems dialing over sea calls do to the fack that when i press 00 on my set to dial overseas the pabx sends the 2 leading 00 to the pri and i get a dial time out
12:26.27Zeeekhave you ever watched frustrated users retype the same command after an error?
12:26.34Mother__lol yeah
12:26.41Mother__as if typing it nth times will make it work
12:26.47Zeeekthen they type it in all caps to see...
12:27.34Mother__SkyTel_DK: is your dialplan OK? I'd check that first - can you dial other numbers OK?
12:27.44SkyTel_DKall okey
12:27.56Mother__so you can dial local numbers?
12:28.21SkyTel_DKi got it to work if i set overlapdialing to yes
12:28.44SkyTel_DKbut that is not to rigth way to do it
12:29.12*** join/#asterisk devi (~dev@gw.01063telecom.de)
12:29.18devihello everyone
12:30.06Mother__hmmm found this http://lists.digium.com/pipermail/asterisk-users/2003-July/015351.html
12:30.54SkyTel_DKthe reason it works lokaly is that the pabx does not send any digits to the pri before the dialplan legth is reached but it sends the 00 to the pri only if it is international calles
12:31.22Mother__hmmm but why is it doing that? it's gotta be something in the dialplan no?
12:31.39SkyTel_DKnope not in asterisk
12:31.53SkyTel_DKits done from the pabx
12:31.56Mother__I use two context for international and local calls personally
12:32.03SkyTel_DKokey
12:32.03Mother__AH, you have a sepparate PABX?
12:32.08SkyTel_DKyes
12:32.16SkyTel_DKan eads
12:32.47SkyTel_DKi lead it into the asterisk and then out to the publicexchange
12:32.55Mother__righto
12:32.58SkyTel_DKon 2 spans
12:33.41Mother__try to use two contexts with the two dial lengths
12:33.45SkyTel_DKso i use the asterisk as a pipe
12:34.13Mother__in mine I have on which is 00XXXX...... and another for local/national numbers (all begin with 9)
12:34.48Mother__it helps if you want to handle things sepparately at some stage, I like to keep things on sepparate contexts
12:35.26SkyTel_DKokey
12:35.37SkyTel_DKbut in dk its
12:35.55SkyTel_DKZXXXXXXX for dk
12:36.40SkyTel_DKand 00Z. for international
12:37.19SkyTel_DKbut the pabx sends first  the 00 then the rest of the digits
12:37.25smurfixanybody know their way around mISDN? my chan_misdn driver doesn't want to start
12:38.03SkyTel_DKit its not in overlap mode
12:38.40SkyTel_DKin overlay mode it sends the hole digit string at once
12:39.34SkyTel_DKbut if for some reason the user sends a wrong number its takes a long time before it times out
12:39.55Zeeekcheck this when you have time to waste
12:39.56Zeeekhttp://www.msnbc.com/modules/airport_security/screener/default.asp
12:40.02SkyTel_DKbut in te way that it sends first the 00
12:40.07*** join/#asterisk forkqueue (~sam@spc1-ward2-5-0-cust27.bagu.broadband.ntl.com)
12:40.36Mother__OK, how long between it sends the 00 and the rest of the digits?
12:40.47Mother__because maybe you'll need to change digit timeout in *
12:40.48*** part/#asterisk FireFoxIXI (~FireFox@floyd.gms.lu)
12:41.04SkyTel_DKthe publicexchange listens to the digits that are sendt 1 by one and test it for wrong nummer rigth away
12:41.59Mother__well, that's the exchange's job, but I don't think * will transparently send digits between your PABX and the rest of the world as they are dialled
12:42.00*** join/#asterisk jakepdev (~jakepdev@pool-68-163-51-71.phil.east.verizon.net)
12:42.10Mother__I may be wrong there though, don't take it for granted
12:42.37Mother__AFAIK when a dial string is complete and it matches an entry in extensions.conf then the extension sequence is executed
12:43.03SkyTel_DKyeah thats what i mean
12:43.06*** join/#asterisk airios (~andres@66.28.87.10)
12:43.29SkyTel_DKcant you just put some thing in to halt the execution
12:44.17Mother__in which case, if it's invalid, * will still dial it and the telco's exchange will send back a response, either tones or a nice voice telling you to rotate your phone 180 degrees and dial again
12:44.20Mother__;)
12:44.33Zeeekheh
12:44.53Mother__so your PABX is sending the 00 then waiting for something? or it sends the 00 then pauses for x seconds then sends the rest?
12:45.10*** join/#asterisk mentat (~Mentat@pcp01260498pcs.nhaven01.ct.comcast.net)
12:45.25SkyTel_DKno it waits
12:46.02Mother__so it waits for something to come back from the PRI saying "shoot, I'm ready?", then as * doesn't send anything it times out¿
12:46.18SkyTel_DKits goes into call mode and waits for the rest of the digits 1 at a time
12:46.29*** join/#asterisk ckruetze (~nospam@131.8.dsl3.ip.foni.net)
12:47.00Mother__then maybe you could increase digit timeout in the * conf
12:47.41Mother__I had to increase it on one I have installed as the (L)users took too long to dial...
12:49.02SkyTel_DKyes but that should not be nessesary since the pub-Xchange dials the b caller 1 digit at a time
12:49.53SkyTel_DKasterisk sould be able to send the 2 00's and then go into dtmf mode
12:50.29Mother__dammit, gotta go
12:50.31SkyTel_DKbut since it sends a "no more digits"
12:50.33Mother__bbl
12:50.38SkyTel_DKcu
12:51.07airiosiaxy question: how i look at the debug information in port 9999 ( when i set debug in the conf )?
12:51.09SkyTel_DKor end of digit string the pub_xchange hangs up
12:53.28vaewynAFKSkyTel_DK: You could have a   exten => 00,....  and it should do something akin to that
12:53.59SkyTel_DKcoma
12:54.02SkyTel_DKokey
12:54.30vaewynthe comma is just the delimeter
12:55.09SkyTel_DKso it is not the same as . "dot"
12:55.28vaewynnow bummer is ... if you do that you get all your cdr logs as calls going to  '00'
12:55.56vaewynsorry... to be more clear it would be   exten => 00,1,Dosomething()
12:56.10SkyTel_DKaahhh
12:56.30vaewynbut it screws up your CDR on * if you are using that for billing
12:56.56*** join/#asterisk yaboo (~jsirucka@220.245.131.131)
12:57.15SkyTel_DKthat is testet i did  00,1,dial(zap/g2/00) but it then hangs up since there is sendt a end of digits
12:57.38airiosanybody with experience using IAXY modems?
12:58.05h3x<PROTECTED>
12:58.20bjohnsonSkyTel_DK: sounds like the problem stems from the pabx you're using
12:58.28*** join/#asterisk Martohtar (Martohtar@82.196.218.80)
13:00.00*** part/#asterisk HoppaZ (~HoppaZ@xdsl-81-173-148-174.netcologne.de)
13:00.54SkyTel_DKbjohnson i dont think so because when i put the pabx direktly into the Pub_Xge it works fine
13:03.34*** join/#asterisk jcims (~jcims@cpe-24-210-60-100.columbus.res.rr.com)
13:06.32dwmw2_goneSkyTel_DK: that works for me when dialling out via mISDN or CAPI
13:07.13dwmw2_goneexten => _X!,1,Dial(mISDN/g:extern/${EXTEN})
13:10.56nesysthere's someone expert on sip that could help me? I've an issue about call-forward between ccme and *
13:12.03*** join/#asterisk shodan (~shodan@216.113.99.223)
13:15.30SkyTel_DKhmmm
13:15.40SkyTel_DKwhat is mISDN
13:15.40shido6:)
13:16.54*** join/#asterisk klictel (~klictel@207.107.208.137)
13:16.58klictelmorning all
13:17.18shido6yep
13:17.24*** join/#asterisk bsdfreak (ninja@enterthebass.com)
13:18.17*** part/#asterisk jcims (~jcims@cpe-24-210-60-100.columbus.res.rr.com)
13:20.28airiosh3x: yes, IAXY  modems?
13:20.50shido6?
13:20.55shido6what are you doing airios
13:20.56shido6?
13:21.27airiosthe question is how to fetch the debug info when the debug option is given in the config file?
13:22.44airiosi see the pacets bing broadcasted in port 9999, how i parse them/see them?
13:22.47*** join/#asterisk LoRez_ (lorez@lorez.staff.freenode)
13:23.38shido6tcpdump, ethereal
13:23.44shido6pick your poison
13:23.59*** join/#asterisk malverian (~malverian@adsl-065-005-207-210.sip.gnv.bellsouth.net)
13:24.45airiosyes ... i am using both, but aren't those packets menat for something that can parse them?
13:25.02airiosmeant
13:25.10shido6what are you going to do with the info?
13:25.14shido6did you misconfigure your IAXy
13:25.15shido6?
13:25.51airiosi am trying to see what the modem is doing, as i am not sure if it register with my astersik server
13:26.02shido6if the light isnt on
13:26.04shido6it didnt
13:26.33airioswhich light?
13:26.34shido6at the CLI if you do a iax2 show peers
13:26.42shido6and dont see your IAXy there
13:26.46shido6then the IAXy did not register
13:26.50airiosdont see it with show peers
13:26.55shido6then its not registered
13:27.17shido6there are 2 lights on the IAXy ( well 4 if you count the link and xmit lights )
13:27.17airiosmmmm ... so that is the point to see the debug
13:27.27shido61 of which stays on when the IAXy is registered
13:27.33shido6........nope
13:27.40shido6configure your IAXy properly
13:27.47shido6show me your iax.conf
13:27.50shido6pastebin.ca
13:28.18airiosi saw that in the man, i saw the link lights in the ether , and another light that seems to be ligthing on when the phone is off the hook
13:28.24airiosis there another one>?
13:28.24yabooanyone got the soyo n400s fxs unit yet?
13:28.39*** part/#asterisk Zeeek (~icechat5@Zeeek.sustaining.supporter.pdpc)
13:28.40shido6what do you have in iax.conf for your IAXY, use pastebin.ca to show us
13:29.00dreamcodedoes anyone use musiconhold ?
13:29.14shido6whats wrong dreamcode ?
13:29.19webmananyone used the s100U (USB FXS) on 2.6.x kernel recently ??
13:29.32shido6heh
13:29.35dreamcodei don't hear any music when puting someone on hold
13:29.40webmanI get a sig segv when running ztcfg .....
13:29.42airiosit is textual from the iaxy config pdf
13:29.44shido6webman you got your hands on an old S100U, luck you
13:29.53shido6airios, that doesnt help
13:29.55shido6pastebin.ca
13:30.23webmanshido6: well, I bought it a long time ago... I'm just re-setting it up at home (I upgraded to the TDM shortly after they were first released)
13:30.41airiosok ... dont have it now, will do. THANKS!
13:30.51dreamcodeshido6, i have instaled mpg123 , MP3Player is working , but the MusicOnHold doesn seems to start.. why ?
13:31.02shido6ps -aux
13:31.04shido6no mpg123?
13:31.09shido6wrong mpg123, perhaps
13:31.09airiosone more quick question, there is a binary in the iaxyprov called iaxydebug or something, what is it?
13:31.27shido6this is such a problem its even got its own install command now in CVS
13:32.02webmandreamcode: what is the CLI output when you try to put some1 on hold??
13:32.03SkyTel_DKhas anyone got the cisco 7970G ipset to work under asterisk ???
13:32.16shido6cd /usr/src/asterisk
13:32.18shido6make mpg123
13:32.23shido6make install
13:32.28shido6SkyTel_DK
13:32.33shido6yes, my 7960 works well
13:32.38shido6as do my students and customers
13:32.44shido6whats happening with yours?
13:33.09SkyTel_DKi just got it
13:33.33shido6so its got skinny on it?
13:33.35shido6:)
13:34.02webman7970 I think is skinny only ...
13:34.16shido6yes, but 7960s new in the box
13:34.19shido6come wit skinny
13:34.22shido6+h
13:34.26dreamcodei don't have nothing on CLI
13:34.41shido6go to  /etc/asterisk/logger.conf
13:34.45shido6and look for "console"
13:34.50shido6then at the end ofo the line add , debug
13:34.55*** join/#asterisk lbarth (~lbarth@62.4.65.13)
13:35.08shido6and stop and restart asterisk with -vvvvvgcd   or /usr/sbin/asterisk -vvvvvgcd
13:35.16shido6or wherever you have asterisk installed
13:35.19shido6and check again
13:35.27shido6but more than likely you have the wrong mpg123
13:35.36shido6get mpg123 installed first
13:35.56shido6how many phones do you have SkyTel_DK
13:35.57shido6?
13:36.55SkyTel_DK1
13:36.58bjohnsonwebman: the s100u won't work on anything but a 2.4 kernel .. and that is according to digium
13:37.15*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
13:37.16shido6would you like me to upgrade your phone and get it working wit asterisk while you watch and ask questions?
13:37.21shido6hhh
13:37.35*** join/#asterisk TheEmperor (TheEmperor@218.111.51.87)
13:38.30Moc____mornuing
13:40.20ariel_morning all
13:43.31dreamcodeshido6, but.. if MP3Player command works in extensions.conf , shouldn't musiconhold work too ?
13:44.00shido6grrrrrr.......
13:44.09shido6what does /etc/asterisk/musiconhold.conf say?
13:44.11shido6pastebin.ca
13:44.19*** join/#asterisk markak2 (~twist@ndn-165-134-119.telkomadsl.co.za)
13:44.23markak2hi all
13:46.35dreamcodehttp://pastebin.ca/8772
13:46.37markak2something strange has happened on my asterisk. for some reason this was working but now just gives an open line. it used to remember the ${EXTEN} and dial the number correctly. this is a part of that que for new calls i was working on.
13:46.39markak2[dialnow]
13:46.39markak2exten => s,1,ChanIsAvail(Zap/2&Zap/1)
13:46.39markak2exten => s,2,Cut(theChannel=AVAILCHAN,,1)
13:46.40markak2exten => s,3,Dial(Zap/g1/${EXTEN:1},20,Ttm)
13:46.40markak2exten => s,4,Hangup
13:46.42markak2exten => s,102,Playback(allbusy)
13:46.44markak2exten => s,103,Wait(5)
13:46.46markak2exten => s,104,Goto(dialnow,s,1)
13:46.54newlahh, shido6, just the person that might have an answer to a perplexing question.  Scenario is this, two internal extension X and Y, extension X is diverted, extension Y calls extension X which diverts, CDR for the diversion gets entered for Y, not X.  Bug, or feature?  If bug, I can file a report.  If feature, how to make X take the CDR entry instead of Y? :)
13:47.00shido6holy paste
13:47.02newlmarkak2: please use pastebin.
13:47.09markak2sorry
13:47.14shido6dood
13:47.17nvrsworkthat guy is out of control
13:47.17shido6s?
13:47.33shido6what are you cutting off the exten line with the exten:1
13:47.36shido6Ttm ?
13:47.52shido6careful as both the called and the calling can xfer if that ever worked which I doubt..
13:47.57markak2supposedly the preceding 9 they dial to get here.
13:48.07shido6what is that supposed to do markak2, what are you LOOKING to do ?
13:48.17markak2exten => _9.,1,Goto(dialnow,s,1)
13:48.26*** join/#asterisk bkw_ (~bkw_@bkw.developer.and.friend.of.asterisk)
13:48.26*** mode/#asterisk [+o bkw_] by ChanServ
13:48.30markak2we have 2 pstn lines
13:48.50*** join/#asterisk langals (~icechat5@196.7.14.183)
13:48.53markak2but too many people using them. if you dial and all lines are busy it notifies you and then connects you when a line is open.
13:49.19shido6dreamcode
13:49.24shido6dreamcode, http://pastebin.ca/8773
13:49.26markak2but 2 days ago instead of connecting you to the number you dialed it just connects you to an open pstn line.
13:49.29langalsjoin #freenode
13:49.41*** part/#asterisk shodan (~shodan@216.113.99.223)
13:49.44*** part/#asterisk langals (~icechat5@196.7.14.183)
13:50.09shido6what is calling these s extens in dialnow
13:50.31shido6oh ok
13:50.34shido6exten => _9.,1,Goto(dialnow,s,1)
13:50.35markak2exten => _9.,1,Goto(dialnow,s,1)
13:50.40markak2yes
13:51.01nesysswitch works only between * servers?
13:51.06markak2the only problem is that it seems to lose the dialed number.
13:52.00shido6thats actually pretty cool.
13:52.10shido6whats the CLI say?
13:52.26dwmw2_goneSkyTel_DK: mISDN is just another way of using ISDN BRI cards.
13:52.32shido6what has changed in the past 2 days?
13:52.42shido6what could have changed in the past 2 days?
13:53.23markak2strangely nothing i can remember doing
13:53.59markak2http://www.pastebin.com/266954
13:54.04markak2all normal just loses the number
13:54.55markak2this was a call when a line is open and it connects me immidiately
13:55.00dreamcodeyes, shido6 , still doesn't work,my mpg123 is Version 0.59s-mh4 (2000/Oct/27), and doesn't work, and what is strange.. that i don't have any message on console when i put someone on hold
13:55.15shido6ok
13:55.16shido6-- Executing Dial("IAX2/2001@2001-6", "Zap/g1/|20|Ttm") in new stack
13:55.16shido6<PROTECTED>
13:55.21*** join/#asterisk pif (ldm@zenon.apartia.fr)
13:55.26shido6what the heck are you "calling" here?
13:56.25markak2Zap/g1 is the group for the ZAP channels it should be follwed by the number ${EXTEN:1}
13:56.34shido6err
13:56.38shido6thats not in the CLI
13:56.51markak2i know
13:57.01markak2its getting lost somewhere ?
13:57.23markak2exten => s,3,Dial(Zap/g1/${EXTEN:1},20,Ttm) this is the line
13:58.32shido6zapata.conf shows what group=1 is , right?
13:59.03mishehudoes anybody know why using delayreject=yes in iax.conf would prevent a trunked call from being created?  Inbound calls work fine if delayreject=yes and there is no trunk=yes defined for the user, but the instant trunk=yes is defined, after AUTHREQ an INVAL is given...
13:59.12tzangermishehu: it's a bug I think
13:59.15tzangerI had that exact same problem
13:59.24tzangerit's an OLD bug too
13:59.29bjohnsonmarkak2: I'm surprised it ever worked .. how is the EXTEN getting to that macro?  You're not feeding it as an arg
13:59.30mishehutzanger: thats what I think too.  and that's what turned out to be my problem yesterday.
13:59.36tzangerahh
13:59.37bjohnsonalso EXTEN won't survive a goto()
13:59.50tzangerbjohnson: it won't?
13:59.56tzangerGoto(Context,${EXTEN},1)
14:00.15mishehutzanger: the instant I commend out delayreject, I could make trunked calls again, with plaintext or rsa auth.
14:00.18bjohnsonthat is feeding it
14:00.24mishehus/commend/comment
14:00.30tzangerbjohnson: you didn't say that was a bad thing
14:00.38bjohnsontzanger: he only has an s exten and trying to use the EXTEN variable
14:00.40tzangerhow do you expect exten to survive otherwise?
14:00.51tzangerwont' work because s *is* the extension
14:00.53bjohnsonthat is exactly my point
14:01.00tzangerbjohnson: ahh
14:01.04tzangerwell now you have some backup.  :-)
14:01.35*** join/#asterisk webman (~adamg@202-44-171-5.nexnet.net.au)
14:01.38shido6either EXTEN or...
14:01.44shido6use ${ARG1}'s
14:01.53*** part/#asterisk lbarth (~lbarth@62.4.65.13)
14:02.22*** join/#asterisk dalabera (~Dalabera@mail2.pmrtechnologies.com)
14:06.27Aze`How use txfax ??
14:08.18Hmmhesaysheh, php has a compiler now, interesting
14:09.07mishehua bytecode compiler.
14:09.08bjohnsonmarkak2: the dial line is ok .. it's the EXTEN that is missing
14:09.10mishehunever used it.
14:09.14*** join/#asterisk IPmonger (~ipmonger@pcp0010577106pcs.coatsv01.pa.comcast.net)
14:09.36bjohnsonmarkak2: look at the superdal macro on the wiki for an example of how to pass the EXTEN to a macro to dial
14:09.49*** join/#asterisk nesys (~nesys@81-174-12-111.f5.ngi.it)
14:10.49malverianI'm setting up a outgoing context for dialing a number and using Festival to read a message to the person who picks up the phone. I'm wondering if there is some way that the file in /var/spool/asterisk/outgoing can somehow define the message that should be spoken?
14:10.51*** join/#asterisk MattB2 (~mattb@pcp01068561pcs.andrsn01.tn.comcast.net)
14:11.05malverianBecause as of right now I just have a Festival('......') hardcoded into the context.
14:11.40*** join/#asterisk jpe (~jpe@www.brooklynbassmint.com)
14:12.00FaithfulHi guys!
14:12.47malverianI'm not sure if I'm being clear.
14:12.55*** join/#asterisk VirTERM (~virterm@shiva.kanatek.com)
14:13.01VirTERMmorning
14:13.20VirTERMres_sms what's that?
14:13.41malverianIn less cryptic terms, I want to have asterisk dial a number and use festival to send a dynamic message.
14:14.49forkqueuemalverian: Is the message 'Congratulations, you have won a prize'?
14:15.17ariel_does anyone here have the latest firmware for the sayson 480i phone that you can let me have?
14:15.59malverianforkqueue: No, it's more along the lines of "Warning, the website is down" :-P
14:16.12malverianforkqueue: I'm replacing voiceshot. And using it in our nagios server.
14:16.12forkqueuemalverian: Heh :)
14:16.36malverianIf something superbad happens, I don't mind if it calls people's cell phones instead of just paging me ;)
14:17.04forkqueuemalverian: That's not a bad idea actually :)
14:17.28*** join/#asterisk mogorman (~mogorman@207.111.174.1)
14:17.38malverianforkqueue: It's a damn good idea.. I think I'm going to have to write my own agi though..
14:17.43malverianFrom the way things look.
14:17.57malverianUnless I want to restart asterisk and edit my extensions.conf every time.
14:18.02malverian(not an option) :-P
14:18.19forkqueuemalverian: Yeah, I think a custom-written AGI would be necessary, but it needn't be the worlds most complicated script or anything..
14:18.46malverianI just need to learn how to write them.. shouldn't be too awfully difficult I imagine.
14:19.00malverianGoing to look at a few of the example ones.
14:19.29forkqueueLet me know how you get on, it's a good enough idea that I might do it myself :)
14:19.43malverianOh, SWEET!
14:20.07malverianIt just pipes it to the program, so you can write it in any language.. perfect.
14:21.00BuckRogersgood morning
14:21.23*** join/#asterisk Slainte (~Slainte@66.55.112.85.ppp.northrock.bm)
14:21.30pifhi, is there a way to test if a SIP phone is available before ringing it?
14:21.41malverianHmm..
14:21.53bkw_<rant>
14:22.06bkw_Ok it pisses me off to have to jump thru hoops to buy a product from a company.
14:22.09bkw_</rant>
14:22.47tzangerbkw_: so don't buy from them, and let them know why you're not buying from them
14:23.10bkw_the problem is a lot of companies do this
14:23.12bkw_its pure bullshit
14:23.26bkw_I don't wanna have to contact them.. go to a distributer
14:23.28*** join/#asterisk Fabe_ (~spamhere@pD95B0BF7.dip0.t-ipconnect.de)
14:23.31SlainteI want a user to be prompted to enter a three digit billing code, if they dial a longdistance number.  Can someone take a peak at http://pastebin.ca/8774  and see if there is anything obivously wrong with my plan?
14:23.34cypromislazy bunny
14:23.41bkw_then find they don't have a website that I can buy from.. nor do they return phone calls
14:24.19*** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net)
14:26.14tzangerahh
14:26.16tzangerget it off, get it off
14:26.52tzangerbkw_: what are you trying to buy?
14:27.56kramget what off?
14:27.57bjohnsonpif: dial will tell you if it is unavaiable
14:28.11kramwere these starving crazed weasels?
14:28.13bjohnsona spider
14:28.14tzangerkram: hahah I was gonna say 'you' but now that's just not right
14:28.16kramah
14:28.34kramtzanger: You're Not Right(TM)
14:28.48tzangerkram: this is true
14:29.06tzangerkram: I emailled greg, hopefully he can find some time in your schedule
14:29.07bjohnsonSlainte: no .. nothing obvious.  Does the noop show the right info?
14:29.17tzangerI'd invite you over but I've not got a hot tub
14:29.27tzangerI have a bathtub I can put hot water in but I don't think it's the same
14:29.35kramtzanger: hopefully, if you don't hear from him in the next couple of days, let me know and i'll try to get it figured out
14:29.40kramtzanger: alas, no, it's not :)
14:29.40SlainteI have not tried it yet, as it is production.  I wanted to make sure that it had a high chance of success before putting it in :)
14:29.41tzangerok
14:29.46Slaintemy test system is down
14:30.01Slaintebjohnson,  you think I did it the correct way?
14:30.27bjohnsonSlainte: put it in as a test exten so you dial the exten to test it
14:30.46bjohnsonlooks ok from the parts I see
14:31.13SlainteThanks,  if it works I will add it to the Wiki for SetAccount because I could find nothing like it there.
14:31.19bkw_tzanger, you wanna know?
14:31.26tzangerbkw_: yeah
14:31.34tzangerdamn it takes forever to copy 1.2G over USB1
14:31.56*** part/#asterisk clive- (~pirch@rndf-146-44-91.telkomadsl.co.za)
14:33.28*** join/#asterisk iq (~iq@70-59-163-239.omah.qwest.net)
14:35.00bjohnsonSlainte: look at the superdial macro in the tips and tricks section
14:35.57Slaintebjohnson, I looked at it but still could not figure out how to get a user prompted request to enter a code, and then set the variable based on what the user entered
14:36.35malverianforkqueue: Looks like the easiest way is just using SET VARIABLE to set a message variable and using that in the dialplan itself.
14:36.44*** join/#asterisk Darwin35 (~Darin@24.3.226.147)
14:36.46malverianforkqueue: Worked like a charm :)
14:37.33shido6Slainte
14:37.34forkqueuemalverian: Care to document it on the wiki?  I'm sure there are plenty of people that might like to have Nagios phone and tell them what the problem is..
14:37.36shido6dont freak out
14:37.40shido6its not difficult
14:38.18shido6we've been using nagio since it was called netsaint
14:38.19shido6:)
14:38.22shido6nagios
14:38.33forkqueueI used to use SNIPS
14:38.38forkqueueBut nagios is loads nicer
14:38.55shido6back when I was at global crossing we used it there, too
14:38.59*** join/#asterisk pluto70 (~god@80.70.179.70)
14:39.15forkqueueHeh, gblx?
14:39.22forkqueueDid they go bankrupt?
14:39.30*** join/#asterisk Lee__ (~lee@ool-44c26ebc.dyn.optonline.net)
14:39.58shido6they filed for protection
14:40.02shido6and laid off a ton of workes
14:40.04shido6workers
14:40.07shido6*grumble*
14:40.22shido6then laid off a ton more
14:40.41shido6now Im a carrier and GX can kiss my black ass
14:40.55shido6at a medium pace
14:42.56malverianforkqueue: I may actually do that.
14:43.55sivanacan anyone here recommend a good spam smtp proxy?
14:43.56Slainteshido, didn't XO buy their NorthAmerican assets for chump change?
14:44.01sivanaanti-spam
14:44.13*** join/#asterisk scanna (~scannachi@81-174-16-211.f5.ngi.it)
14:44.44*** join/#asterisk marno (~marno@213-182-114-20.teleos-web.de)
14:44.52ariel_spamassasin
14:45.00sivanafor windows?
14:45.47sivanaactually.. does that install before the mail server?
14:46.16bjohnsondoesn't MS have one you can buy?
14:46.22sivanaya.. right
14:46.45bjohnsonwe run linux on our mail server and use spamassassin with procmail
14:47.20sivanadon't worry.. I'm teetering on the fence.. OS has its own problems
14:47.52ariel_sivana, for windows server I have been using IpSwitch imail it comes with a good spam blocker
14:48.08sivanaI'm using a NoSpamToday system that's suppose to be based on SpamAssassin, it uses tokens?
14:48.58*** join/#asterisk focks (~craig@nsc66.147.95-93.newsouth.net)
14:49.08ariel_I was reading that CamAV has released a version for Windows. Maybe there is a version of spamassasin for windows out there?
14:50.03sivanaariel_: I'm pretty sure these other folks placed a win32 GUI on the SA engine
14:50.13blitzrageanyone know what that toolkit is so that you can popup screens from the taskbar in Windows? (like the Gmail email notifier program)
14:50.15malverianWhat's the recommended codec to use?
14:50.21focks200/200          10.10.10.60      D          255.255.255.255  5060     Unmonitored
14:50.36focksthat shows a valid registered SIP extenstion no?
14:50.41blitzrageyes
14:50.43ariel_blitzrage, if your using IE msn has a free popup blocker
14:50.51nesysfocks try quality=yes :)
14:51.00forkqueuemalverian: That very much depends on your situation :)
14:51.07eKo1Man, some of these debug messages are just plain weird.
14:51.13focksnesys can you tell me why?
14:51.16blitzragefocks: if you see IP, then prettty much yes. qualify=yes will show you the latency where Unmonitored is
14:51.19malverianforkqueue: Well, my sony ericsson mobile phone is giving me issues on my pbx.
14:51.21eKo1e.g. Oooh, something is weird, backing out
14:51.33focksblitzrage ahh
14:51.36forkqueuemalverian: If you've got the bandwidth, use ulaw or alaw
14:51.38*** join/#asterisk jpe (~jpe@66.114.77.37)
14:51.47malverianforkqueue: This is for analog.
14:52.03blitzrageariel_: not what I mean. You can install a little program and send info to it to make it popup a cool little window in the taskbar. Look at the gmail notifier program for an example of what I mean.
14:52.04malverianforkqueue: When I dial in from my cell phone, it doesn't recognize keypad presses correctly sometimes.
14:52.10malverianBut works fine from a land line.
14:52.19blitzragemalverian: ulaw
14:52.28*** join/#asterisk CoderCR (~creyna@ip68-6-244-85.sd.sd.cox.net)
14:52.29CoderCRhello all
14:52.34focksblitzrage i'm still having this pesky Call Not Allowed issue
14:52.49focksblitzrage just been playing around with *@Home
14:53.04jpe<PROTECTED>
14:53.13webmananyone here know anything about the zaptel kernel modules? I'm trying to 'fix' the wcusb driver...
14:53.35jpeI beleive I have everything set up correctly, and I can see the calls coming in on the console, but get rejected with no authorization
14:53.49jontowwebman; your best bet is to actually ask a question with meat to it so someone will pay attention ;)
14:54.15webmanbasically this line fails:                 if (usb_submit_urb(&p->datawrite[x].urb, GFP_KERNEL))
14:54.24jpeI have the inbound contexts set up correctly for each sixTel and voicepulse-in-01, and the dids set up in amp
14:54.26ariel_for help with  Asterisk@home for the AMP use there is a location here for that. #amportal
14:54.42webmanwith -EINVAL which I assume is a) Invalid transfer type specified (or not supported)
14:54.45malverianblitzrage: Hmm.. is that an option I can set in my zapatta.conf ?
14:54.50jpethanks for the #amportal info
14:54.51dalaberaguys on my cdr reports appears IAX2/gw1@xx.xx.xx.163:21 for my inbound calls, isn't suppose to appear only IAX2/gw1 since it's configure in iax.conf?
14:55.07webmanor less likely, but possible: b) Invalid interrupt interval (0$<=$n$<$256)
14:56.29jpeI don't quite think it is an amp problem as I had it working before before i went to 0.8, it gave the same trouble in 0.6 and I did something simple and stupid to get it working
14:57.03shido6try configureing the conf files manually as amp needs a lot of improvement
14:57.24focksjpe did you run into any access problems with SIP clients?
14:58.40bjohnsoneww .. editing AMP conf files by hand is not for the type of person that typically wants to use AMP
14:58.44jpeno, everything else is working fine with the system
14:58.46*** join/#asterisk anthm (~anthmct@CPE-69-76-83-52.wi.res.rr.com)
14:58.46*** mode/#asterisk [+o anthm] by ChanServ
14:58.57*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released
14:59.36jpeWhen I started with *, I did everything by hand and had a nice little thing going, then I came across aah and have been using it
15:00.26jpeall is working well with zaps and outbound
15:00.45*** join/#asterisk nDuff (~cduffy@64.128.31.220)
15:00.46*** topic/#asterisk by bkw_ -> Asterisk: The Open Source PBX || 1.0.7 Released || ClueCon Dev Conf Aug 3rd - 5th
15:02.03SlainteGotoIf($["${accountcode}" != "???"]?3)                        Is this valid?  Will it require a three digit number?
15:02.10nesysI've a problem with call forward between cisco ccme and *:
15:02.16*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
15:02.17jpethe only other proble that I have not tackled yet is the sipura 2000 not reckognizing *xx stuff (*69 etc) , flash hook transfer to call waiting and things like that, that is next on my list
15:02.21nesyshttp://www.nesys.it/sipwork.txt is ok
15:02.30*** join/#asterisk angler_ (~angler@suid.digium.com)
15:02.32nesyshttp://www.nesys.it/sipdnwork.txt doesn't work :(
15:04.18*** join/#asterisk trig_hm (~jb@home.monkeypr0n.org)
15:04.40*** join/#asterisk ikey (ikey@220.226.28.82)
15:04.46*** join/#asterisk wmoran (~wmoran@pa-plum-cmts1e-68-68-113-64.pittpa.adelphia.net)
15:05.06wmoranThis is probably an old, tired subject, but ...
15:05.11wmoranwho wants to talk about echo problems?
15:05.27Slaintewmoran,  Ok lets talk because I am having issues
15:05.41wmoranWe got most of ours fixed ...
15:05.57Slaintegood, I have not :)
15:06.00wmoranThe biggest problem we have now is the _other_ end hearing echo ...
15:06.07wmoranWe can't seem to do anything about that.
15:06.28SlainteDid you have incoming echo issues?
15:06.29nDuffMy system is trying to dial 7 digits in to an international number. Presumably it's not recognizing them correctly. There's a "international_pstn" section in extensions.conf that specifies "exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})", which looks about right -- but it doesn't appear to be referenced from anywhere else. Should it be? Am I on the right path in debugging this?
15:06.57*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
15:07.19ManxPowernDuff: ${TRUNK} and ${TRUNKMSD} are variables that you have to set in [global]
15:07.20SlainteNDuff,  make sure you have the international_pstn context included in your internal dial plan
15:07.20wmoranThere's a HOWTO on how to properly set your rxgain and txgain ... that helped a LOT
15:07.54wmoranOf course, if you're using PSTN ... that doesn't help if you're 100% VoIP
15:08.27nDuffSlainte, where would that be that I should include it? [default]
15:08.45nDuffs/$/?/
15:08.47SlaintenDuff,  default is fine, for now.  Make sure people dialing in cant then dial back out
15:08.56Slaintewmoran, we have a PRI
15:09.47*** join/#asterisk loko-w- (~rbrown@host1.vocollect.com)
15:10.09Slainteso I should not have cross talk
15:11.00SlainteBut certain places we call always have echo, and other places not.
15:11.10tzangerSlainte: all that PRI means is that you do not GENERATE echo
15:11.30wmoranYeah, if my brother calls the office from his house, the echo on his end is almost unbearable
15:11.34Slainteyes, I know that.  Thats what I was referring by "cross talk"
15:11.40wmoranBut he never gets echo when he calls anywhere else
15:11.44wmoranIt's difficult to track down.
15:12.35malverianI got it set up so if I press 1, I can reschedule the next check of the service for 15 minutes later ;)
15:13.00Slaintemalverian, is that event handling in Nagios?
15:13.25malverianI'm just using the command queue file directly.. there might be a more "proper" way to do it.
15:13.40Slainteyeah, but that will work.
15:13.49*** part/#asterisk CoderCR (~creyna@ip68-6-244-85.sd.sd.cox.net)
15:13.52malverianI know, it -does- work, I just tested it :-P
15:13.54SlainteI have mine setup with SMS back and forth
15:14.00malverianYeah, have had that set up for a while.
15:14.24Slaintewell my daddy can beat up your daddy, and my daddys dead.
15:14.26malverianI'm doing it via email though..
15:14.28*** join/#asterisk Corydon-w (cinnamon@vcchgate.vcch01.springfield.tn.us.vcch.net)
15:14.35malverianSo there is probably a better way.
15:14.56malverianeg.. 3492834@cingular.net (etc)
15:14.59Slainteso you email to SMS< and the SMS emails back?  I have mine setup with direct SMS out, incase the internet or mail gateway goes down
15:15.03malverianWhich is a problem if our mail server goes down ;)
15:15.37malverianSlainte: Hmm.. how hard is that to set up for direct sms? And do you have to do something special with your phone service provider?
15:15.39*** join/#asterisk rephorm (~brian@ip67-95-13-60.z13-95-67.customer.algx.net)
15:16.33Slaintemalverian, very easy,  I bought a special siemens little box, that takes a sim card, has a long antenna, and it sits like a cell phone.  You then plug it into your serial port,
15:16.51malverianHmm.. neat, didn't know such devices existed :)
15:17.04SlainteFirst one I set up like that was in 2000,
15:17.17Slainteback in the Netsaint days
15:17.17malverian$$ ?
15:17.35Slainte150 bucks or so.
15:17.39malverianWow, not bad.
15:17.43Slainteits a phone without any screen or number pad
15:18.04malverianAnd you operate this little jewel through asterisk or some other software?
15:18.24*** join/#asterisk Mimmus (~viggiani@ext.pitagora.it)
15:18.34Slaintenope, through a small app, that listens to the serial port, creates a queue, and then responds.
15:18.38Slaintevery much like an MTA
15:19.13MimmusHi, I'm trying to receive fax by chan_chapi, using latest CVS with all patches but I get a coredump
15:19.25malverianSlainte: qpage?
15:19.33Slainteno,  but very similar
15:19.40SlainteI use qpage as well
15:19.50malverianI use those for my arch pagers.
15:19.53malverian(qpage)
15:19.59malverianI just noticed it had SMS support.
15:22.17ChkDigitIs there a project to add Skype "channels" to asterisk?
15:22.18Wonkai hope not
15:22.29Wonkait would encourage people to use that closed-source stuff
15:22.42*** join/#asterisk jaiger (~jaiger@fire.innovationsw.com)
15:23.02Mimmusany help with Asterisk coredumping?
15:23.51Nuggetdunno, maybe you should ask in #asterisk.
15:24.11Mimmus:-)
15:24.35Nuggetcore dump files are a black art.  I'd help if I could, but I have to settle for being unhelpful.
15:24.36*** join/#asterisk h4mm3r` (~h4mm3r@213-140-17-106.fastres.net)
15:25.04MimmusNugget: thanks, asterisk coredumps when receive a fax
15:25.07Nuggetprobably not a bad idea to not load any modules you don't need, just to minimize the potential for issues.
15:25.18Nuggetstrange
15:25.33MimmusNugget: it dies just after capiAnswerFax
15:25.34Nuggetis that with spandsp or something, or just the mere fact that faxes are talking on a normal channel that does it?
15:25.37Nuggetah
15:25.43Nuggetno clue, sorry
15:25.58MimmusNugget: (I'm using chan_capi)
15:28.05*** join/#asterisk Corydon76-home (three@pcp08665860pcs.500ash01.tn.comcast.net)
15:30.09*** join/#asterisk umb (~java@adsl203-158-089.mclink.it)
15:30.11umbhi
15:30.43umbwow, this channel is much much more popular since I last visited
15:31.45java_frankly I have a lame question
15:32.00*** join/#asterisk bublbobl (~chatzilla@nat-80-64-224-17.e-qual.fr)
15:32.04jontowwhat the hell.. ZapRAS() ? :o
15:32.17java_eheh even worse
15:32.39java_how can I download a sip image for a 7905?
15:32.48bublboblHello all, sorry for my question (unrelated to *) is there a newbiez channel on freenode ? :-$
15:32.56nDuffAnyone have an international [relative to the US] number I can use for testing purposes?
15:33.08java_nDuff: i do
15:33.33*** join/#asterisk MikeJ[Laptop] (~icechat5@65.170.43.34)
15:34.55java_anyhow, lame question=no answer, nothing to argue
15:36.48*** join/#asterisk Mother_ (~mother@93.Red-80-32-127.pooles.rima-tde.net)
15:36.51Mother_greetings
15:36.54*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
15:36.57Mother_professor Falken...
15:37.08java_these poor 7905s keep belinking
15:37.33*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
15:37.49PBXtechcan you send a CLI command from the extension.conf file?
15:38.19Mother_anyone here has successfully configured the RTP ports on a Cisco 7960?
15:38.30Mother_no matter what I do they keep going back to default
15:38.45Mother_be it configured from the phone or via tftp
15:39.04JunK-WPBXtech: extension.conf is a text file, how ya want to send a CLI command?
15:39.05*** join/#asterisk Faithful (~Faithful@202-6-145-116.ip.adam.com.au)
15:39.35PBXtechi just want to destroy a meet-me bridge from the extension.conf  thats my goal
15:40.07bjohnsonPBXtech: system()
15:40.26PBXtechso system(MeetMe kick <confno>)   will work?
15:40.34Corydon-wSo write an app to destroy a meetme bridge
15:40.52bjohnsonPBXtech: no .. system runs a system command
15:41.27bjohnsonPBXtech: you could likely do something like asterisk -rx 'MeetMe kick <confno>'
15:41.41PBXtechyea ok
15:41.51PBXtechthx
15:42.37Mother_so I'm out of luck with this Cisco...
15:42.39Mother_dammit
15:43.10*** join/#asterisk enots (dimka@freelsd.net)
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15:47.14java_anyone have a 7905 sip image to pass me?
15:48.13*** join/#asterisk iCEBrkr (icebrkr@chrome.cyberdyne.org)
15:49.27*** join/#asterisk fugitivo (~ajf@201.255.105.150)
15:49.40*** join/#asterisk jeffik (~jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com)
15:49.59*** join/#asterisk klictel (~klictel@207.107.208.137)
15:50.05klictelhello all
15:50.20*** join/#asterisk dave_mwi (~dave_mwi@harpo.dreamhost.com)
15:51.18*** join/#asterisk _GiGi_ (gigi@jabber.szczecin.pl)
15:51.25*** join/#asterisk gpled (~gpled@firewall.fccfurn.com)
15:51.55Lee__what's the command line command to show what codec a current channel is using?
15:52.48Lee__java_: you have to get one from Cisco with their service contract
15:53.15java_ok, this is the official answer, where's the dirty one? ;)
15:53.39gpledhas anyone seen any docs on using asterisk as a deleayed paging system?
15:53.48Hmmhesaysdelayed paging?
15:55.06*** join/#asterisk phpkid (~phpkid@adsl-068-153-207-210.sip.bct.bellsouth.net)
15:55.09loko-w-does anyone know the truth behind the livevoip screw up
15:55.23Mother_Lee__: sip show channel x
15:55.30Lee__thanks
15:55.31Mother_where x is the call ID
15:55.42Mother_you can get the call IDs from sip show channels
15:56.11java_PSALLOC=early
15:56.35Lee__wait, this is for an IAX device
15:56.44Mother_iax2 show channels
15:58.08Lee__the format field is "unknown"
15:59.10Lee__ah, it only shows the codec for a two-way connection, not just dialtone.
16:00.04Lee__I was talking to a stranger and they asked me if something was wrong with my phone cause the sounds was dropping out, so I'm thinking I should start tweaking codecs
16:01.00robl^there is no codec for dialtone
16:01.08robl^dialtone is generated by the phone itself
16:02.11java_I have a question about the 7905 sip image, can somebody (willing to help) query me?
16:02.42*** join/#asterisk The_Duke (~the_duke@80.92.64.103)
16:03.07MimmusI have a problem with Asterisk coredumping on capiAnswerFax, can somebody help me?
16:03.33The_Dukehello does someone have some experience with the Junghanns Cards configured in NT/Network modus???
16:03.57*** join/#asterisk |nix (~inix@cm240.gamma116.maxonline.com.sg)
16:04.10*** join/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca)
16:04.33The_DukeI'm missing a setting, I cannot remember it. I have ISDN Phones connected to the Junghans Card, as soon as I go off-hook, i get :    == Primary D-Channel on span 3 up for TEI 64
16:04.33The_Duke<PROTECTED>
16:04.56The_Dukehow do I make asterisk wait for digits to be dialled???
16:05.12|nixguys, did CVS head copy changed voicemail extensions from .WAV to .wav49?
16:06.49webmanThe_Duke: change immediate=no in /etc/asterisk/zapata.conf
16:07.14webmanThe_Duke: I mean, for that channel, change it to immediate=no ...
16:07.29*** join/#asterisk sudhir492 (~sudhir@wbar1.wdc2-4-8-141-004.wdc2.dsl-verizon.net)
16:07.34sudhir492Hi all
16:07.42cypromisor put an exten => s,1,digittimeout(5)
16:07.47cypromisin the context where the phones land in
16:07.52webmanooops... capi... have no idea...
16:08.02webmansorry, time for my sleep I guess...
16:08.21*** join/#asterisk neopher (~crazy@mail.techhelpresources.com)
16:08.52ManxPowerNOBODY uses immediate=yes
16:09.03greg_worki have a tdm400p with 3 fxo connected to it .. for some reason, it'll only answer calls on the first port, though I can make calls out from any port.. my zapata.conf is here: http://pastebin.ca/8782   anyone know what would cause this?
16:09.14Mimmuswebman: thank you anyway
16:09.38greg_workconsole shows nothing when a call comes in. i know it's ringing because i plugged an analog phone in
16:09.50ManxPowerimmediate= should  be renamed donotwaitfordigits=
16:10.30greg_workand it's not the physical line, because if i plug any line into port 1, it answers.. but plugged into 2 or 3, nothing
16:11.33*** join/#asterisk JerJer[mobile] (~nonyobizn@RtrHSTF-FC.hstf.interop.net)
16:12.48*** join/#asterisk Zeeek (~Zeeek@Zeeek.sustaining.supporter.pdpc)
16:13.13viLeRHi, I am trying to authenticate to me with Free World DialUp, this is my line sip.conf: register => login:passwd@fwd.pulver.com/1020  but asterisk log says: Failed to authenticate on REGISTER
16:13.22*** join/#asterisk outtolunc (~me@adsl-66-218-53-170.dslextreme.com)
16:13.52Zeeekis your user account 1020?
16:14.34viLeRZeek: no 1020 is my local extension.
16:14.41ZeeekI'm guessing it's not, so replace 1020 with your real one
16:15.13NuggetFWD doesn't care what your local extension is.
16:16.55viLeRFailed to authenticate again.
16:16.59|nixhi all, i've upgraded my copy of asterisk to CVS HEAD 01/04
16:17.34|nixafter upgrading, my voiemail failed to work, i noticed that upon recording, the voicemail now lives with a extension of .wav49 instead of .WAV like in the past
16:17.40|nixwas anything broke in HEAD / something changed?
16:17.42JunK-Win France, for year, do u say dix-neuf-soixante-dix-neuf ou mille neuf-cent soixante-dix-neuf?
16:17.47mgth|nix: its head
16:17.49*** join/#asterisk moy (~kvirc@201.137.229.111)
16:18.11*** part/#asterisk gpled (~gpled@firewall.fccfurn.com)
16:18.37jpegreg_work: I am a real newbie, but if all the ports are on one card, I think you have to name them 1-1, 1-2, 1-3 etc. I have two single port cards in my set up  system and have to name them 1-1 and 2-1 to get them reckognized
16:18.42bkw_|nix, gotta patch voicemail now
16:18.43|nixmgth: thanks for the correction?
16:18.44ManxPower|nix: Anyone that uses Asterisk CVS and is not on the asterisk-cvs mailing list is an idiot.
16:18.59bkw_ManxPower, please stop spewing crap like that.
16:19.06ManxPowerbkw_: that is not crap.
16:19.10bkw_it sure is
16:19.20|nixManxPower: i might be an idiot, but i didn't install this copy
16:19.24ManxPowerHow many changes have been made to CVS-HEAD in the last 40 hours?  50 or so.
16:19.37cypromisso ?
16:19.39bkw_but I know what change did this..and it wasn't in the past 50 hours
16:19.40cypromisit should still work
16:19.41JunK-Wexacly, how can someone knows all that?
16:19.41vaewynanyone that uses CVS and is not on the asterisk-cvs actually has time to develop   ;P
16:19.44cypromisor the changes are nonsense
16:19.57Slaintedoes the rxgain and txgain work for a PRI setup?
16:19.59ManxPowercypromis: So people wine about something broken. A known broken issue.
16:20.13cypromisso they are right
16:20.15bkw_its not broken really... its doing exactly what it should
16:20.16cypromisif something is broken
16:20.17cypromisit is wrong
16:20.18cypromisno ?
16:20.21bkw_just not in the way you would expect it
16:20.24bkw_:P
16:20.33vaewyn"undocumented feature"   :P
16:20.41cypromisas in usual feature ?
16:20.46|nixManxPower: if it so pleases you, i'm still knew in this asterik thingni as I took over it recently, but i'll take your advise and join asterisk-cvs mailing list
16:21.01|nixbkw_: thanks for the help, but can i say that its a known bug?
16:21.11|nixbecause i'm wondering if its my issue, or an issue with asterisk
16:21.17ManxPower|nix: asterisk-cvs mailing list will tell you EVERY change made to CVS.
16:21.29cypromisso ?
16:21.32malverianHas anyone used Linphone for connecting to Asterisk SIP ?
16:21.37cypromisdoes usage of asterisk require to be a c guru ?
16:21.43|Vulture|no
16:21.45ManxPowercypromis: Do you are not suprized when, oh, say, they remove Voicemail2.
16:21.47cypromisyour comment looks like it
16:21.52ManxPoweror change the codec processing order
16:21.54|nixManxPower: i'll take note of THAT, thank you very much
16:22.15ManxPoweror accidently comit a patch that totally breaks Asterisk (see kpflemming last night)
16:22.25JunK-WManxPower: i fully disagree with you, using HEAD doesnt make u an idiot if u dont read -cvs mailing daily.
16:22.28ManxPowerGranted, that happens very seldom, but it does happen.
16:22.31Slaintecypro,, if you can Read, use patch and make  then you should be ok.  You better know how to backup aswell
16:22.35bkw_I don't recall a patch from lastnight that totally broke asterisk
16:22.44zoame neither
16:22.48zoadid he break something ?
16:22.53zoaFLEMMIMG! :P
16:23.12vaewynOk... that's it /// KP duty for him
16:23.19ManxPowerModified Files:
16:23.19ManxPowerframe.h
16:23.19ManxPowerLog Message:
16:23.19ManxPowerfix breakage from slin endianness commit earlier today (sorry :-()
16:23.40zoaah, thats only for some people
16:23.43zoawith sparcs or so
16:23.48bkw_that didn't break asterisk
16:23.53ManxPowernow I don't know if that totally broke everything, but it does seem pretty serious and you would NEVER know about it if you were not on the mailing list.
16:24.10bkw_it didn't break it... it just wasn't quite right.
16:24.19ManxPowerYou would also not know about the bug in ztcfg that was fixed this morning
16:24.22vaewynManxPower: I am glad you have all the time in the world to read email... :}   cause none of us do
16:24.30zoabut he still deserves spanky spanky!
16:24.38ManxPowervaewyn: Dude, it takes me 5 mins per day to read the asterisk-cvs mailing list.
16:24.57ManxPowerUnless you are using a web based e-mail interface, but you are far beyond my help if that is the case.
16:25.19vaewynManxPower: yes... but you also say that we all 'have' to read the -users... and -dev...and.... ... etc...   which... by your ideals would be like 4-5 hours of reading a day
16:25.21ManxPowervaewyn: I don't READ the list, I glance at the comment to see if it might apply to me.
16:25.24*** join/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com)
16:25.32ManxPowervaewyn: Naw, -dev is optional.
16:25.43*** join/#asterisk mutilator (~animenodv@65.111.201.79)
16:25.45*** part/#asterisk NK123 (~p645@cpe-024-163-078-012.nc.res.rr.com)
16:25.45bkw_ManxPower, we can agree to disagree on this.  Running head is fine.
16:25.46vaewynand -users is a @#$@#$stream  :}
16:26.05bkw_-users is hopeless
16:26.06ManxPowerbkw_: I agree that running CVS-HEAD is OK in some situations.
16:26.17ManxPowervaewyn: I guess my procmailrc filters are working well then. 8-)
16:26.19PBXtechh,1,System(asterisk -rx 'meetme kick ${UNI} 1')    why isnt the '  passing?
16:26.22vaewynNow... if someone is to lazy to check the archives... then they need to be bashed upside the head
16:26.32ManxPowerI'm down to about 50-100 messages per day on -users
16:26.39bkw_PBXtech, full path
16:26.49ManxPowervaewyn: the problem is that google doesn't index often enough
16:26.59|nixbkw_: any help on the voicemail issue?
16:27.01PBXtechthat will make the '  pass?
16:27.05vaewynWho needs google... browse the threads
16:27.16ManxPowerbkw_: I think you should be on the asterisk-cvs mailing list if you are using 1.0.x CVS too.
16:27.27|nixi've checked the cvs-head mailing list, only 1 voicemail.c entry
16:27.29vaewynThere are these things called  'Subjects' that generally tell you what the email is about :P
16:28.26ManxPower|nix: check for codec or format changes too
16:28.56|nixManxPower: Thanks. I'll take note of it
16:29.24PBXtechbkw still didnt pass the '
16:29.50ManxPowerUGH!  I just CANNOT remember how to correctly do ARP spoofing.
16:30.15forkqueueManxPower: man ifconfig
16:30.20ManxPower<PROTECTED>
16:30.28pifis there a way to set an ALERT_INFO in a queue ? I try setvar(ALERT_INFO=blah) before Queue(myqueue) to no effect.
16:30.31ManxPowerforkqueue: that should do it, but doesn't seem to
16:31.05mutilatora cashiers check is guaranteed money right?
16:31.15vaewynin the US yes
16:31.32*** join/#asterisk DEEZED (~Scrilla@adsl-065-006-189-182.sip.bct.bellsouth.net)
16:31.38mutilatork cause i got this email from a guy who wants to buy my car, sounds kinda fishy tho
16:33.31ManxPowervaewyn: NOT always
16:33.34*** join/#asterisk gpled (~gpled@firewall.fccfurn.com)
16:33.49ManxPowerit's guarnteed if it's not a FAKE cashiers check.
16:34.00mutilatorheh yea
16:34.16ManxPowerOne of my vendors got ripped off by giving the UPS guy a fake cashiers check for a COD order.
16:34.21ManxPower..er..
16:34.26PBXtech" worked
16:34.27ManxPowerOne of my vendors got ripped off by someone giving the UPS guy a fake cashiers check for a COD order.
16:34.48ManxPowerUPS would not cover their loss, their insurance company would not cover their loss.
16:34.55ManxPowerI think it was about US$5,000
16:35.07mutilatorhttp://pastebin.ca/8784
16:35.14mutilatorthis would be a bit more than 5k
16:35.23DEEZEDhey guys.. I downloaded the asterisk sounds from asterisk.org and im getting an error when it plays it: Apr  4 12:33:41 WARNING[4699]: app_playback.c:90 playback_exec: ast_streamfile failed on IAX2/sixTel@sixTel-3 for im-sorry-unable-to-connect-to-eng.gsm
16:35.36DEEZEDany idea why im getting this?
16:36.09ManxPowermutilator: can you call the bank that issued the check to confirm it.
16:36.39ManxPowerDEEZED: don't put the file extension on Playback or Background
16:36.46mutilatori just got the email this morning, havn't even done anything about it yet
16:37.02DEEZEDlol
16:37.06DEEZEDoops
16:37.09DEEZEDthanks
16:38.20jpemoney orders or cashiers checks are easily forged. I would have him give you the check number and issuing vendor to call and make sure its good before accepting it.
16:38.45jontowwoo, got the new PRI to the voicemail server up :)
16:38.53BuckRogersHey has anyone configure wake up calling , i just recently did and the voice menu works i can set my time check it delete it but it will not execute, ?
16:39.52mutilatoryea,
16:40.00mutilatorthats what i just emailed him jpe
16:40.19mutilatorsucks the only hit on my car in 2 months is fraud
16:40.20mutilatorheh
16:41.12jpemutilator: that is a scam, scam ,scam.
16:41.22*** join/#asterisk cjk (~cjk@80.92.64.103)
16:41.32jpeI just looked at the link, you wont hear back
16:41.37munchausenwallpaper the closest college with ads
16:41.41cjkhi, is there any way to define the callerid for iax2 like for sip?
16:42.02ManxPowercjk: Yes.  The SAME way.
16:42.22cjkManxPower, thats what i did. logically does not seem to work and thats strange
16:43.05cjkManxPower, i have *1 connectiong to *2 i set the caller id on *2. can *1 overwrite this setting
16:43.13ManxPowercjk: then it's not patching
16:43.25shido6Im going to smack someone
16:43.26jpemutilator: how that scam works is this, they send you a cashiers check, you deposit it, if you have the funds to back it, it clears right away. You take the " balance" and sent it off, Western Union or some other untracable route. 3-4 days later when the check comes back bad, you get his for the full amount. A laser printer and MICR toner is all you need to make forges
16:43.30shido6spandsp and r2
16:43.38ManxPowercjk: explain it better to me.
16:43.51mutilatorya
16:43.56mutilatorthats what i figure
16:44.35mutilatorand as i don't think i've ever had a balance more than $100 in my account he's sol anyway
16:44.36Gand_DJI get scams also.. don't worry. Had a couple on my business site.. hate those.
16:44.51cjkManxPower, you should see *2 as the main server. *1 is a server connected to a pabx using bri's. i define the caller_id on the main server but it is somehow not applied
16:45.14mutilatorso.. anyone want a 2002 intrepid ;)
16:45.14ManxPowercjk: You use * as part of your extension
16:45.22mutilatorhas in dash dvd player :O
16:45.29mutilatori know lot of you are in michigan heh
16:45.59cjkManxPower, not really 2 bris in and 2 bris out
16:46.06cjkso its between the telco and the pabx
16:46.21ManxPowercjk: Perhaps you could type the word Asterisk.
16:46.33ManxPowercjk: So where the hell are your extensions?
16:46.34malverianApr  4 12:46:03 NOTICE[6286]: Registration from '<sip:user@some.host.com>;tag=3564100787' failed for 'XXX.XXX.XXX.XXX'
16:46.45*** part/#asterisk gpled (~gpled@firewall.fccfurn.com)
16:46.46malverianI keep getting this error.. and the logs dont' provide any more useful information than that :-/
16:46.54ManxPowermalcolmd: you don't have a [user] entry in sip.conf or the password is wrong.
16:47.08cjkManxPower, the pabx sends everything to the small asterisk gates which does only this exten => _X.,1,Dial(IAX2/dcluxpabx@voipgate/${EXTEN},60)
16:47.19|nixManxPower: i tried upgrading to the latest CVS and its still not working. you mentioned checking codecs, can you give me moreadvise to point me in the correct direction?
16:47.32ManxPowercjk: What callerid are you getting on the far side.
16:47.41cjknothing
16:47.53ManxPower|nix: That was just an EXAMPLE of a significant change in CVS in the past.
16:48.02mutilator;)
16:48.12ManxPowercjk: What device are you using to make the call?
16:48.21malverianManxPower: I have [user] with username=user and secret=thepassword
16:48.40|nixdarn
16:48.44|nixok then
16:48.44|nixthanks
16:48.46ManxPowermalverian: What is the type=
16:49.04cjkManxPower, an pabx system phone
16:49.20*** join/#asterisk _Sam-- (~sam@207.245.79.253)
16:49.32ManxPowercjk: and you have a callerid= set for the channel?
16:49.55FaithfulHow do I get asterisk to autoload zaphfc.ko ?
16:50.12cjkManxPower, no i just do it in the friends part of my iax.conf and sip.conf
16:51.28*** part/#asterisk java_ (~java@adsl203-158-089.mclink.it)
16:51.34ManxPowerWell, if anyone has questions about Asterisk 1.0.x stable I'll be in, oddly enough, #asterisk-stable
16:51.38*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
16:53.13*** join/#asterisk focks (~chatzilla@nsc66.147.95-93.newsouth.net)
16:54.40*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
16:54.57tzafrirFaithful, asterisk does not load kernel modules. What distro do you use?
16:55.00ZeeekJunK-W Mille neuf cent soixante etc
16:55.28tzafrirbasically you need to run 'modprobe zaphfc' sometime in the boot process
16:56.42Gand_DJWhat voip service do you guys use for calling out? I've browsed through a couple companies offerings so far.
16:59.05vaewynnufone... they rock...
16:59.16*** join/#asterisk langals (~icechat5@196.7.14.183)
16:59.28Gand_DJI haven't had a chance to readup on them. Appears the servers are being upgraded still
16:59.42JunK-WZeeek: thx
16:59.46Zeeeknp
16:59.48robl^Nufone isn't accepting new customers at the moment
16:59.56ZeeekI was ont he phone forever so I didn't see the quest
16:59.56langalshi there...I am using asterisk meetme and trying to get better quality from the gsm codec
17:00.08vaewynThey are having growing pains :P
17:00.09jsharpI've used nufone, voipamericas, and voicepulse.
17:00.20ZeeekJunk-W Dix-Neuf Cent cinquante is also correct
17:00.21langalsWhen I use the gsm codec with another conferencing server the quality seems to be quite a bit better
17:00.52Gand_DJRight now I'm kinda eye-ing Broadvoice... since you can call upto 35 countries.
17:00.53JunK-WZeeek: in quebec, we're not saying Dix-Neuf Cent cinquante anymore, only old ppl saying that.
17:00.55langalsDoes anyone have any idea how I could try and improve the quality? Would frame rate have an effect?
17:00.57Gand_DJalso allows asterisk.
17:01.06ZeeekJunk-W I think it's rare here too
17:01.18JunK-Wso u opt for my patch?
17:01.18Zeeekanyway now we're in Deux Mille
17:01.37JunK-Wyes, but if u want to say the birthdate of someone for example.
17:02.03Zeeeksomeone over 5? Ya, mille neuf cent soixante-neuf
17:02.54JunK-Wfine.
17:03.00nestArgah
17:03.29JunK-Wi've remark some problem with other options in the ast_say_date_with_format_fr aren't correct.
17:05.14*** join/#asterisk jwitte (~jwitte_su@firefly.alpha-lab.net)
17:05.56ZeeekI don't use the French stuff
17:06.18Zeeek(sounds)
17:06.31ZeeekIn fact I wasn't using the indications for a long tiùe
17:06.35Zeeektime
17:08.41DEEZEDIm trying to script my IVR. I want to add time between prompts so the user can have time to press what they need. I tried Wait(1) but it seems like the wait command doesn't allow a response. What is the correct command i should use?
17:09.13jakepdevDEEZED - use ResponseTimeout
17:09.27Hmmhesaysor you could put silence in
17:09.33nestArDEEZED: you can do a Backround(silence/1)
17:09.39DEEZEDyeah couldn't find a silence sound
17:09.49nestArthere's a silence directory
17:09.58nestAr/var/lib/asterisk/sounds/silence
17:10.02DEEZEDi put the command and it wasn't there...
17:10.05DEEZEDill check again
17:10.08DEEZEDthanks guys
17:10.13nestArno problem
17:10.14nestAr:D
17:10.32*** join/#asterisk bjohnson (~bjohnson@jecinc.tor.istop.com)
17:11.17DEEZEDsilence: No such file or directory
17:11.28moyhi guys...... im having troubles to receive the digits from a Grandstream Phone when i use Background(), im using INFO
17:11.31DEEZEDodd
17:11.37moydoes any one knows what could be?
17:11.53Hmmhesayschange to rfc2833?
17:12.00vaewynmoy: done an 'Answer' ?  :}
17:12.05vaewynno... info works
17:12.32moyive already tryed rfc2833...... the funny thing is that i use the same phone in other PBX, and it works
17:12.53Zeeekthere is something unusual about GS + codecs + DTMF
17:12.57Zeeekwhat codec?
17:13.05ZeeekiLBC by any chance?
17:13.08moysip debug says that is using Ilbc
17:13.10moyyep
17:13.13vaewynmoy: make sure you have done an Answer  otherwise unpredictable things happen
17:13.14ZeeekI think that was it!
17:13.27*** join/#asterisk imagmo (~imagmo@c-24-20-249-117.hsd1.or.comcast.net)
17:13.32ZeeekI had that problem - I believe it was with iLBC
17:13.57moyyeawyn: you mean doing an Answer() before Background() ???
17:14.07moyZeeek: so what codec do you recommend?
17:14.30vaewynmoy: yep
17:14.34ZeeekDepends but what I see right now is that I'm using RFC at the moment
17:14.44ZeeekI wan't that thrilled with iLBC
17:15.01Zeeek(er... RFC for DTMF obviously)
17:15.23vaewynilbc is great for packet loss problems
17:15.34*** part/#asterisk langals (~icechat5@196.7.14.183)
17:15.55moyok, mmm let me see, i will add the answer and try may be again with rfc... thanks
17:15.59*** join/#asterisk infra (~infra@216-251-177-106.ips.cpinternet.com)
17:15.59moy:)
17:16.36infrahello all; I need help with IAXTEL!!!
17:16.53Zeeekmoy it depends on the GS firmware too
17:17.03blitzrageIAXtel doesn't work right, unless they've fixed it.
17:17.05Zeeekthere was an issue early on
17:17.13blitzrageuse Free World Dialup
17:17.20blitzrageit has IAX connections
17:17.27infraI am limited to GSM codec...can't use FWD
17:17.40blitzrageit only used ulaw?
17:17.44blitzrageuses*
17:17.45infraindeed
17:17.48blitzragehow odd
17:17.56Lee__iaxtel won't register here. i'm using voicepulse.
17:17.58infraso IAXTEL really is down?
17:18.10infrayeah, I can only register for about 20 seconds
17:18.15blitzrageyes, I found it unusable, but that was a few months ago, but I don't imagine its any better...
17:18.44Lee__I think it's popularity outgrew it's capacity
17:18.55infrawhen registered, if I make a call I get: Max retries exceeded to host 69.73.19.178
17:19.10ZeeekLee__ we use FWD to test iax
17:19.24Zeeektoo bad iaxtel hasn't been working for ages
17:19.27Lee__pay the $11 for voicepulse connect. it's working good and you can talk to the PSTN
17:19.42ZeeekI have vp connect with no monthly
17:19.48infradoes it use GSM or iLBC or G.729?
17:20.05Lee__Zeeek: how'd you get that?
17:20.11Zeeekno DID
17:20.15Zeeekjust outgoing
17:20.26Lee__ah. we need the DIDs for the old skoolers  :)
17:20.37Zeeekheh
17:20.44ZeeekI'm using nufone for DID
17:21.02Lee__is that a commercial service?
17:21.19Zeeekdebatable :)
17:21.31Zeeekdepends entirely on the definition of "commercial"
17:21.31vaewynmoy: which firmware you using on that GS phone?
17:21.53vaewynmoy: cause 1.0.5.22 lists  "Fixed we do not use RFC2833 to send DTMF when the incoming SDP contains iLBC and the immediate next "a" line is not the fmtp line for iLBC"
17:21.54moyvaewyn..... let me see
17:22.03Lee__how would you define "commercal"?
17:22.08jakepdevcommercial - usually dealing with commerce - money
17:22.14*** join/#asterisk boch (~as24@200.59.172.98)
17:22.32Zeeekwell, if you mean "available to the general public" they mostly are
17:22.48vaewynmoy: not exactly our issue... but a good hint they were messing around in there
17:22.54vaewyns/our/your
17:23.29Zeeekbut I keep hearing they aren't opening new accounts
17:23.52moyyaewyn, im using the version 1.0.0.7 ......... i haver tryed using Answer, no results yet, im trying to change the codec now
17:23.55Lee__that's what the web page says
17:23.55ZeeekLee__ what about voipjet?
17:24.04mgthvoipjet sucks
17:24.06*** join/#asterisk A-Tuin (~a-tuin@steves.ip.v4.me.uk)
17:24.11ZeeekI've had good luck with them
17:24.30Lee__looked at them too. Eventually we'll become our own origination/termination but I imagine that's hard.
17:24.42*** join/#asterisk scardinal (~supreme@port816.ds1-suoe.adsl.cybercity.dk)
17:24.44Zeeekvoipjet is so cheap I can call my cell in France cheaper than I can from a phone in France!
17:24.56vaewynI am glad JerJer had the guts to stop accepting customers until they can get to handle the influx... that takes big nuts
17:25.20Zeeekinspite of the "bad rep" they sometimes get, those boys rock
17:25.42Lee__I'm looking for a good backup o/t service
17:25.55Zeeeklet's face it, no one gives good cust service of the kind you got...ummmm last century :)
17:26.00vaewynNufone gets my vote cause they are rock stable
17:26.11Zeeekcellphone providers are horrible,
17:26.18vaewyncells are abysmal
17:26.25Zeeekno way to get anything fixed with cell providers
17:26.38ZeeekI refuse to engage in a contract with any of em
17:27.03vaewynThe best cell i can get in these parts is Nextel...  and egads... dealing with them is still >< close to and amputation
17:27.15*** join/#asterisk PTG123 (~PTG123@ip68-106-24-139.ph.ph.cox.net)
17:27.30ZeeekIn this country internet access is so cheap it's impossible to get service for anything
17:27.51Zeeekit's now like $18/mo for 8Meg/256
17:27.59Lee__!! where?
17:28.03ZeeekFrance
17:28.10Lee__word!
17:28.17Zeeekand they just rolled out theis "20meg" (16 actual)
17:28.26DEEZEDwhat is the best IAX service? pay per min allowing multiple connections?
17:28.37DEEZED256 upload isn't that hot
17:28.41Zeeekdepends so much on how you use it
17:28.48vaewynDEEZED: IMO Nufone... but they arn't taking new customers at the moment
17:29.05ZeeekVP connect isn't bad tho
17:29.20mgthvp is expensive
17:29.28Lee__The pay per miniute has a large wow factor if you are selling it to someone else. One DID, unlimited lines  :)
17:29.30Zeeekyes, and voipjet is cheap
17:29.40vaewynVP gives me issues about 2 hours/week
17:29.46Zeeekoh?
17:30.01*** join/#asterisk Gh0sty (~Ghosty@ip-81-11-221-170.dsl.scarlet.be)
17:30.03vaewynI think it is a link between us and not actually them... but still... not acceptable
17:30.11Zeeektrue I don't spend the day calling on any of the services I use, I just rotate them around for testing
17:31.04mgthLee__ who is one did unlimited lines
17:31.11mgthdoes that mean unlimied imcomming calls?
17:31.14ZeeekI hooked up a toll free from nufone->asterisk->outgoing SIP provider and it worked great they tell me
17:31.14Lee__yeah
17:32.11mgthlee___ nufone?
17:32.21Lee__voicepulse
17:32.50DEEZEDi have iax.cc and it lets me do that... but I need access to 1800 numbers and i had one that didn't work with them
17:32.53nestAranyone having trouble building CVS-HEAD? It crashes at app_queue
17:34.34bjohnsonmgth: unlimited is a funny word in voip .. it rarely means unlimited
17:34.46cypromislimited unlimited
17:34.49cypromislike unstable stable
17:34.53DEEZEDlol
17:34.59bjohnsonmgth: also, you should compare to per minute services which oftenn come out to be cheaper
17:35.08jsharpSure, its unlimited.  You can use it for an unlimited time, just as long as you pay 2 cents a minute.
17:35.17Zeeekcypromis what the latest on farfon?
17:35.27moyZeek, yaewyn: still wrong my digit reception in Background, just one last question, do yoy thing that the fact that im running in a chroot enviroment could have something to do? because exactly the same config in my other PBX is doing it well
17:35.28DEEZEDyeah per minute owns.. especially when you can handle multiple calls at once
17:35.33bjohnsonuh oh .. he poked his head out
17:36.25Zeeekmoy no idea
17:36.47Lee__the limit is your bandwidth not how many "phone lines" you have
17:37.04bjohnsonLee__: also depends on the service
17:37.09moyok, many thanks anyway, im going to move more config stuff to see what happens
17:37.13bjohnsonsome limit to one at a time
17:37.38bjohnsonmost limit to 4 to 6 at a time
17:37.43Lee__well that's kind of lame.
17:37.57bjohnsonstart your own .. shown them how it should be done
17:38.09Lee__that's the plan  :)
17:39.19Lee__don't hold your breath
17:39.35Gand_DJHow does one get setup though for offering free unlimited calls to certain places.
17:39.46*** join/#asterisk brianj (~brian@guardian.pmt.org)
17:40.09brianjGents, in the cvs head a lot of caller id was changed for the * manager, on a state ringing why does it show my callerid as my extensions caller id?
17:40.23bjohnsonGand_DJ: let us all know when you find out
17:40.36*** join/#asterisk FirstSword (~Miranda@host6614613596.biz.tor.fcibroadband.com)
17:40.40Gand_DJ:P
17:40.41Gand_DJlol
17:41.17bjohnsonyou offering your pstn for our use or looking to use someone else's for your own use?
17:42.24Gand_DJWhat I was meaning was, vonage & most places offer customers free calling to usa & canada.. but broadvoice offers free calling to 35 countries
17:42.47SlainteGand_DJ,  the business modelling is very complex for setups like that
17:43.22Zeeekyou mean tollfree?
17:43.43SlainteYou have to understand how they make their money
17:43.47Slaintenothing is ever free
17:45.32*** join/#asterisk convey (~chatzilla@63.115.106.66)
17:47.34*** join/#asterisk jtodd (~jtodd@207.230.254.134)
17:47.50PBXtechmy IAX channel is hanging up immediatly after the Background command
17:48.25PTG123GRand: you make your money by counting on people won't ever use enough to bust the amount of money it costs you
17:48.30*** join/#asterisk devel (~devel@wiggum.digitalcoven.com)
17:48.42*** join/#asterisk jtodd (~jtodd@207.230.254.134)
17:48.42PTG123if it costs you 1penny a minute, and you sell $25 plans, then they won't exceed 2500minutes
17:49.05PTG123and if you see someone exceeding it you bust them saying, you are not using this line for general use.. it must be business use so upgrade
17:50.09jaigerPTG123, nothing is free?  next you'll tell me there is no easter bunny!!
17:50.36PBXtechdoes anyone know if the Background command is messed up in todays HEAD?
17:51.18PBXtechResponseTimeout doesnt seem to work
17:53.24bjohnsonGand_DJ: last I checked you had to pay for that service .. not sure if it qualifies for free if you are paying for it
17:53.55conveyHas anyone used Asterisk realtime with MySql 4.1?
17:54.17*** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net)
17:54.23*** join/#asterisk anderiv (~anderiv@207-67-87-34.gen.twtelecom.net)
17:54.57Slainteconvey, I am using 4.0.23
17:55.36Slaintedoes the rxgain and txgain work for a PRI setup?
17:55.45PBXtechyes
17:56.04conveySlainte: I am also using 4.0.23.  I tried 4.1 and had authentication errors due to the password hashing in 4.1
17:56.11*** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net)
17:56.18Slaintephuket, I have no idea why I have not tried it yet
17:56.30Slainteconvey;  why do you need to upgrade :)
17:56.49conveySlainte: my developer ;)
17:57.02Slainteconvey;  why dont you null the passwords before you upgrade
17:57.37jontowheheh, my music on hold has 'crosby, stills, nash & young - carry on' stuck in my bosses head
17:57.38jontow;)
17:57.48harryvvIs there a way to know how far a companies or goverment 1800 number will reach to in a state or region by looking it up online?
17:58.18brianjGents, in the cvs head a lot of caller id was changed for the * manager, on a state ringing why does it show my callerid as my extensions caller id????
17:58.20brianj:P
17:58.20conveySlainte: the problem is in the client.  Mysql is looking for a shashed password and the Asteriskk client does not support it.  I have been trying to turn off the password hashing with very little suck.
17:58.49conveyslainte: sorry bout the type-o's :)
17:58.50jontowconvey; simple solution.. you need more suck.. :)
17:58.53harryvvbetter suck harder :)
17:58.54conveyLOL
17:59.06Slaintedamit you guys beat me too it
17:59.11jontow(very much walked into it..)
17:59.16vaewynbest of suck to you
17:59.32*** join/#asterisk SPoon_TSX (~SPoon_TSX@24.83.96.211)
17:59.51emrahAnyone can please help me with the Astcc application? I'm having a problem with the timeout before it says "The number is not answering".
17:59.55Slainteconvey:  What I did before I had the sql compiled in, is had a wrapper script populate the database from the csv file
17:59.56SPoon_TSXhello everyone, just wondering do you know if I can unregister a SIP peers from the CLI?
18:01.08Slainteconvey:  Change management for a system I set up for a swiss bank.
18:01.20Slaintethats why I needed the workaround
18:01.33*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
18:01.33conveyslainte: maybe I can try something like that.  I am afraid it is in the database authentication at the SQL driver level.
18:01.54Slainteconvey,  do the Perl mod's support 4.1?
18:02.02Slainteyou can use perl.
18:02.32bjohnsonharryvv: just get people here to call it
18:03.08InfraRedhow cheap?
18:03.12InfraRedyou buying a bank ?
18:03.25Slainteemmmm,  something like that yea.
18:03.28bjohnsoncheap as in .. don't want to pay or cheap as in .. low service fees
18:03.34Slaintedont want to pay
18:03.48InfraReduk banks are free
18:03.49bjohnsonit's habitual
18:03.56InfraRedmost of them anywya
18:04.05bjohnsonthey are used to money flow being one way
18:05.04*** join/#asterisk ChulJin (~chuljin@65.211.236.166)
18:05.36ChulJinGood morning Gentlemen!
18:06.57*** join/#asterisk L|NUX (~linux@202.5.145.58)
18:07.46ChulJinis there a current universal favourite among low-price (~$100) hardware SIP phones? My co. is seeking to replace our BT101's with something better (as now fewer of them work than don't)
18:08.41*** join/#asterisk michael_t (~michael_t@c-24-20-234-51.hsd1.or.comcast.net)
18:09.16Gand_DJWe use Lucent phones at my work, but don't know the cost per phone
18:09.40nestArthe sipuras are in that price range, i think
18:10.09nestArhttp://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-38735287552.htm
18:11.13malverianWhat do you guys use to listen to .gsm files?
18:11.29malverianThey sound REALLY crappy on my Alsa sound card when I do tcat some.gsm > /dev/sound/dsp
18:11.44malverianIs there a better way/utility?
18:11.59jontowsox? :)
18:12.01ChulJinright, Sipura is what I'm currently recommending to them.
18:12.26jontowsox file.gsm -t wav file.wav ; audacity file.wav
18:12.48ChulJinwe're already a fan of their ATA's, of which we have several, and with which we've had not a single problem.
18:14.31malverianAh.. okay
18:14.43jontowi wrote a script for it.. 'gsmplay' ;)
18:15.17jontowchecks for existance of the file, automatically changes the name and converts to wav, then runs audacity on it and when i exit audacity, asks if i want to remove the .wav (leaving the .gsm)
18:15.21jontowwill be glad to provide it if you'd like
18:16.32malverianSure.
18:16.45malverianOut of curiosity, why play it in audacity?
18:16.58malverianInstead of a command line one.
18:17.37jontowi wanted the quick potential of removing bits and pieces as necessary
18:17.54jontowit is very easy to change it to use a command line player instead of audacity though.
18:18.44*** join/#asterisk bannerman (~bannerman@c-24-20-88-59.hsd1.wa.comcast.net)
18:18.54malverianI'm trying..
18:19.00jontowhttp://web.slic.com/~jontow/gsmplay.txt
18:19.06malveriansox /var/lib/asterisk/sound/foo.gsm -t wav /dev/dsp
18:19.11malverianAnd I get static.
18:19.34vaewynIf you have sox try  'play /var/lib/asterisk/sound/foo.gsm'
18:19.34_Sam--sox winwave.wav -r 8000 -c 1 linwave.gsm
18:19.56_Sam--sorry
18:19.59bannermanI asked last week, but I lost my notes. ulaw sounds good, but with varying network conditions tends to degrade and get junky. Anyone have a recommendation for the right codec to use with moderately poor QoS?
18:20.18zoajust use quicktime
18:20.20bannermanI don't mind paying for a commercial codec if that's the right way to go.
18:20.35malverianvaewyn: That works :-P
18:20.43vaewynbannerman: The less traffic the better... so smaller codecs do better...  and for really bad packet loss ilbc is the nicest
18:20.50vaewynmalverian: good good :}
18:21.09*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net)
18:21.34bannermanI don't think I have really bad packet loss .. maybe I do. I have a full T1 for a fairly small office. I only have trouble with ulaw on rare occasions.
18:25.15*** join/#asterisk MattH (~matth@noc-wireless.chilitech.net)
18:26.06*** join/#asterisk PCadach (~paul@www.east.telecom.kz)
18:26.13hardwirehey smart people
18:26.21MattHlol
18:26.34hardwireI need a T1 going into my office w/ DID's coming in.
18:26.37MattHhey... anyone here using broadvoice... what's the limit on the number of incoming and outgoing calls?
18:26.46hardwirewhere you can have as many as you want until the t1 is funn
18:26.47hardwireerr
18:26.50hardwireuntil the span is full
18:27.00hardwireof different DID's mapped to your span
18:27.06hardwirewhat exactly should I be asking for?
18:27.58*** join/#asterisk [cc]smart (~smart@gw.ptr-62-65-149-158.customer.ch.netstream.com)
18:29.48*** join/#asterisk jpe (~jpe@66.114.77.37)
18:31.45Slaintehardwire,  how many users?
18:32.02hardwireneed a full span coming in
18:32.09hardwiregoing to only have 8 numbers
18:32.14bjohnsonMattH: read their terms of service
18:32.32bjohnsonhardwire: numbers don't matter .. how many concurrent calls
18:32.38hardwirea full span
18:32.41Slainteso whats your question, if you know you need 8 numbers and a full span
18:32.42bjohnsonso 23
18:32.46hardwireyup
18:32.55bjohnsonso ask for a PRI
18:33.05bjohnsonse what they have to offer
18:33.08*** join/#asterisk DougCoker (~dcoker@adsl-67-37-176-252.dsl.chcgil.ameritech.net)
18:33.19hardwirecool
18:33.34hardwireSlainte: what the terms to use would be :)
18:33.39hardwireif its exactly that.. then I am all set
18:33.44Slainteah,,, yes a PRI
18:33.55bjohnsonhardwire: it's more a question of what they have available
18:34.03hardwireesp here
18:34.05hardwirein alaskatown
18:34.14bjohnsonhardwire: also .. if you want internet access through it you'll have to discuss that with them too
18:34.21hardwireI don't
18:34.36hardwirewe have a 2Mbps sync DSL for the office.
18:34.41hardwirefrom that same company
18:34.44bjohnsonalso ask about installation schedule
18:35.03*** join/#asterisk marno (~marno@213-182-126-48.teleos-web.de)
18:35.11marnohi
18:35.14hardwireI asked about lead time on it
18:35.30DougCokerAnyone successful in making Audiocodes MP-104 work with asterisk ?
18:36.02marnoafter some time my asterisk uses 100% cpu. is there a command to see what is going on?
18:37.03blitzragetop
18:37.06MattHbjohnson: I did read them.. I see nothing limiting it
18:37.38bjohnsonlast I read them they had a clause saying more than one concurrent call would be billed at some posted rate
18:38.24bjohnsonMattH: in any event .. they would know more about their current service offering than we would.  To date that info has been pretty scarce
18:38.31*** part/#asterisk Moc____ (~mochouina@h66-201-214-109.gtconnect.net)
18:38.42Gand_DJbjohnson, I recall that being true if multiple devices are registered at once for 1 number
18:38.50Gand_DJThey are slow for answering questions
18:38.54*** join/#asterisk Cadu20 (~Cadu83@200-215-114-219.fnsce7001.e.brasiltelecom.net.br)
18:38.56Gand_DJI emailed this friday, and it's now monday
18:39.09hardwirebehold the power of a weekend.
18:39.13Gand_DJthey are open 24/7 they say.
18:39.15Cadu20Hi there, anybody could explain me better what the "ignorepat" command does?
18:40.04marnoblitzrage, how can top tell me, what asterisk is doing???
18:40.05nvrsworkanyone have much success running a NAT'd asterisk server behind a linksys WRT54G router connecting to SIP termination services? im getting what I think are sip connection timeouts.
18:40.29Gand_DJnvrswork I did it. seemed ok.
18:40.54Gand_DJI also ran it through dmz just in case. seemed ok also that way
18:41.23QwellCadu20: If the first digit(s) you dial are the same as ignorepat, you'll keep your dialtone
18:41.35QwellCadu20: Ever been somewhere that required "9" to dial out?
18:41.59Gand_DJI had my server linking to FWD
18:42.17marnono idea?
18:42.38Cadu20Qwell, ah ok!... thank you very much.
18:42.58emrahAnyone can help me with the timeout in Astcc? (The number is not answering) I want to incrise the time before I get it
18:43.02Cadu20Qwell, so, there is no really use for it... just to make the user "think" it is in the old system...?
18:43.15QwellCadu20: no, there is plenty of use for it
18:43.29QwellWhen somebody dials 9, they expect a dialtone still
18:43.38hardwireok
18:43.40hardwirethey were cool
18:43.47hardwirePRI will be installed with a few temp DID's for testing
18:43.48Qwell(if you require it for dialing out, that is)
18:43.59hardwirethen POTS lines will be moved over the the PRI when the system is ready to rumble.
18:44.04Qwellhardwire: They're giving you a test PRI?
18:44.28hardwireI wouldn't call it that.. I am buying a PRI..
18:44.31Cadu20Qwell, right.. i got it... thank you very much!
18:44.32Qwelloh
18:44.41hardwireits my money during the test :)
18:45.00hardwireand I will have signed a contract to use it for a certain minimum length of time under the terms of service.
18:45.01hardwireso yhe.
18:45.02hardwireheh
18:45.38bjohnsonMattH: http://voxilla.com/voxstory71-nested-order0-threshold0.html .. in addition to monitoring usage patterns to look for suspect activity, which many providers do, BroadVoice will also charge the end-user 3.9 cents per minute if more than one outbound call is active using the same set of SIP credentials (except in the case of a three-way call).
18:45.56bjohnsonanother limit on "unlimited"
18:46.48Gand_DJSo much for using an * server, and giving people in the house extensions for their softphones
18:47.24bjohnsoncadu20:  there is indeed a use/need for dialing 9 for an outside line
18:48.08vaewynbjohnson: yeah...  screwing your employer over with RSI lawsuits :}
18:48.12bjohnsonGand_DJ: use a per minute service
18:48.21bjohnsonvaewyn: ?
18:48.41vaewyncarpal tunnel from hitting the extra digit all the time
18:48.52Gand_DJbjohnson, I'm going to setup the asterisk@home system, and give family members in the house an extension each.. and install a softphone on each pc.
18:48.59bjohnsonvaewyn: if you use a 9 .. you can make a pattern that doesn't require a timeout on the dial plan = faster execution = same as ending with '#'
18:49.11Gand_DJwas going to use broadvoice because of the 35 country free calling
18:49.16bjohnson(btw .. '#' is also an extra digit
18:49.17bjohnson)
18:49.33bjohnsonGand_DJ: IT ISN"T FREE!!!!
18:49.51nvrsworkGand_DJ, Linking to FWD?
18:49.52vaewynbjohnson: Umm... you can also plan your internal extensions to not use local prefixes and voila... same effect :P
18:50.04nvrsworkoh FreeWorldDialup
18:50.06Gand_DJFine..... I was going to get the $24.95 plan that includes calling to 35 countries for no extra charge
18:50.07nvrsworkhmmm
18:50.28*** join/#asterisk mcukstorm (~mcukstorm@neo.matrix-lan.net)
18:50.40jaigerbjohnson, let hom get the bill and see how free it is
18:50.45nvrsworkI have my server in DMZ as well.,
18:50.56bjohnsonGand_DJ: $24.95 / 0.02 = 1200 minutes of calling if you go with a per minute provider
18:50.58vaewyn'9' prefix is only for those that can't choose extensions correctly
18:51.11mcukstormHi, does anyone know where the wcfxs module has dissapeared to? i just grabbed the cvs and it isnt in there :$
18:51.31bjohnsonvaewyn: I'd love to see that .. a dial plan that doesn't rely on timeout for internal OR external calls
18:51.41Gand_DJ0.02 is probably for USA/Canada only though right? or anywhere on earth?
18:51.54nvrsworkthe problem i keep getting is, when I connect to my SIP termination service, it will register then people can call the number for a couple minutes and it will reach me. then a little later and it will just ring once then give busy signal
18:52.02vaewynbjohnson: Just don't use extensions that overlay your local dialing prefixes... and you are set
18:52.07jsharpOn an IAX call, if the called system doesn't get frames for a certain time, it drops the call, yes?  Does it send back a "BUSY" indication to say that it did?
18:52.12nvrsworksort of like the sip connection is timing out
18:52.15bjohnsonGand_DJ: you will have to do you're own shopping.  I noticed on the livevoip international rates that many countries are available for 0.02
18:52.19Gand_DJMy mom has a friend in australia she likes to talk to, I know a couple people in malaysia & UK
18:52.39bjohnsonvaewyn: it isn't that simple
18:53.07*** join/#asterisk obelix-o (~fabio@200-138-246-242.fnsce7006.dsl.brasiltelecom.net.br)
18:53.08vaewynbjohnson: yes it is... because for any non local prefixes people must dial '1' first... :}
18:53.13bjohnsonvaewyn: give me an example of an internal extension that doesn't overlap an outbound number
18:53.18hardwirebigjohnson :)
18:53.18obelix-ohi guys
18:53.40vaewynbjohnson: 6103  :}
18:53.47obelix-othere are a open g729 codec for asterisk?
18:54.08hardwireno
18:54.20vaewynYes... not legal in the US of A though
18:54.28vaewynnot even for 'testing'
18:54.33vaewynso don';t ask :}
18:54.36QwellIts not "open" then
18:54.44hardwiresure it is..
18:54.46hardwirein Antartica
18:54.47vaewynit is in every other country
18:54.50marnoafter some time my asterisk uses 100% cpu. is there a command to see what is going on?
18:55.00Qwell"open" or "hacked and legal"?
18:55.04Qwellmassive difference
18:55.08hardwiremarno: did you change nicks?
18:55.16bjohnsonvaewyn: what pattern do you use for outgoing local
18:55.19vaewynQwell: it is open.. the code is available... you just can't use it in the US
18:55.29hardwiremarno: you should probably check out strace, and asterisk in debug mode
18:55.35DougCokerAnyone successful in making Audiocodes MP gateways working with asterisk ?
18:55.38obelix-oi have tried to install g729-041103.diff
18:55.42hardwirethere is no one specific command to find out the one thing asterisk is doing to your system
18:55.48obelix-oon l_ipp_ia32_itanium_em64_eval_p_4_1_2_ev05
18:55.52*** join/#asterisk kmest (~kmest@adsl-158-39-198.asm.bellsouth.net)
18:55.55vaewynbjohnson: local prefixes and then 1 prefix for long distance...  0prefix for international...etc...
18:56.07obelix-obut can't run
18:56.11dmabeI just got an IAXy and provisioned it, but I can't get it to make calls or be called.  It gets a fast busy when dialing out. Everyone is busy congested when trying to call it.
18:56.13marnohardwire, change nicks?
18:56.16obelix-othis lib works on i386 ?
18:56.16hardwirenm
18:56.16dmabeany ideas?
18:56.26hardwireI thoguht somebody else just asked that question verbatim a few minutes ago
18:56.35bjohnsonvaewyn: like exten => _NXXXXXX,1, ?
18:56.47Gand_DJQwell: http://www.readytechnology.co.uk/open/g729/
18:56.57Gand_DJif you arn't in usa
18:57.23QwellI wouldn't use it anyways
18:57.25vaewynbjohnson: nope...  local defined prefixes...  ie   208XXXX,1,...   471XXXX,1,...  etc...
18:57.29obelix-oGand_DJ i have tried but can't install this
18:57.33bjohnsonvaewyn: how many do you have listed?
18:57.38marnohardwire, there is nothing on the interface, this seems to be "inside" of the asterisk
18:57.42vaewynbjohnson: 41
18:57.44bjohnsonvaewyn: most people wouldn't bother
18:57.44jontowmarno; "gdb"
18:57.49marnohardwire, that was me, with the same nick
18:58.03jontowthats the one command to tell you exactly what is happening inside asterisk at the moment.
18:58.04vaewynbjohnson: no... but I prefer to make it hard on the admins and easyon the users...   9 is stupid :P
18:58.26hardwiremarno: run asterisk on the console in full debug
18:58.38vaewynbjohnson: and really.. around here locals are only added once a year if that
18:58.42marnojontow, what is gdb?
18:59.03bjohnsonvaewyn: you should point that out when people are asking about it .. most people would not appreciate the extra config involved in not using '9'
18:59.07marnohardwire, it is debuglevel 9 but there is nothing interesting
18:59.10vaewynbjohnson: and they can still dial them 1(area)+local till we add it
18:59.32hardwiremarno: sorry..
18:59.41jontowoh dear.. yeah, good luck
18:59.44hardwireturn on channel debugging then if you think a channel is doing something bad
18:59.47jontowgdb is the GNU Debugger
19:00.18jontowit allows for a specifically intensive point of view over a running process (ANY running process) on a linux/bsd/etc system
19:00.19bjohnsonvaewyn: what do they get if the dial plan doesn't match?
19:00.29jontowhowever.. you also very much have to know how to use it properly to get anything useful from it..
19:00.34vaewynbjohnson: Other (simpler for admin... harder for user) is require 1+area+local for everything :}
19:01.13vaewynbjohnson: "sorry no such extension"  for 4 digits...  "unable to understand the number you suppliied" for 5+
19:01.25bjohnsonvaewyn: someone here was trying to use agi to do a dynamic lookup of local prefixes
19:01.51marnohardwire, there is no active channel
19:02.04hardwirerm -rf /
19:02.05hardwire:)
19:02.10vaewynbjohnson: is doable...  but stupid... that stuff changes so little...
19:02.14hardwireyour machine is pocessed.
19:02.20hardwirepocessed.. I don't think thats a word
19:02.26hardwireposessed.
19:02.38vaewynbjohnson: even in DC area we only add 1 or 2 new prefixes per month
19:02.47*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
19:02.48vaewynand that is slowing down again
19:02.58Qwellhardcoded prefixes and extensions?
19:03.09vaewynhardcoded local prefixes
19:03.10QwellMust be one (pointlessly) massive dialplan
19:03.29jontowhmm
19:03.31vaewynno... it has a point... everyone dials normally... and it works fast
19:03.33hardwirevaewyn: it would be nice to have a list of all prefexes for certain dialing areas
19:03.44hardwireand the dialing area code.. if one even exists.. for that area
19:03.47Gand_DJHas anyone tried using sipura fxo boxes to work on asterisk (instead of using digium cards)
19:03.51Gand_DJI'm looking at http://voipstore.atacomm.com/shops/ViewItem.aspx/27934028032-34702223616.htm
19:04.02hardwireGand_DJ: I have
19:04.10vaewynhardwire: SBC provides a TXT file of that for us...  one of the few things I like about them...  comes with our call rating data
19:04.15Qwellvaewyn: I can 7 digit, or 11 digit dial, with no wait
19:04.25Qwelland no hardcoded prefixes
19:04.47vaewynQwell: yes...  but this is 4digit 7digit and 11digit
19:04.55hardwirevaewyn: we have SBC up here
19:04.56vaewynmuch more difficult :P
19:04.59BuckRogersHardwire, sbc does the same for us
19:05.05hardwirewell crap
19:05.09hardwireI want one
19:05.09vaewynhardwire: talk to your rep... they can give it
19:05.15*** join/#asterisk Jearil (~Jearil@216-224-56-213.client.dsl.net)
19:05.17hardwirewell thats the weird thing
19:05.23Qwellvaewyn: _XXXX ?
19:05.26hardwirein alaska things are very different
19:05.27Gand_DJI want to setup the houseline to be answered by asterisk.. and then someone can either use voip to call out internationally (if they can authenticate for using that option) or just leave a voice msg in the proper voicemail box for the right family member
19:05.33*** join/#asterisk _SMP_ (~SMP@pandora.burned.net)
19:05.42Gand_DJThinking of using asterisk@home setup
19:05.42ChulJinGand_DJ: I'm not prepared to configure it for you (see the wiki :P), but I can tell you I've found that the SPA-3000 works great with *
19:05.43*** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net)
19:05.45hardwireI don't have an SBC rep.. because none of my lines come from SBC.. everything goes through ACS..
19:05.46vaewynQwell: yep...  is why I have to hardcode locals
19:05.48hardwireand they know crap.
19:05.54Qwelleh?
19:05.57SPoon_TSXSorry everyone, does anyone know how can I force to unregister a SIP client from CLI on asterisk?
19:06.07hardwireGand_DJ: thats what I have set up at home
19:06.19hardwireuse the SPA-3000 if you wanna use your standard lines.
19:06.29Qwellhmm, I see...
19:06.30vaewynQwell: You either have to hardcode the local prefixes or use 9 for outside dialling... or wait for timout... that is the options
19:06.32hardwireI have the SPA-3000 answering.. and then redistributing the FXS to the rest of th ehouse
19:06.59_SMP_Hi folks, does anyone know whether there are debian packages for AMP out there?
19:07.03vaewynand I'll take 41 extra dialplan lines over having to dial '9' first anyday :P
19:07.06*** join/#asterisk lliquibop1234 (~lliquibop@dataflo-office-rtr.dataflo.net)
19:07.47Gand_DJhardwire: so if someone calls your home... the phone plugged into the FXS port will ring until asterisk picks up the line?
19:08.09hardwireyou scare me
19:08.25hardwirethats not how it works at all if you set it up right
19:08.31hardwireevery single incoming call goes to asterisk
19:08.36hardwireasterisk decides to ring the fxs
19:08.48Qwellvaewyn: what happens when your area adds a new 603 prefix? (assuming your extensions are 6000-6050, or something)
19:09.29Gand_DJhardwire.. ah so the fxs port would be setup like an extension.. and is someone wants to reach extension xxx, then asterisk rings that line (instead of a softphone that would be setup for another extension)
19:10.59vaewynQwell: They are not... prefixes in our area are assigned as 2XX and 4XX until those run out... and then they are slated for 3XX and 5XX
19:11.33jontowman.. my old house.. we only had a single prefix :P
19:11.42vaewynand plus... not all 4XX and 2XX patterns are valid... remember they can't use them all :P
19:11.43jontowthe entire 40mile^2 area ;)
19:11.52Qwellshit...phone company should have let you 4 digit dial your neighbors then :p
19:11.54jontowactually, bigger than that now that i think about it
19:11.58jontowno kidding
19:12.42ChulJingand: that's right.
19:13.04ChulJingand: if set up as hardwire did.
19:13.05Gand_DJChulJin: sounds better then using a digium card
19:13.37Sedoroxbbiab.. g/f needs help installing something
19:13.37hardwireChulJin: send me an email
19:13.42Gand_DJhow else could it be setup? (guess you could not use fxs port at all)
19:13.42hardwirehardwire at bogomip dot com
19:13.45hardwiresame with you gand
19:13.49ChulJinhardwire: about what?
19:13.56hardwireso I know who to notify when I publish my really simple config
19:14.00emrahPlease, anyone have experience with Astcc?
19:14.05hardwirethe SIPuras are the PITA
19:14.31ChulJinhardwire: I've had no trouble...mine's already configured and working fine.
19:14.39hardwireoh
19:14.45hardwirewell then you help him :)
19:14.58hardwireheh
19:15.14hardwireall non CID calls go directly to voicemail
19:15.22hardwiresimple
19:15.24*** join/#asterisk gtigene (~gnadenx@c-67-184-112-58.hsd1.il.comcast.net)
19:15.31hardwireI want to use PrivacyManager() at some point
19:15.47ChulJinhardwire: mine's a bit different than yours...
19:16.05ChulJinI use the 'spouse-friendly' style setup I found [somewhere]...I think on the wiki
19:16.05gtigeneWhat version of Debian Linux should I use for a new * system?
19:16.07hardwireI would assume so.
19:16.14hardwiregtigene: I use testing
19:16.29ChulJincalls on the fxo ring through to the fxs
19:16.54ChulJin(* never answers the fxo)
19:16.57gtigeneharwire: any particular reason?
19:17.05Gand_DJhardware & ChulJin, you figure the best (reliable) and cheapest fxo device to use for linking * to a houseline is the sipura 1000+,2000+,3000+ cards
19:17.07*** join/#asterisk illek (~Mike@ip68-13-238-168.ok.ok.cox.net)
19:17.20QwellGand_DJ: the 1000 and 2000 aren't FXOs
19:17.22ChulJincalls from * ring through with a different ringtone...
19:17.39Qwellcan an analog phone have a different ringtone? ;/
19:18.04hardwireGand_DJ: what about the X100P?
19:18.08ChulJinqwell: with the sipura spa's, you can set alert_info
19:18.15ChulJingand_dj: 3000 only.
19:18.22QwellChulJin: I've got a tdm
19:18.27Gand_DJok. Isn't the X100P really expensive?
19:18.39hardwiredo I like like pricegrabber.com ?
19:18.42QwellGand_DJ: Its either expensive, or a non-reliable clone
19:18.44jontow$99?
19:18.51jontowyeah, i've got a crappy clone..
19:18.58jontowi paid $40 for 3 of them
19:19.03jontow(notice the giant price gap)
19:19.05ChulJin1000 and 2000 are the best (reliable), but not cheapest, way to burn up 1000 and 2000 when ringing voltage curses through their FXS-only ports :P
19:19.19jontowand i have volume problems like crazy on them all
19:19.27Gand_DJSipura card isn't that expensive compared to X100P I don't think
19:19.33*** join/#asterisk peted20 (~chatzilla@24-113-67-25.wavecable.com)
19:19.34Qwellmeh, just go voip only :p
19:19.38Qwelldrop the fxo idea
19:20.01*** join/#asterisk johnnyb (~johnnyb@sdsl-38-17-139.tulsaconnect.com)
19:20.22johnnybWhat is the "family" parameter of DBput and DBget for?  Is it just a namespace issue?
19:23.23*** join/#asterisk Blissex (~Blissex@82-69-39-138.dsl.in-addr.zen.co.uk)
19:25.54bjohnsonvaewyn: I documented 6 ways to combine local calling and internal extensions at http://www.voip-info.org/tiki-index.php?page=Asterisk+Dialplan+Planning
19:26.04Gand_DJI would drop the fxo, but if someone calls in the houseline, I want to setup * to allow the user to get ahold of the proper family member (provided they have their softphone activated on the pc), ring the FXS port (for general answer), or goto voicemail
19:27.12DougCokerAnyone successful in making Audiocodes gateways work with asterisk ?
19:27.21bjohnsonGand_DJ: sipura aren't cards .. they are standalone ATAs that plug into your LAN
19:27.39Gand_DJyeah
19:27.56Gand_DJsaw them at atacomm or whatever the site was someone mentioned earlier
19:28.01bjohnsonand if you want to tie a telco line into * .. you HAVE to have a fxo
19:28.13Gand_DJthe sipura 3000 has fxo
19:28.16bjohnsonGand_DJ: all sorts of places have the Sipuras
19:28.23bjohnsonyes .. I have 3 SPA 3ks
19:28.42bjohnsonbtw .. the X100P is like $20
19:28.51bjohnsonthe SPA 3k are about $100
19:28.55Gand_DJthe clone / intel voice modem?
19:29.00bjohnsonyes
19:30.20bjohnsonhardwire: I started to set up a privacy manager .. but then saw that getting callerid from telco was going to cost extra $10 / month
19:30.41hardwirebjohnson: hah
19:30.53bjohnsonChulJin: why don't you have the fxo calls go to *?
19:30.57*** join/#asterisk julianjm (~julianjm@250.Red-80-59-67.pooles.rima-tde.net)
19:30.59Qwellbjohnson: Thats one of the reason I dropped Verizon
19:31.11*** join/#asterisk Drel (~drel@dsl254-029-130.sea1.dsl.speakeasy.net)
19:31.57bjohnsonChulJin: I guess you've also answered the question about analog phones having different rings since you do that
19:32.00DEEZEDafter recording a file with the Record function, Is there an automated way to attach the gsm file to an extension?
19:32.08bjohnsonDEEZED: no
19:32.39DEEZEDbut it can be done right? with a macro?
19:32.57bjohnsononly if you always want to do exactly the same thing with each
19:33.00bjohnsonmost people do not
19:33.37bjohnsonmost people find doing by hand much faster and flexible than writing a macro that will only do what they want SOME of the time
19:34.02DEEZEDmy dad has a Virtual PBX with a company... he types in a random extension then presses a *7 and it lets him record
19:34.15DEEZEDthen when you call that extension it plays the file back
19:34.46bjohnsonnot very useful .. is it
19:35.22DEEZEDwell if i could do that i would be farther along then i am now
19:35.42bjohnsonfarther along to what?  recording ramdom sound files?
19:36.25bjohnsonprovide us with an example purpose for such a thing
19:36.54vaewynit's like a phone post-it-note... cool idea... but man... I can't see much use for it :P
19:37.01bjohnsonalso .. wouldn't he end up potentially overwriting valid extensions with those "random" extensions to play back sound files?
19:37.19bjohnsonvaewyn: he could use voicemail for that
19:37.41DEEZEDok. My dad is a realtor. He has a 1800 number and he creates a pin number for each house that he is selling.. When his client calls in, they can type in the pin and hear about that house
19:37.56vaewynbjohnson: but voicemail requires authnetication... this is nothing but placeholders
19:38.13DEEZEDmy dad can record his pins by typing in the new pin, then a command which will ask for a password, and then allow him to record the pin
19:38.25DEEZEDthus creating a new extension that plays the sound file he recorded
19:38.38bjohnsonso make an exten to record a sound file .. then make a bash script or something to edit the extensions.conf file to make an exten to that
19:38.52vaewynfairly easy to do... but I would turn it around and make him dial *7 first... and then record
19:38.59bjohnsondon't forget to check that you're not overwriting another extension
19:39.18bjohnsonand he'll need a way to delete them too I guess
19:39.25*** join/#asterisk DannyF (~dannyf@h27n3c1o848.bredband.skanova.com)
19:39.26DEEZEDoh ok... so this would be more of a linux feature then an asterisk feature
19:39.28vaewynbjohnson: or just make them all in one range and never touch the extensions.conf...  use an AGI to drive it
19:39.32DEEZEDto modify the extensions file
19:39.41bjohnsonDEEZED: combination of both
19:39.47DEEZEDic..
19:40.02bjohnsonDEEZED: or agi which allows whatever programming language you want
19:40.10vaewynhmm.. wait... don't even need a program...  js
19:40.15DannyFevening folks
19:40.22QwellYou could do all that in the dialplan...
19:40.39QwellYou can force Record to give a specific filename, can't you?
19:40.57vaewynYep... trivial
19:41.07QwellSo, no agi or anything needed really
19:41.21bjohnsonthe recording isn;t the problem .. it's the making of a new extension to play it back that needs something that can edit text files and reload the dialplan
19:41.27DEEZEDyeah.. but the idea is to be able to create extensions on the fly
19:41.33Qwellno dialplan reload needed
19:41.53vaewynhahaha... 2 liner :}
19:41.54vaewynexten => XXXXX,1,Play(${EXTEN})
19:41.54vaewynexten => *7XXXXX,1,Record(${EXTEN})
19:41.55QwellWhen an extension(pin) is dialed, check the existance of ${EXTEN}.wav
19:42.07Qwelland yeah, if it exists...do that
19:42.10vaewynand a 'i' to catch wrong ones
19:42.28DrelDoes the 'Asterisk Community' recommend Polycom IP 500 phones in general?  There seems to be some mixed opinions on voip-info.org, so I thought I'd run this by #asterisk. :-)  The phones would be used in a small-office setting, but there'd be very limited tolerance for misbehavior/need for resets/call completion failure.
19:42.40bjohnsonahh .. just play it and if it doesn't exist .. you get nada
19:42.42vaewynok... so 3 lines
19:42.44vaewynexten => XXXXX,1,Play(${EXTEN})
19:42.45vaewynexten => *7XXXXX,1,Record(${EXTEN})
19:42.45vaewynexten => i,1,Play(no-such-home)
19:42.53DEEZEDwow thanks
19:42.54Qwellvaewyn: yeah, something like that
19:42.54bjohnsonwhat about overwriting existing files?
19:42.55vaewynDrel: They rock
19:43.02Qwellbjohnson: fuck em :p
19:43.03DEEZEDthats going to help me
19:43.34bjohnsonmake sure he doesn't type a pin for a file he wants to keep
19:43.36Drelvaewyn: Are they at the "fire and forget" stage, where you can set them up, plug them in, and unplug them a year later for an office move, without doing anything but making and receiving calls in between?
19:44.17vaewynDrel: pretty much
19:45.18vaewynDrel: we run a couple call centers with them and they are drop and forget
19:45.40*** join/#asterisk _chad (~Chad@c-24-6-142-55.hsd1.ca.comcast.net)
19:45.59shido6im going to smack Mr spandsp
19:46.00_chadanyone know any good PSTN termination in california?  pref bay area?
19:46.13_chadand on the cheap side
19:46.36Drelvaewyn: Do you mind my asking what the general call center setup is?  These would be used on a 10/100 Mbps ethernet network, all connected to a 10/100 switch.
19:46.50DrelDHCP assigned via a Netgear firewall.
19:47.01shido6how many simultaneous calls, Drel?
19:47.16vaewynDrel: 10/100 Cisco... 35 phones per
19:47.52*** join/#asterisk ikey (ikey@220.226.16.105)
19:48.20Drelvaewyn: Do you use static ip allocation or dhcp?
19:48.40emrahPlease, anyone can help me with Asterisk Calling Card application?
19:49.06vaewynDrel: ultra long lease dhcp
19:49.07*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
19:50.06*** join/#asterisk bah (048830696@AC9AEC59.ipt.aol.com)
19:50.15PBXtechif i only get audio in 1 side of the convo, is that a firewall prob? only happens to 1 carrier
19:50.32Drelshido6: per phone or on the LAN?  For the former, two, I guess, with one call on hold, for the latter, maybe 8-10 max?
19:51.55emrahWhy no one answer?
19:52.43harryvvemray check your msg
19:52.49Drelemrah: Why not ask your specific question instead?
19:53.36*** part/#asterisk Gand_DJ (~gandalf@ptr-207-54-104-24.ptr.terago.ca)
19:53.56*** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com)
19:54.22emrahI've asked it before, sorry
19:54.27emrahMy question is:
19:54.49emrahHow is it possible to incrise the time before the timeout comes with Astcc, when we hear "The number is not ansering"
19:54.54harryvvemrah did you get my message?
19:55.13malverianHow can you give on hold music while asterisk is trying to contact a channel?
19:55.15emrahharryvv: nop :S I will search it
19:55.57malverianEg, while the line is ringing, or is this not possible?
19:56.45emrahYES
19:56.57emrahsorry for the capslock
19:56.58hardwire:)
19:57.09_asrany polycom ip500 users in the house?
19:57.11hardwiremaking a very old school *@home install :)
19:57.18*** join/#asterisk MatsK (~NNSCRIPT@107.80-202-57.nextgentel.com)
19:57.25Drel_asr: You must be my echo ;)
19:57.28bjohnsonmalverian: after * has answered the call .. you can use 'm' for the dial command
19:57.40harryvvhardwire thats cool
19:57.53hardwireit owuld be nice .. you know ..
19:57.54emrahI'm lost with this all messages. Anyone have the anser of my quesiton
19:57.58hardwiretoo bad I can't uinterrupt
19:58.12emrahthe answer of my question*
19:58.15_chadman my latency makes voip useless from here.. ugh
19:58.21_chadanyone know of good termination in california?
19:58.23hardwire_chad: where are you?
19:58.28_chadhardwire, bay area
19:58.32bjohnsonI "think" that is possible .. have you looked at the examples using CONSOLE=
19:58.35hardwirewhats the latency
19:58.40hardwireI have do deal with 600+ ms
19:58.43*** join/#asterisk truescot (~na@a62-216-27-177.adsl.cistron.nl)
19:58.47_chadhardwire, jesus, how many hops?
19:58.52hardwireone
19:58.52bjohnson_chad: some voip providers have west coast servers
19:59.13bjohnsonI was told that was why my voipjet lagged excessively .. they're west coast
19:59.27bjohnsonhardwire: sat?
19:59.29_chadhardwire, I have 15 hops to my colo (about 80ms average), and then another 13 from the colo to nufone (60ms average)
19:59.30hardwireyeh
19:59.33malverianbjohnson: Sample dialplan?
19:59.34_chadbj, awesome i'll take a look
19:59.54hardwire_chad: are you doing nufone to nufone calling?
19:59.57_chadhardwire, how do you accomidate that?  the jitter buffer?  literally every call i make/rcv *crunches* out after 5 minutes
20:00.03emrahAnyone please can anser my question?
20:00.06_chadhardwire, nufone to pstn
20:00.14hardwireits weird you are having problems.
20:00.16_chadhardwire, using them to dial out and rcv calls
20:00.31bjohnsonmalverian: exten=>s,1,Answer()
20:00.36hardwirethats a little further away
20:00.36_chadhardwire, I have not been able to complete one real call for more than a minute since i got in with them
20:00.48_chadhardwire, wheres your * box located?
20:00.54hardwirein Anchorage.
20:00.59bjohnsonexten=>s,2,dial(sip/1000,60,m)
20:01.05malverianAh... ,m
20:01.10malverianAutomatically does on hold music?
20:01.19_chadhardwire, the only thing I can think of is that I'm connecting via this crappy cable modem to my colo in texas, which is then going to them in michigan
20:01.26bjohnsonmalverian: I think so
20:01.30_chadhardwire, how do you handle that amount of latency on your box?
20:01.38hardwire_chad: yeh.. crappy ass routing could be an issue :)
20:01.48emrahAnyone have an idea for an interesting provider for pstn calls? Voipjet have bad routing to Switzerland mobile
20:01.53Dreldoes anyone here use wireless bridges -> ip phones on desktops in combination with 802.11g wireless lan?
20:02.03jsharplong latency isn't going to kill a link.  Massive changes in latency will though.
20:02.13_chadbj, you don't know the switch voipjet uses so i can traceroute it do you?
20:02.13hardwireI have low jitter on our VSAT
20:02.15emrahVery big latency too
20:02.22hardwirebut over starband I have a shrinking jitter buffer set up
20:02.28bjohnsonemrah: if you are in Switzerland .. try looking for a Swiss based voip provider
20:02.29hardwirebut really.. you shouldn't need anything drastick
20:02.33_chadhmm
20:02.43hardwire_chad: why all the hoops and ladders?
20:02.50_chadany ideas for me, I did enable the jitter correction.. what else might i take a look at
20:02.53emrahbjohnson: Switerzland based VoIP company are very expensive
20:03.00_chadhardwire, how do you mean? :)
20:03.06hardwire_chad: you are in the bay area
20:03.07bjohnson_chad: 216.118.117.46
20:03.10_chadhard, yah
20:03.14_chadbj, thanks
20:03.17hardwireand you are doing somethign in texas too
20:03.23hardwirewhich then goes to nufone
20:03.30_chadi'm in cali, my colo running * is in texas, then to nufone yeah
20:03.52hardwireok
20:03.58bjohnson_chad: I get 45ms but it goes > 2000 multiple times a day when my other voip providers do not
20:03.59hardwirewhy are you doing that?
20:04.36bjohnson_chad: surely you have a local isp
20:04.52_chadhardwire, originally just ended up that way.. we have an ev1server box out there for our web stuff.. stuck * on there when I started geeking with it... ended up going w/ nufone because of the rates and the reccomendations i read on the mailing list
20:05.07hardwireok
20:05.14_chadbj, yeah i connect locally via a comcast cable modem, then to my * box in texas, then nufone
20:05.20bjohnsonahh .. move * to a more sane server
20:05.32hardwire_chad: or have a (*) box at your house
20:05.37hardwireor a sip proxy that will do what you want
20:05.38bjohnsonexactly
20:05.49hardwirebut
20:05.59hardwireif you want all incoming to go to your (*) box in texas
20:06.02hardwirethat could be an issue
20:06.14hardwireI bet sip "REDIRECT" can help you a lot
20:06.20_chadHMM
20:06.23hardwireHMM!
20:06.31bjohnsonHMS
20:06.33hardwire_chad: what about you calling non nufoners
20:06.35hardwireoff of your colo
20:06.37_chadthat essentially snaps the call free of my * box and shoots it right to my sip phone yah?
20:06.44hardwiresay.. have somebody else register w/ your account and call them
20:06.48hardwireand see if its just as shitty
20:06.57hardwirehow well do echo tests turn out
20:06.58bjohnson_chad: no .. run your own * server
20:07.00_chadhardwire, i've only had calls w/ non nufoners
20:07.08*** join/#asterisk pfn (500@netblock-66-245-252-239.dslextreme.com)
20:07.15_chadbj, ahh it redirects from the texas colo * box to another local * box then?
20:07.25hardwireredirects the session
20:07.33hardwireI dunno how the auth works on things like that however
20:07.44hardwirebut I would run my own (*) server
20:07.50hardwireat your house
20:07.55_chadhttp://www.pastebin.com/267115
20:08.03_chadlol didnt realize that pastebin was only for php code
20:08.06_chadbut you get the idea
20:08.17hardwireits not
20:08.19_chadthe first traceroute is to nufone from my local machine, the 2nd is to voipjet
20:08.45hardwireit might have something to do with your box rebelling against the name you gave it.
20:08.51hardwirethats my first hunch
20:08.55hardwiregost in the machine syndrome.
20:08.57hardwireghost :)
20:09.01_chadhahahaha
20:09.04_chaddr frankendoodle?
20:09.05_chadlol
20:09.16hardwirethen again I am on a machine I called bastard
20:09.24hardwireour routers name is fauker
20:09.44hardwireall named after planes :)
20:09.47hardwirebut we chose the bad ones.
20:09.53_chadlol
20:10.06_chadgrabbing a traceroute to my server
20:10.23hardwireI don't think hops is your issue
20:10.23hardwireheh
20:10.41hardwiremm
20:10.42hardwirebeer
20:10.44_chadhttp://www.pastebin.com/267116
20:10.48_chadno, you don't think so?
20:11.02_chadthat last paste is from my local machine to my colo, and then from my colo to nufone
20:11.17_chadhardwire, whats your best guess?
20:11.25hardwireGITM
20:11.39hardwiretraceroute from (*) to 209.193.36.94
20:11.43hardwireand find out your latency
20:11.52hardwirethats me in anchorage
20:12.29hardwireor you aren't paying enough so they just enjoy plugging/unplugging you at random
20:12.30_chadhttp://www.pastebin.com/267118
20:12.43_chadlol probably both
20:12.56hardwireI would take your phone
20:13.00hardwireand connect it directly to nufone
20:13.02hardwireand try to call
20:13.02PBXtechis there some command that will play a random .gsm file ?
20:13.22_chadhardwire good idea, let me give it a go
20:13.34bjohnson_chad: forget the Texas * server .. go directly from voip provider to your house
20:13.59_chadbj, what about inbound calls though?
20:14.12_chadwould still need to go through the great state of texas :D
20:14.14bjohnson_chad: where do you need it to ring?
20:14.27_chadhome office, but it has to hit the pbx first and whatnot
20:14.35bjohnson_chad: put a * server as close to that location as possible
20:14.38_chadmenu's prompts, etc
20:14.46_chadbj, better to have it closer to me than the termination?
20:14.47bjohnsonwhy does it need to hit the texas pbx at all?
20:15.13bjohnson_chad: wouldn't matter if it were a straight network path .. but it isn't
20:15.25bjohnson_chad: minimize the ms
20:15.37bjohnson_chad: run the ivr from home if low load
20:16.18_chadmaybe i have a fundamental flaw with my thinking... i thought it had to go from the texas pstn to my texas colo "welcome to such and such company, press 1 for sales, 2 for support, 3 for burgers", they hit "1", then it gets routed from texas through my cable modem in cali and my sip phone here rings
20:16.53bjohnsonyou have a texas pstn or a texas voip did?
20:18.31truescotcan anyone tell me what wireless phones are best with asterisk?
20:18.33harryvvafter watching supersizeme I dont like them as much now :)
20:18.53_chadbj :)
20:19.04jsharp_chad:  You can get a closer voip provider that can probably get you texas DIDs.
20:19.09bjohnson_chad: try running a traceroute directly to nufone
20:19.21_chadbjohnson, from my local machine right?
20:19.25Blissex_chad: there may be indeed a fundamental flaw with your thinking...
20:19.25bjohnsonyes
20:19.42*** join/#asterisk ACiDV (~joel@iteckGW.infoteck.qc.ca)
20:19.44_chadblissex, lol hopefully you guys can help me get this axe out of my forehead :)
20:19.50bjohnsonyou have 76 ms from home to Texas .. cut that down or even route around it if possible
20:19.58_chadbj, http://www.pastebin.com/267115
20:20.11_chadfirst set is to nufone from local, 2nd is voipjet
20:20.28terrapeni wish nufone sold texas DIDs
20:20.30ACiDVHmmm, weird IAX problem... one way audio.. 2 servers linked in IAX, if I have a sip phone on each server calling other, I have one way sound, if both phone are on the same server, all work ok
20:20.34jpehello all, Im still here trying to get dids working with aah 0.8. been at it all day to no avail
20:20.34terrapenIAX.cc was a total wash for us
20:20.35Blissex_chad: in theory, hopefully the caller and your home phone get connected directly, if REINVITE is possible
20:21.13bjohnsonif you truely have a Texas pstn connection .. then you will want to have a texas * .. and route calls from that to home .. but you may want to route outgoing from hoem directly to voip provider and incoming from voip provider should go to wherever is faster for end destination of most calls (also depends on call volume)
20:21.50_chadbj, hmm
20:22.40Blissex_chad: the problem with your sewtup is that PSTN does not do REINVITE... :_0
20:23.09_chadbj, my pstn termination service is in michigan (via a 1800#), if I understand you correctly I should ditch them, setup a pstn closer to my colo in texas... or ditch my colo in texas and setup a local machine or a closer machine running * and keep the pstn in michigan?
20:23.31Blissex_chad: depends on what you want to do!
20:23.34_chad:D
20:23.50_chadblissex, low volume office use (2-5 lines max)
20:23.59_chadblissex, without my phones crapping out :(
20:24.00bjohnson_chad: I thought you said you had a Texas pstn
20:24.02Blissex_chad: if you want your calls to end up on your SIP phone at home, put Asterisk and PSTN termination at home.
20:24.21_chadbjohnson, * box is at a server in texas, i use nufone in michigan for the termination
20:24.32Blissex_chad: that's a mad mad setup...
20:24.40_chadblissex :D hahaha :D
20:24.43*** join/#asterisk PBXtech (~nik@67.107.241.3.ptr.us.xo.net)
20:24.52_chadthis is my first attempt, at least i'm learning :D
20:24.53_chadlol
20:25.02bjohnson_chad: btw .. your home isp is killing your home use .. looks like you either go through AT&T or Verizon and both are adding about 50ms to your routing
20:25.07PBXtechhow does the Random command work?
20:25.17Blissex_chad: and if PSTN terminates at Nufone, then Asterisk in TX should REINVITE Nufone to your SIP phone at home.
20:25.43_chadbjohnson, yeah this is through comcast cable modem.. i've tried traceroutes on other dsl providers and gotten about the same unfortunately
20:26.20bjohnson_chad: you're going to have trouble with voip to/from your house until your get faster traceroutes
20:26.23_chadblissex, reinvite would essentially take the texas * out of the picture and connect the call directly from nufone to me correcT?
20:26.37bjohnson_chad: it looks like the bottleneck is somewhere around LA
20:26.40Blissex_chad: yes, ideally, if Nufone and your home SIP phone are compatible.
20:26.51*** join/#asterisk khaled (~khaled@CPE000625627f09-CM001225d88b2e.cpe.net.cable.rogers.com)
20:26.56*** part/#asterisk khaled (~khaled@CPE000625627f09-CM001225d88b2e.cpe.net.cable.rogers.com)
20:27.05Blissex_chad: but if the problem if your home line as bjohnson suspects, you can't do anything.
20:27.25_chadokay so a reinvite if the problem is here would be like polishing the brass on the titanic yah?
20:27.26bjohnsonwell .. he can save 50ms by not routing through texas first
20:27.31Blissex_chad: do you have the same problem with outgoing calls as with incoming ones?
20:27.37_chadyep
20:27.40bjohnsonbut that won't be enough .. he still has 80ms +
20:27.55tzangerwhat's wrong with 80ms latency
20:27.58mutilator80ms is fine..
20:28.08_chadblissex, but both incoming and outgoing are routed through my * and then to nufone
20:28.19_chadtz/multilater, I have about 30 hops of it
20:28.23Blissex_chad: as in, routed or resolved?
20:28.31mutilatoruhm..
20:28.38mutilator30x80ms?
20:28.48Blissex_chad: routed as in the data travels to TX and then to to Nufone?
20:28.50tzanger2400ms is a bit much yes
20:28.53mutilatoryou live in somolia with a satellite phone?
20:28.55_chadblissex, its my understandig its routed yeah
20:28.57bjohnsonsorry .. wrong numbers .. he can end up with 78 ms if he drops the connection to texas which makes the trip to nufone add up to 143ms
20:29.03_chadmultilator hahaha
20:29.05tzanger143ms isn't much
20:29.09tzangerit's noticeable but acceptable IMO
20:29.18tzangeryou're gonna have around that with a jitter buffer anyway
20:29.23mutilatorif i get >100ms over wireless it breaks pretty bad
20:29.25bjohnsonlook at his pastebins
20:29.28mutilator80ms is tolerable
20:29.30tzangeruh
20:29.33tzangerlatency doesn't cause breakups
20:29.35tzangerjitter does
20:29.44_chadhmm
20:29.46mutilatorhence i say wireless
20:29.56mutilatorwhere both latency and jitter are affected
20:30.08*** join/#asterisk BoRiS (boris@S01060040ca1e5b54.wp.shawcable.net)
20:30.22*** join/#asterisk clive- (~pirch@rrba-146-99-95.telkomadsl.co.za)
20:30.35Blissex_chad: why don't you try with some other provider? You might find one much nearer to you in network terms...
20:30.53_chadblissex, in regards to the colo or the termination (or both? ) :)
20:30.55tzangermutilator: but you're giving latency numbers
20:31.01Blissex_chad: and then make sure you can do REINVITE, so you dont have the leg to TX.
20:31.01mutilatoryes
20:31.02tzangerwhich do NOT affect breaking
20:31.21Blissex_chad: well, _ideally_ both.
20:31.30mutilatorfor ME it's a very good estimate of breaking
20:31.32DrelWe currently have two analog telephone lines.  I am putting together an Asterisk box, and looking at the Digium Wildcard TDM400P to bring those two analog phone lines into the Asterisk system (using 2 FXO modules).  We also have 4 Polycom analog telephones that are currently connected to the analog phone lines.  How many FXS modules would I need to support these four telephones?  Would one...
20:31.34Drel...be sufficient, or would I need four?  Could I disconnect our internal telephone wiring from the PSTN, connect the incoming lines to the FXO modules, and run a cord from the FXS module to the internal telephone wiring with all four phones plugged in?  Sorry for the lengthy question!
20:31.40mutilatori have 100 or so customers on our wireless network across northern michigan and thats what i've come to find
20:32.05tzangerDrel: you need an FXS port for each analogue phone, just like you need an FXO for each POTS line
20:32.19Blissex_chad: if you have setup things so that REINVITE works, then it is more important for the provider to be ''near'' your home, because all your Asterisk instance then is just tell Nufone to connect to your home phone.
20:32.30nestAranyone build CVS-HEAD today?
20:32.45mutilatorthe latency on wifi will determine a lot about your 'jitter' atleast for me it does
20:32.49_chadblissex, I see, and then for outbound calling I would just connect to nufone directly?
20:32.53harryvvmutilator how are you doing
20:32.56tzangermutilator: odd but ok, I have not specifically tested it
20:32.57Dreltzanger: Darn.  That sounds like a wiring nightmare!  I guess ATAs are in order instead of FXS modules.
20:33.00nvrsworktzanger, if you connected 2 phones to one FXS module, would it blow?
20:33.03*** join/#asterisk iceyp (~icepick@202.150.105.150)
20:33.04nvrsworkblow the card
20:33.05Blissex_chad: if you want, or you get it handled by Asterisk too.
20:33.09harryvvtzanger by any chance is that releated to udp checksum error? I had a string of them one and causes a 15 second delay in a connecting into the states during a conversation.
20:33.12tzangernvrswork: no, but they will have the same extension
20:33.16mutilatorharryvv.. fine and you?
20:33.20_chadblissex, and it could reinvite in the same way and connect me directly to nufone?
20:33.21tzangerharryvv: is what related to that
20:33.26Qwellnvrswork: it also depends on the REN value, if it can even do that
20:33.32harryvvjitter and breakup
20:33.32tzangerDrel: 6 lines is a wiring nightmare?
20:33.34Blissex_chad: again, if Asterisk handles only call setup, and not data transmission, where ti is does not matter a lot.
20:33.35nvrsworktzanger, how many phones can one FXS support in parallel
20:33.42tzangernvrswork: depends on the REN of each phone
20:33.56iceyphi guys, could someone on fwd, please test my connection on 265744 , i've recently change the port on my asterisk to 5070 and want to ensure it's still working, i've also changed the context
20:34.05Qwelliceyp: one sec
20:34.08mutilatoravg of 5
20:34.11iceypthanks qwell
20:34.17PBXtechhow does the Random command work?
20:34.19tzangermutilator: I am guessing that latency has ntohing to do with it but rather when you get higher latency links it's because the wirless network is more congested, leading to higher latency due to more collisions and collision avoidance
20:34.25Blissex_chad: well, yes as to connect directly. Asterisk is a fully symmetrical (even too symmetrical) switch. It does not know which end is the pprovider and which end if the home phone...
20:34.27harryvvmutilator do you have windows and would care to test a voip connection to my server with xlite?
20:34.28nvrsworkREN values?
20:34.31Qwelliceyp: 1 ring - fast busy
20:34.34bjohnson_chad: if you can cut Texas out of the routing you can save 50ms on your calls to your house.  You an always route outgoing directly from your house sip phones .. and would be a good test for call quality if you hosted your * locally there
20:34.38_chadblissex, ahh good stuff
20:34.46Dreltzanger: Well, a bad headache anyway :)  We're going to have to run ethernet anyway, I think (we're currently using 802.11g to all desktops, so our current wiring (or lack thereof) makes any wiring addition look unpleasant.
20:34.55tzangerahhh
20:35.08iceypqwell: mmm, let me lookie
20:35.09mutilatoryea tzanger, thats whats going on
20:35.16Qwelliceyp: I had verbose off, lemme try again, see what it says on my end
20:35.18mutilatorbut it's pretty much a direct correlation on wireless
20:35.24_chadbj,bliss, good deal
20:35.26mutilatorwhereas dsl it's not
20:35.35Blissex_chad: all Asterisk does is given an extension, find the address for that extension, and then tells the requesting endpoint to setup the call to that address, ideally.
20:35.35*** join/#asterisk ariel_ (~Ariel@ip67-93-229-222.z229-93-67.customer.algx.net)
20:35.36_chadbj/bliss: would upping my jitter buffer help some as well?
20:35.37Qwelliceyp: yeah, it hits FWD, then dies at you
20:35.40DrelThis is an old bldg w/ no ceiling panels (we have 17' ceilings instead) and an open layout, so running wiring isn't easy.
20:35.49tzangermutilator: I suppose...  distance would give you more latency without more CSMA action :-)  but it'd be long links
20:35.52_chadblissex, i see.. and take itself out of the picture
20:36.00tzangerseventeen foot cielings... NICE
20:36.06tzangerecho cho ho o...
20:36.06mutilatorwe've got links up to 15 miles on some customers
20:36.06harryvvI need somone to test a voice connection to a remote client.
20:36.13mutilatormost run about 5
20:36.15Blissex_chad: it will help if you are getting breakups, but it will make delay worse...
20:36.30iceypQwell can you try again plz
20:36.37bjohnsonDrel: I thought I saw someone mentrion REN .. you only need one fxs per phone if you want them to act as independant extensions
20:36.41iceypoops
20:36.42iceypApr  5 08:36:17 WARNING[99164]: chan_iax2.c:4300 iax2_register: Host 'iax2.fwdnet.net/01' not found at line 12
20:36.43mutilatorsome backbone links run 20+ miles
20:36.44iceyp1 sec
20:36.47Qwelliceyp: no go
20:36.48_chadhttp://www.pastebin.com/267135 w/in the norm?
20:37.32DannyFanyone played with FastSMS btw?
20:37.32bjohnson_chad: jitterbuffer adds delay to the beginning of the call usually
20:37.37Drelbjohnson: having the legacy analog phones we've invested in relegated as "extra"/floating phones that share a single extension would probably be acceptable.
20:37.48tzangerbjohnson: it adds delay thoughout the call, that's its purpose
20:37.53Blissex_chad: yes, if Asterisk cannot take itself out of the picture, because the calling and called endpoints cannot connect directly, then it will switch to transparent mode, in which it will store-and-forward data from endpoint to another, over two connection isntead of one.
20:37.58facek_hm
20:38.00*** join/#asterisk madounet (~mad|net@juvenal-3-82-226-155-19.fbx.proxad.net)
20:38.04Drelbjohnson: In that case, do you see any problem with having 4 phones on a single FXS?
20:38.19tzangerDrel: they will act as one extension... is that what you want?
20:38.32_chadAh, bliss- so if I understand you.. the jitter buffer is not really even a factor once the reinvite is made?
20:38.42*** join/#asterisk RubyT (~joeblow@lan.mobilcom.net)
20:38.57Dreltzanger: It's not ideal, but if it saves a few hundred dollars on either scrapping the analog sets or investing in ATAs, it might be acceptable.
20:39.02Blissex_chad: the jitter buffer in _Asterisk_ is no longer an issue _if_ the reinvitation succeeds.
20:39.14tzangerDrel: well you can't have it both ways, so it's your choice.  :-)
20:39.21bjohnsonDrel: depends on the ata and how much voltage the phone draw from the line.  I have 6 powered phones and 2 non-powered phones on one SPA 3k fxs port
20:39.46Blissex_chad: but the endpoints themselves will usually have buffering too, for the same reasons why Asterisk has buffering too.
20:39.57_chadi see
20:40.05Drelbjohnson: The analog sets are Polycom SE-220 powered sets, we'd probably go with a Digium TDM400P
20:40.25jpeanyone have any insight as to why I cant get dids entered through amp to work via aah 0.8?
20:40.31bjohnsonDrel: you could always do a mix .. often not all the extensions need to be independant (think home system) but in business .. you usually want them to be independant (how does it look if on with a customer and someone else picks up to order pizza)
20:40.41*** join/#asterisk klictel (~klictel@207.107.208.137)
20:40.56_chadchanging my cisco right now to dial out directly
20:41.31klictelhi all
20:41.43iceypWhy would i get this when trying to receive a call from FWD
20:41.44iceypApr  5 08:41:21 NOTICE[99654]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 65.39.205.121
20:41.45Blissex_chad: Asterisk also has buffering because it has _three_ modes of operation: one is resolve the call and then reinvite the endpoints to connect directly to each other. Another is transparent mode, where it passes data between the endpopints. The third is when it acts an endpoint itself, that is terminates call in itself, like when it does voicemail.
20:41.53iceyp65.39.205.121:4569    265744      202.150.105.150:4569       60  Registered
20:41.56mutilatorbjohnson: least you know that there is pizza on it's way :P
20:41.56Drelbjohnson: Well, as long as the line light was illuminated showing the extension to be in use, I don't think it would be an issue, it wouldn't be any different from our current situation.  The phones are two line, they show which line is in use, and if one is in use, they pick up the other line when you pick up the handset.  What I'd probably do is buy two FXS modules and hook those up to the...
20:41.58Drel...internal phone wiring so that each module handled a line on the analog sets.
20:42.03Drelmutilator: hehe.
20:42.12facek_what i need to allow my peers connect by by h323?
20:42.37Blissexfacek_: it is a FAQ... Check the Wiki.
20:42.39jakepdevfacek - you need JerJer
20:43.14facek_jakepdev right ;]
20:43.42RubyTIs there a way to have a TDM400 FXO port detect if a working phone line is connected to it?
20:43.46iceypApr  5 08:43:15 NOTICE[99654]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 65.39.205.121 <--- Why do i receive this error when trying to receive a call via FWD?
20:43.46*** join/#asterisk platcd (Atropine@S0106000f3d37a96d.va.shawcable.net)
20:43.56_chadbliss/bj/etc its amazing how much you guys know about this stuff
20:44.27hardwireits because they have no women!
20:44.28hardwire:)
20:44.33mutilatorooo disssss
20:44.38hardwireI would be so insanely smart if a woman wasn't stealing all that energy
20:44.40bjohnsonDrel: or .. with an internet connection, and fxs per phone, and some dialplan routing you could give each phone access to what would like like it' own independant line for outgoing calls.  In the end .. you decide what works best for you .. ATA's come for as little as $35 USD per fxs port
20:45.02_chadhardwire, haha :D
20:45.26Qwellhardwire: I wouldn't be able to find my socks. :p
20:45.32Blissex_chad: <hardwire> may me more right than you think... Freud and sublimation :-).
20:45.33hardwirehaha
20:45.53platcdgood day all.  I just starting reading about Asterisk, Digium hardware etc, and I must say I am stoked to build a PBX for my small home office.  What do I need hardware wise.  I simply require 2 unique lines such that one operator can be talking while another call can come in... Are there boards with simple POTS interfaces ? What do I need to get started ?
20:46.07Drelbjohnson: Good to know.  Basically, I need to come up with some suggestions and pricepoints for my boss to consider, so it's good to know what the alternatives are!
20:46.11bjohnsonDrel: of course you could also stick regular cordless phones on a fxs
20:46.23sudhir492facek_: I have recently compiled h323 with CVS head. openh323 1.17.1, pwlib 1.8.1. Works well so far
20:46.32BlissexFreud said that sublimated horniness is the very motor of civilization. Or at least geekyness I say :-)
20:46.40_chadSo all I need to is punch in my nufone register information (from iax.conf) into my cisco 7940 sip preferences (as the proxy and whatnot) and i would be connecting directly yah?
20:46.55tzanger_chad: your 7940 can do IAX2?
20:47.07iceypQwell you there?
20:47.10Qwellyeah
20:47.12Qwelltry again?
20:47.13Blissex_chad: if you want, but I suppose it's better for you to resolve calls via your Asterisk anyhow.
20:47.18iceypyou using fwd via iax?
20:47.20bjohnson_chad: register tells the only end what IP address to send incoming calls to
20:47.21Qwellyeah
20:47.27iceypcan i see your iax.conf entry
20:47.37iceypsomething is broken with mine
20:47.54iceypi'm rejecting the calls
20:47.55iceypApr  5 08:46:59 NOTICE[99654]: chan_iax2.c:5405 socket_read: Rejected connect attempt from 65.39.205.121
20:47.57Drelplatcd: I'd recommend a Digitum TDM400P with two FXO modules to handle your incoming POTS lines.  Keep in mind I have very little real world experience with Asterisk so far, though :)
20:48.00iceypdont know why
20:48.04bjohnson_chad: I think you want to leave texas as your incoming * server until you do some testing on your home connection
20:48.10Blissex_chad: so on outgoing you can access all your menus/stored numbers/whatever logic you got. Phones are a lot dumber than Asterisk.
20:48.42Qwelliceyp: http://pastebin.ca/8803
20:48.43_chadwhew lot to learn
20:48.47Blissex_chad: also on incoming you want Asterisk to do voicemail etc. for you.
20:48.58_chadblissex, yeah exactly
20:49.09_chadblissex, reinvite dosn't take those elements out of the picture does it?
20:49.20*** join/#asterisk AgiNamu (~AgiNamu@216.230.151.230)
20:50.01Blissex_chad: no, reinvitation only handles the data transfer bit. All those elements are about call initiation etc.
20:50.06bjohnsonif you decide your home connection can handle enough calls ok .. you can move your * server there and forget the texas location .. but as far as * is concerned, I think you should consider getting a colo closer to you or your voip provider
20:50.30_chadbj, definitely.. seems like i have just too many bends in the pipes
20:50.36iceypQwell you dont have the host iax2.fwdnet.net & user / pass?
20:50.53Blissex_chad: however bjohnson has a good point if your phone can do IAX2 and SIP, make sure it uses the one that your provider uses.
20:50.54iceypdo you provide your user/pass in the dial?
20:51.00Qwelliceyp: yeah
20:51.44iceypcan you try call me now 265744
20:51.44AgiNamuyes, make sure you use the right protocol. that'
20:51.48AgiNamuis some good advice :)
20:51.56_chadblissex, aiee another concept entirely :)
20:51.58AgiNamuI had someone flash a phone to IAX2, then send me a SIP debug log and ask why it didn't work
20:52.14clive-AgiNamu what phone is this?
20:52.18Qwelliceyp: yeah, that worked
20:52.18AgiNamuanyone here have experience with PortaOne?\
20:52.22_chadI don't believe this cisco 7940 supports IAX2 does it?  I know I went through a grip to get it converted to SIp
20:52.22AgiNamuclive- PA168
20:52.27Qwell"Please enter the extension of the user you wish to connect to"
20:52.30bjohnsonnope .. just PortaPotties
20:52.43neophergot probs with x100p, I did load properly and i still get Apr  4 16:25:00 WARNING[3942]: Ignoring switchtype
20:52.43AgiNamuPA168 is the only phone I know of in mass production that supports IAX
20:52.58bjohnson_chad: no cisco phones currently support iax2
20:53.03bjohnsonnone
20:53.06AgiNamuPA168 is a chip made by a Chinese company that's used a LOT
20:53.07clive-AgiNamu, I have th pa168 also....just doesnt do native transfer...yet
20:53.20AgiNamuclive- no. I might implement it some day
20:53.24_chadbj, k
20:53.25AgiNamuif they dont do it for v1.43
20:53.39AgiNamuI added some more support to their 1.42 IAX firmwar
20:53.48AgiNamuand i will probably work on it a bit more soon
20:54.08clive-AgiNamu, are you the one doing the iax firmware?
20:54.08_chadbj/bliss, so 1) check to see if reinvite gives me the results I want.. if not 2) setup a local * box  and 3) start trying closer pstn boxes
20:54.13AgiNamui dont know what all is involved in native transfer. i'll have to read chan_iax2 a bit more
20:54.14malverianneopher: I get that as well, but everything still works fine.
20:54.20_chadthat the gist of it?
20:54.24AgiNamuclive- no, they do it
20:54.29AgiNamubut I wrote some patches for them, to help
20:54.31iceypQwell  thanks
20:54.37AgiNamucause i needed, for example, POKE/PONG support
20:54.40AgiNamuso I wrote it, sent it to them
20:54.45neophermalverian: i am not able to make or receive calls via zap
20:54.46AgiNamuand they're including it in 1.43
20:55.04AgiNamuclive- do you know much about IAX2 , so as to write firmware? if so, i'll send you the source and you can work on it
20:55.11malverianneopher: Probably an unrelated issue then.
20:55.12jaigerAgiNamu, they sent you their source code?
20:55.16AgiNamuyes
20:55.36clive-AgiNamu, I wish I had the time and knowledge to do it all myself, but I cant:(
20:56.09AgiNamuSteveK told me that their iax is based on an old libiax
20:56.13AgiNamubut with every line of code replaced
20:56.33malverianneopher: Do you have wcfxo linux module installed?
20:56.43facek_what is the best way to divide outgoing calls by prefix to specifi channel to make call ?
20:56.48clive-AgiNamu, I am surprized no-one from Digium has partnered with atcom to create a good iax product...oh well, just have to wait for my native bridge transfer to happen
20:56.53facek_and when channel is must other some other channel
20:57.07AgiNamuclive- when you say native transfer
20:57.15AgiNamuyou refer to performing a transfer? or some kind of bridge?
20:57.33AgiNamucause native bridging works fine. its just that the transfer button doesnt do anything
20:58.05clive-AgiNamu, native bridging,,,doesnt work for me, Woody said its the phone thats not transferring
20:58.24AgiNamuyea there is no transfer support
20:58.35AgiNamubut when i place a call, * natively bridges it to my gateway no problem.
20:58.59clive-AgiNamu what is your setup that allows the bridge?...
20:59.12AgiNamuPA168 -> Asterisk -> Asterisk -> Telica
20:59.16AgiNamu( -> DS3)
20:59.34hardwireooh
20:59.36AgiNamumy asterisk machine bridges just fine.
20:59.40clive-and the first asterisk transfers out of the loop
20:59.54AgiNamuno, no transfers. just native bridging
20:59.59AgiNamui dont want that any ways
21:00.02AgiNamufucks with my CDRs
21:00.15AgiNamuand im worried about the NAT issues (which is why im using IAX in the first place)
21:00.46AgiNamumaybe im misunderstanding. i see native bridging as when asterisk can just passthrough the frames
21:00.48neopherbeing that i am using kernal 2.6.9-5, do i need to do make linux26 instead of make
21:01.07AgiNamuand transfer when it actually can reconnect to another server
21:02.40*** join/#asterisk hajekd (~hajekd@21.208.65.212.contactel.net)
21:02.55hajekdis it possible to pass argument to agi script?
21:03.33bkw_yes
21:03.35bkw_go read the notes
21:03.52bkw_[E|Dead]AGI(command|args):
21:04.06bkw_and next time say Hi before you burst in to ask questions please
21:04.51QwellHi!
21:05.09rvhiany one knows how to extrace 'username' in 'from' header of a sip packet?
21:05.23rvhie.g. From: <sip:123@test.com>
21:05.35rvhii'd like to see 123 in one variable
21:06.08rvhioops,
21:06.10rvhiHi!
21:06.51bkw_*SMACK*
21:06.52bkw_hehe
21:07.05*** part/#asterisk platcd (Atropine@S0106000f3d37a96d.va.shawcable.net)
21:07.17bkw_I think thats already done for you
21:07.22Blissexrvhi: I think that's documented.
21:07.28*** join/#asterisk KristinG (~KristinG@muppet.geekgirls.us)
21:07.31jakepdev~docs
21:07.33jbotDocumentation 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, or http://www.asteriskdocs.org
21:07.33Blissexrvhi: it is a prese3t variable.
21:07.39bkw_rvhi, you have ${SIPURI}
21:07.46KristinGgood afternoon
21:08.02bkw_Blissex, thats not what he asked
21:08.02Beirdo~beer
21:08.03jbotit has been said that beer is ummm, ummm good!, or good for you!
21:08.06*** join/#asterisk AmigaMan (~fsck@cpe-68-203-212-229.rgv.res.rr.com)
21:08.09Beirdogood afternoon, KristinG
21:08.26KristinGcan anyone tell me if there are  any open source voice mail systems that would work with asterisk?
21:08.28Beirdoit has one built in
21:08.32MikeJ[Laptop]~guatamala
21:08.33jbotguatamala is, like, small central america country jointly owned by the USA government and the Illuminati.
21:08.40KristinGcomedian is lacking in a number of ways
21:08.44bkw_KristinG, dude
21:08.48bkw_asterisk has one already
21:08.50Beirdowell, it's open source :)
21:08.50BlissexBeirdo: she did not ask that, she asked if anyone could tell her...
21:09.05BlissexKristinG: the answer to your question is "yes". :-)
21:09.14jakepdevwhat you don't like you can change
21:09.15BeirdoBlissex: be nice :)
21:09.16bkw_its app_voicemail.c
21:09.19KristinGI'm not a dude but yes, i am awarrrrrrrrrrrrrrre of the native app
21:09.24rvhiis sipusr only in head?
21:09.38bkw_KristinG, YAY another girl in here
21:09.44MikeJ[Laptop]another?
21:09.50bkw_Katty and KristinG
21:09.52KristinGthere are features that my booooooooooooooooooooooss would like that asterisk doesnt offer
21:09.55BeirdoKristinG: "dude" is often used generically, but from your name, I was fairly sure you aren't a dude :)
21:10.04bkw_KristinG, what is that?
21:10.04Beirdoso add them
21:10.09KristinGthank you :)
21:10.11Beirdothe source is right there :)
21:10.18eKo1You should have used dudett
21:10.22file[laptop]cause it's easy once you know how it's done
21:10.22jakepdevette
21:10.23BlissexKristinG: thats very unlikely, but... What is one such feature?
21:10.25bkw_first lets hear what the PHB wants
21:10.26file[laptop]you can't stop now it's already begun
21:10.39Beirdooh good point.
21:10.44KristinGplaying back your recorded grettings
21:10.47facek_anyone had create a good dialplan using AGI? or made in AGI without extensions.conf (only ro run some scripts by agi)
21:10.51bkw_KristinG, thats there already
21:10.57QwellNEXT!!!
21:11.00Qwell:p
21:11.06KristinGhold on
21:11.07bkw_if not its a few lines of code to add it
21:11.07KristinGbrb
21:11.15bkw_but I'm sure it is
21:11.21file[laptop]bkw_: back to work you slacker!
21:11.26bkw_WORK?
21:11.28bkw_call 996 you hoe
21:11.33file[laptop];)
21:11.38file[laptop]nah
21:11.40*** join/#asterisk objectivelogic (~tinyiko@196.46.67.175)
21:11.42bkw_YES YOU WILL
21:11.44jpecalling from a cell into the syste,
21:11.44bkw_and you WILL NOW
21:11.52file[laptop]I'm quite tired and am currently being in a semi-concious state on my bed
21:11.53jpecalling from a cell into the system
21:11.56file[laptop]THINKING OF YOU DEAR!
21:11.56MikeJ[Laptop]hoe... now that's not very nice... is that the way to make friends bkw_?
21:11.58file[laptop]:p
21:11.58MikeJ[Laptop]:D
21:12.55Beirdostupid ebay arsehole didn't have the manual
21:13.26rvhii looked at http://voip-info.org/wiki-Asterisk+variables, can't find the username field
21:14.15luke-jr_Does any LC routing exist that does not require SQL?
21:14.41Darwin35who did what to who
21:14.47Blissexrvhi: but you probably have found something from which you can extract it...
21:15.01iceypshould asterisk be able to handle a call in this way.... FWD -> Asterisk -> Ser -> Voip Phone ?
21:15.14iceypMy problem is when the call connects between asterisk and Ser, it dies on first ring
21:15.22luke-jr_iceyp: ...why?
21:15.32luke-jr_are * and SER on the same box?
21:15.40KristinGok back
21:15.52iceypluke-jr_  yes
21:16.12KristinGplayback-options: rewind/pause/foward through a message
21:16.18iceypdiff ports obviously, 5060 for ser & 5070 for *
21:16.19luke-jr_iceyp: I'd check to make sure SER is configured for a non-default port
21:16.29KristinGtimestamp of rmessage
21:16.45facek_can i use ChanIsAvail from AGI scripts?
21:17.05luke-jr_iceyp: * dying on first ring sounds like its looping to me
21:17.17KristinGand notification, ie via, email, pager or remote phone
21:20.02objectivelogicneed help getting my asterisk to terminate to POTS
21:20.15QwellKristinG: It does that too
21:20.32objectivelogicno ringing,just says Spawn extension (from-sip, 2719, 1) exited non-zero on 'SIP/tinyiko-9d1d'
21:20.32objectivelogic<PROTECTED>
21:20.32objectivelogic<PROTECTED>
21:20.32objectivelogic<PROTECTED>
21:20.41FirstSwordanyone heard of Sangoma's card?
21:20.49file[laptop]objectivelogic: ${EXTEN} not $EXTEN
21:20.51Qwellobjectivelogic: ${EXTEN{
21:20.55Qwellyeah, what he said
21:21.11objectivelogicok, let me try that
21:21.57KristinGok guess i have to pick the source apart then
21:21.57AmigaManhas anyone setup h.323 with g729?
21:22.21KristinGjust to confirm though, there are no other packages that will work with it?
21:23.16iceypok, can someone do me a favour and dial 64273040757@max.fast.co.nz
21:23.33BlissexKristinG: no need usually to look at the source...
21:23.46*** join/#asterisk Zipper_32 (~none@s207-6-25-182.bc.hsia.telus.net)
21:23.54BlissexKristinG: there are quite a few examples of even some obscure features online.
21:24.14KristinGoh? i'd love a link or two then
21:24.51BlissexKristinG: for example, from one such example I think I discovered that there can be two sections with the same name in 'sip.conf', one with 'type=peer' and one with 'type=user'. Amazing.
21:25.03BlissexKristinG: as usual, there is the Wiki...
21:25.04Nivex~wiki
21:25.38BlissexKristinG: but if you do a clever Google search you can find a number of fairly sophisticated sample 'extensions.conf' that people have posted on the web.
21:26.22BlissexKristinG: by «clever Google search» i mean one that contains a few keywords that are likely to be present only in an 'extensions.conf' file.
21:27.42KristinGok thanks
21:27.49KristinGwill look into it
21:27.58KristinGappreciate the info
21:28.04BlissexKristinG: for example "static=yes", "exten", "voicemail", "playback", "gotoif"
21:29.03*** join/#asterisk Carp1 (carp_xigon@ip-204-97-151-247.modem.logical.net)
21:29.19Carp1I just updated Asterisk last night.  Last update before that was like 6 months ago.
21:29.28Carp1I'm now getting errors trying to load chan_zap
21:29.39Carp1and setup_zap
21:29.39*** join/#asterisk r0d3nt|m (nobody@wsip-24-234-241-84.lv.lv.cox.net)
21:29.39iceypcan someone please ring 64273040757@max.fast.co.nz and tell me if they hear ringing or if it goes dead
21:29.53`SauronI tried to call it with my cell phone
21:29.54eKo1eh, did you zapata?
21:29.57`Sauronbut it complained about the @
21:29.57`Sauron;)
21:30.05eKo1Or zaptel or whatever
21:30.23objectivelogicthanks guys that helped
21:30.34BlissexKristinG: for example try this: http://www.google.com/search?num=100&as_q=exten+exten+exten+voicemail+playback+gotoif
21:31.06Nivexwww.voip-info.org
21:31.11Nivexall hail the wiki :)
21:31.23Carp1any one know anything about my problamo?
21:31.31BlissexNivex: Google is giant Wiki :-)
21:32.01*** join/#asterisk kFuQ (~somedude@c-24-17-224-78.hsd1.wa.comcast.net)
21:32.11file[laptop]colddddddd
21:32.37eKo1Carp1: I told you, upgrade zaptel.
21:34.38Carp1I did.
21:35.15*** join/#asterisk Sedorox (brandon@Neptune.client.wlmsprt.pa.sed6.net)
21:36.30*** join/#asterisk RaYmAn-Bx (user@213.237.12.147.adsl.vby.tiscali.dk)
21:37.49eKo1What kind of error is it?
21:38.02*** join/#asterisk RaYmAn-Bx (user@x1-6-00-11-80-c3-8a-d7.k441.webspeed.dk)
21:38.02eKo1Better yet, post the error somewhere.
21:38.16Carp1Just a second
21:39.49Carp1www.pastebin.ca right?
21:40.23eKo1Whatever, as long as you don't flood this channel.
21:40.46Carp1http://pastebin.ca/8806
21:42.30eKo1What does ztcfg say?
21:46.22*** join/#asterisk Goshen (~Goshen@70-57-80-147.slkc.qwest.net)
21:46.32*** join/#asterisk hohum (corbe@snoop.burghcom.com)
21:46.35hohumhey
21:46.43hohumI have a very SIP specific question
21:47.29hohuma B2BUA is supposed to take one call, terminate it on the gateway, initiate another call and proxy the communication between the two call legs, correct?
21:47.38hohumso
21:47.46hohumI have a SIP proxy that acts as a B2BUA
21:48.58hohumit takes an INVITE from another sip proxy from one UA and sends it back to the same gateway where another UA is registered
21:49.11facek_how to allow calling people to make a transfer to another consultant?
21:49.35Qwellhmm
21:49.39hohumI noticed that my proxy server stamps Via: into the headers
21:49.48Qwellfacek_: You just gave me an odd random thought
21:49.53hohumwith a unique TAG line
21:50.12hohumhowever the Call ID is the same
21:50.16facek_Qwell what?
21:50.17QwellWhen you call a tech support place, before the person answers, "If at any time, this rep says anything retarded, please press 1.  The call will be logged, and you will be transfered to another rep"
21:50.27Qwell:p
21:50.29*** join/#asterisk MasterYoda (~mnicholso@207.111.174.1)
21:50.32hohumand Asterisk that sent the original invite responds with a 482 Loop Detect
21:50.34facek_Qwell right
21:50.39hohumis that because the Call ID didn't change?
21:50.43file[laptop]hohum: yes
21:50.43*** part/#asterisk MasterYoda (~mnicholso@207.111.174.1)
21:50.51file[laptop]hohum: a call id distinctly identifies a call
21:50.56Qwellfile[laptop]: Help me get ISPs to implement that, would ya?
21:51.02hohumand if so is that Asterisk's fault for not respecting the TAG line or my B2BUA's fault for not changing the Call ID?
21:51.14hohumfile: a call?  or a call leg?
21:51.15file[laptop]hohum: call id should change because it's a different call
21:51.20file[laptop]a call leg
21:51.23hohumokay
21:51.41file[laptop]a B2BUA acts as if there are two different calls in progress, as a bridge between the two... Back to Back User Agent
21:51.51malverianSo what's all the hubub about IAXcom ?
21:51.52file[laptop]it's a user agent for both... so the Call ID should be different
21:51.53hohumthis is all a massive excersize in debugging a call forwarding problem from my soft switch
21:51.58facek_how to allow calling peolpe to make a transfer
21:52.01facek_o configiue mgsp.conf
21:52.04malverianiaxtel rather
21:52.05facek_some extensions?
21:52.07file[laptop]hohum: sounds like it's acting as a proxy
21:52.47jpe<PROTECTED>
21:52.56jpei wish i could. i have been stuck all day trying to get did to work
21:53.22jpesorry as you can guess I am new to this irc thing, my bad
21:54.00hohumI'm trying to see if my Cisco AS5300 behaves in a similar manner
21:54.37malverianHere's a better question.. how useful is FWD ?
21:54.42malverianAnd how hard is it to set up?
21:54.43*** join/#asterisk three55ml (~none@cpe-66-68-98-68.austin.res.rr.com)
21:54.54Qwellfwd is good, and really easy to setup
21:55.05Qwellthe sample configs show how to connect to fwd even
21:55.28three55mlAnyone with a SIP phone or ATA interested in testing out a VoIP service please PM me.  I'll give you a DID to use during testing.
21:55.28facek_how to allow calling peolple to make transfers?
21:55.32malverianQwell: What numbers can you dial?
21:55.43eKo1jpe: How did you get here anyways?
21:55.44Qwellmalcolmd: other fwd users ;p
21:55.49Qwellerm
21:55.51malverianQwell: Figured.. that kinda sucks :)
21:55.52Qwellyeah...
21:56.21eKo1three55ml: DID from where?
21:56.32three55mleKo1: US or 800
22:00.45*** join/#asterisk Kyrin (~gostlund@d198-53-224-65.abhsia.telus.net)
22:04.03eKo1What area code?
22:06.54KyrinAnyone here get Asterisk working under Gentoo?
22:07.02Sedoroxyes
22:07.05facek_Kyrin yes
22:07.16Kyrinfacek_: my ebuild keeps breaking
22:07.22*** join/#asterisk PTG123 (~PTG123@66.213.239.122)
22:07.33facek_Kyrin i installed from scratch
22:07.57*** join/#asterisk Pinhole (~nuxi@lin-dsl-static-206-222-194-115.inetnebr.com)
22:07.59KyrinAh, haven't tried that, kinda the purpose behind Gentoo, isn't it?
22:08.09KyrinI mean, it's doing the "build from scratch" for me
22:08.32SedoroxI did a ebuild
22:08.44KyrinSedorox: No issues I take it?
22:09.02Kyrin[ebuild  N    ] net-misc/asterisk-0.9.0  +alsa +apache2 -doc -gtk -mmx +mysql -nopri -nozaptel 0 kB
22:09.17Sedoroxnot for me
22:09.19Sedoroxwow
22:09.20Sedoroxdude...
22:09.22Kyrinthat's my flags, what should I have?
22:09.22Sedoroxemerge sync
22:09.25Sedoroxthen run that again
22:09.37Sedoroxthats a REALLLY old version
22:09.45Kyrinwas sync'd yesterday, what version should it be?
22:09.52Kyrinther 1.0.7 is masked by x86
22:09.54SupaplexI should write a xmms control to have it pause while the phone is ringing, or while I'm on the phone
22:09.59Kyrinshould I just accept keywords that?
22:10.28Sedoroxyes
22:10.39KyrinOh, what are those last two options 'nopri' and 'nozaptel' do I need those?
22:10.49SedoroxACCEPT_KEYWORDS="~x86" emerge asterisk -p
22:10.59Sedoroxno.. that installs the pri and zaptel stuff
22:11.01Sedoroxits fine
22:11.17Sedoroxor whatever it is to add the keywords.. just for that build
22:11.21Kyrink, 'cause my error messages said something about pri so I wasn't sure
22:11.33Sedoroxwell I'm sure there are new versions
22:12.08SupaplexCan I get some comments on link2voip and the overall service/expecations etc?
22:12.31Sedoroxthey SUCK
22:12.39Kyrin[ebuild  N    ] net-misc/asterisk-1.0.7  +alsa -bri -debug -doc -gtk -hardened -mmx +mysql -postgres -pri -resperl -speex (-uclibc) -vmdbmysql -vmdbpostgres -zaptel 0 kB
22:12.41*** join/#asterisk MarkS_ (~marks__@cpe-70-112-81-84.austin.res.rr.com)
22:12.43Sedoroxme abd Beirdo  have waited over a  month for a toll free number
22:12.47Sedoroxwe have YET to get it
22:12.53Sedoroxand I signed up Feb. 25
22:12.56Supaplexsucky :(
22:12.59SedoroxI do not recommended them at all
22:13.06Supaplexyea, I need a 800 number
22:13.08BeirdoI have one
22:13.17Beirdobut not through those assmasters
22:13.20SedoroxBeirdo: tell him the company that we ended up going with
22:13.23Sedorox'cause I forget
22:13.23Sedorox:-p
22:13.27Beirdothinktel.ca
22:13.40SupaplexI already have vonage, but I wanna ditch their linksys-ata and use asterisk with sipura or something
22:13.41Sedorox:)
22:13.46Sedoroxheh
22:14.11Supaplexsome day we'll learn how to fix that issue :)
22:14.31Sedoroxlol
22:14.50Kyrinhrm... 2.4.28 kernel
22:19.04Zipper_32Inside a small office, is it recommended to have a dedicated ethernet connection running to each desk for VOIP, or should it share a standard data cable?
22:19.21*** join/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
22:19.35ManxPowerI'm having a really bad day.  What newbie can I abuse?
22:19.44KyrinOOh, pick me
22:20.02_Vileabuse bkw, he likes it
22:20.07KyrinI know very little 'bout VoIP, just what I've read on the 'net, and lots of questions
22:20.07Supaplexthinktel.ca is in 0 wiki articles ;/
22:20.10Zipper_32ManxPowe,r you can abuse my last question...
22:20.52*** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net)
22:20.56KyrinWhat would be the recommended down/up bandwidth for 12 lines to an office?
22:21.30*** join/#asterisk dalabera (~Dalabera@mail2.pmrtechnologies.com)
22:21.41hardwireKyrin: python makes a nice console calculator
22:21.45*** join/#asterisk zilas (~1@c-24-30-75-206.hsd1.ga.comcast.net)
22:22.11zilashello
22:22.20harryvvbetwen my router and somone elses end router we have ports 5060,5061,10001-10010 open for a sip client to server connection but still cannot hear each other. Getting this error in cli. WARNING[9601]: chan_sip.c:787 retrans_pkt: Maximum retries exceeded on call 4AFFF90F-2C0D-456E-A91D-EB178C3B0100@192.168.1.100 for seqno 24565 (Non-critical Response)
22:22.41*** join/#asterisk montoya (montoya@200.195.87.110)
22:23.04Kyrinhardwire: so does echo $[x*12], but I don't know the numbers I ought to put for 'x'
22:23.06harryvvhi zilas
22:23.30hardwireKyrin: voip-info.org has the specs
22:23.39Kyrinhardwire: thx, will read that now
22:23.46zilasfor call parking exten => a,b,(SIP/phone1,20,tr) what does here a and be stand for?
22:24.03Kyrinomfg
22:24.09hardwireomfg?
22:24.11Kyrinhow come I never found this one before
22:24.16*** join/#asterisk ChulJin1 (~chuljin@adsl-68-121-94-237.dsl.irvnca.pacbell.net)
22:24.21hardwireKyrin: because you are a crazy person
22:24.24Kyrinthis site, this wiki
22:24.25Kyrinwow
22:24.42harryvvany of you link a remote sip phone to your asterisk box?
22:24.55*** join/#asterisk juice (~juice@mo-69-68-108-44.dyn.sprint-hsd.net)
22:25.03KyrinIt's like never I saw it, dunno why, reading now
22:25.14ChulJin1harryvv: all over the world, yeah
22:26.29zilaswhy this call parking so purely documented I can't make it work :( can someone share some tips with me ples...
22:28.19AgiNamuanyone care to explain how IAX2 transfers work?
22:28.30AgiNamuI'm looking at a netcap and it is... wierd.
22:28.54Kyrinhardwire: So, if I'm reading this right... if i have a 1mbit upload, that might be enough for 12 lines if I don't use a the uncompressed PCM codec
22:28.55rowteram trying to work the app_directory, exten => 30,1,Directory(office,f) it does not 1 => 123,Test,test@tes.com from voicemail mmh..
22:28.56AgiNamuI have Asterisk1 calling Asterisk2 which calls Asterisk3. 2 should transfer out, and thats what I see.
22:29.11AgiNamubut from the tcpdump, it seems as if Asterisk1 initiates the transfer :\
22:30.05*** join/#asterisk Hogie (~daniel@alpha.dfwservers.net)
22:30.18eKo1That info. is classified I'm afraid.
22:30.37*** join/#asterisk captrb (~crozierm@64.65.134.42)
22:31.44AgiNamueKo1, judging from chan_iax2 ... yea :P
22:31.45*** join/#asterisk dmccollum (~dmccollum@69-164-245-72.atlaga.adelphia.net)
22:32.34eKo1Ask kram, he's the iax2 wizard of the north.
22:32.58eKo1I rest my case.
22:33.04*** join/#asterisk goobster (goobster@c-67-168-105-166.hsd1.wa.comcast.net)
22:33.11file[laptop]AgiNamu: I have a doc around here on the procedure
22:33.15*** join/#asterisk xkev (kevin@orbit.xmission.com)
22:33.59*** join/#asterisk chfn (~adolfo@c906ff56.virtua.com.br)
22:34.02*** part/#asterisk ManxPower (~eric@dsl-209-16-67-160.datasync.com)
22:34.24KyrinHow easily can several different numbers be routed through and managed (transfers, holds, conference calls, etc) one asterisk system?
22:35.03AgiNamufile... mgg@atrevido.net :)
22:35.16file[laptop]AgiNamu: doesn't mean I'll find it
22:35.16AgiNamulol ok
22:35.20AgiNamuinstall MSN desktop search.
22:35.51Supaplexs/MSN/google/
22:35.53Supaplex:P
22:36.05*** join/#asterisk loko (rbrown@67.171.66.213)
22:36.06eKo1how about using grep instead
22:36.47chfnhi guys.
22:37.55chfnCan I do this with asterisk? a script will run a 'tool' of asterisk then asterisk will call this number to play a mp3 message.
22:38.00chfnwill I need to code?
22:38.49chfna script of asterisk will be runned to call some number then will play a mp3 message. Can i Do this ?
22:38.58Nivexchfn: http://www.voip-info.org/tiki-index.php?page=Asterisk%20auto-dial%20out
22:39.01lokoDarwin35, you around?
22:39.07eKo1chfn: yes
22:39.21chfneKo1, Nivex so will I be able to remember people of something ?
22:39.23eKo1Thank you for participating.
22:39.35Nivexchfn: using /var/spool/asterisk/outgoing, code will be minimal
22:39.49*** join/#asterisk elriah (~jfulcrum@adsl-068-209-198-242.sip.bhm.bellsouth.net)
22:40.21elriahHi guys, on a production asterisk system that services 25-250 phones, running bare-bones linux and no X windows at all, would 256MB of RAM be enough?
22:40.25*** join/#asterisk bjohnson (~bjohnson@ip202-181.tor.istop.com)
22:40.36chfnNivex, i love you man! :) sorry for telling you that but I(stupid guy) told a owner(company) that asterisk probably does that. asterisk will be everything talking about PBX.
22:40.47chfnNivex, cool project. I like it.
22:41.05chfnNivex, whats the language of source code of asterisk ?
22:41.22cypromisc
22:41.24dalaberaelriah, I think so since * it's cpu eater....
22:41.28AgiNamuC#
22:41.31Pinholechfn, English
22:41.31cypromislol
22:41.46chfnPinhole, lol
22:41.49AgiNamuit was gonna be x86
22:41.50KyrinHow easily can several different numbers be routed through and managed (transfers, holds, conference calls, etc) one asterisk system?  Or is this something that I have to worry about coordinating with my VoIP provider, someone like Vonage?
22:41.53AgiNamubut that wasn't as portable.
22:41.59AgiNamuSee, C is just a hack for those who can't think in machine code.
22:42.04Pinholeactually, it might be Canadian
22:42.14AgiNamujust as C++ is a hack for people who can't think in objects for themselves.
22:42.29chfni always help gnu software with bugs, correction... I will use asterisk and i will report if I find! :)
22:42.31AgiNamuthis whole "high level", "productivity" stuff is a load of crap.
22:42.37*** join/#asterisk fugitivo (~ajf@201.255.105.150)
22:43.32AgiNamuKyrin, it can be easy or hard
22:43.37AgiNamudepending on what you are actually doing.
22:44.05AgiNamuI dont see any way for asterisk to natively scale.
22:44.21AgiNamuif you only have a couple hundred clients, itll be fine
22:44.31AgiNamuapart from that, then you gotta start getting creative.
22:44.32eKo1* was made in C because it was first made to work on Linux and C is the most supported language on that platform.
22:45.04AgiNamui know. im making sarcastic jokes based off of a certain checkin comment i read
22:45.14eKo1AgiNamu: Define 'natively scale'.
22:45.18AgiNamuthat said "C++ is just a crutch for people who can't think in objects for themselves"
22:45.25AgiNamuI mean, there's no built-in clustering system in asterisk
22:45.29Pinholewell, if Linux was written correctly, (asm), we wouldn't have problems being forced to use C. ;)
22:45.31Darwin35?
22:45.37chfnIAX is for comunicating with another IAX right ?
22:45.45Darwin35someone  call me
22:45.49AgiNamuchgn, IAX is a protocl, like SIP, H323, etc.
22:45.53*** join/#asterisk E818 (anonymous@rrcs-24-199-5-190.west.biz.rr.com)
22:45.54AgiNamuIAX is just a good protocol thats easy to use
22:45.57eKo1AgiNamu: Give it time.
22:45.57AgiNamuunlike SIP, H323, etc.
22:46.09AgiNamueKo1, sure, in time. meanwhile, today, there's no cluster capability
22:46.19AgiNamuBUT
22:46.22eKo1Pinhole: Parts of Linux are in ASM.
22:46.24AgiNamuit's not that hard to add your own in
22:46.34captrbcomputers are just a crutch for people who can't manipulate electrons properly
22:46.37AgiNamulol
22:47.04AgiNamuim writing my backend in C# and going to have asterisk use SOAP calls to figure out how ot route calls
22:47.11Pinholeelectrons are just a crutch for people that can't manipulate strings properly.
22:47.23captrbha
22:47.26AgiNamustrings?
22:47.35tzangerAgiNamu: google for string theory
22:47.49AgiNamuI got as far as a bit of quantum mechanics in high school chemistry
22:47.51AgiNamuAP
22:47.55AgiNamuand then i dropped out
22:48.39eKo1AgiNamu: Why SOAP? That seems like overkill and will not scale well in my opinion.
22:48.48tzangerSOAP scales well
22:48.50KyrinAgiNamu: I need like ten different numbers starting with 220 to go through the same system that is also handling two 990 numbers
22:48.56AgiNamubecause SOAP is an easy way for me to do cross-platform communications
22:49.02AgiNamuit's fast, development wise
22:49.07AgiNamuand the performance kicks ass, in my tests
22:49.07chfnwhats the cheapest hardware for 4-port analog + voice + fax ? i mean whats the price of a VFX/41JCT-LS ?
22:49.13chfndoes someone knows an e-store to buy a VFX/41JCT-LS intel card ?
22:49.20*** join/#asterisk r0d3nt (nobody@wsip-24-234-241-84.lv.lv.cox.net)
22:49.22tzangerchfn: why don' tyou contact your distributor and find out
22:49.25AgiNamuwww.atrevido.net : I ran a test using gSOAP on linux, calling ASP.NET on Windows. ASP.NET service queries a DB and returns an answer.
22:49.26E818what's the best way to implement a wake-up call? AGI?
22:49.30eKo1AgiNamu: Only if you have a good environment to work in.
22:49.37AgiNamutesting against my DESKTOP, I could do 470 requests/sec
22:49.40tzangerE818: no.  callfiles
22:49.40*** join/#asterisk jeffik (~jeffik@CPE00c049565af7-CM0012256ead9e.cpe.net.cable.rogers.com)
22:49.45AgiNamuat which point it became CPU bound.
22:49.52AgiNamua single P4
22:49.55chfntzanger, because I live in a FUCKING country called brazil! i have a friend in NY that could buy for me but i need prices
22:50.09tzangerchfn: so ask him
22:50.11AgiNamueKo1, Visual Studio 2005
22:50.14Kyrinheh
22:50.15AgiNamuthat's a kick ass environment to work in
22:50.17tzangermind you I don't think I'd mind living in a fucking country
22:50.20AgiNamuthats where I do all my devs
22:50.25AgiNamudev. even for asterisk.
22:50.31chfntzanger, :( if I call intel from brasil they will say: ARE YOU nutz? whats VFX/41JCT-LS?
22:50.33AgiNamujust plug in GCC into VS2005
22:50.35eKo1Hmm...I use vim and gcc.
22:50.41eKo1That's it.
22:50.45chfntzanger, stupid people in call center
22:50.47AgiNamuchfn, i know how it feels.
22:50.53elriahnotepad.exe
22:50.55AgiNamui live in guatemala
22:51.00tzangervim
22:51.05elriahheh
22:51.08AgiNamuelriah, notepad doesnt handle large files well
22:51.13eKo1So how much did you pay for VS2K5?
22:51.21AgiNamunothing...
22:51.28eKo1warez?
22:51.29elriahOh, I was kidding.  I use vim on *nix, textpad on windows.
22:51.30Kyrinchfn: http://shopper.cnet.com/Dialogic_VFX_41JCT_LS_voice_fax_board/4014-3004_9-30387014.html?
22:51.31AgiNamuI was an MS MVP (didnt get reawarded this year)
22:51.40AgiNamuand they gave me an MSDN universal suib.
22:51.46AgiNamuat any rate, the express version of VS2005 is gonna be $49
22:51.51AgiNamuunless you are a student. then it's $5.
22:52.04chfnAgiNamu, :|
22:52.18eKo1I remember paying about $300 for VS6.
22:52.20AgiNamuand MS SQL 2005 Express is free.
22:52.25eKo1What a rip off.
22:52.35chfnKyrin, thanks.
22:52.43AgiNamueKo1, yea, Professional is like $1000, and Team versions are $2500, and the ripoff is Team Suite, which is $11K
22:52.54eKo1But back then, I) was a M$ junkie.
22:52.54Kyrinchfn: np
22:52.55AgiNamuBUT, Visual Studio team system is EXTREMELY powerful. and it's cheap, compared to Rational, etc.
22:53.04chfnwhats the cheapest pci card for 4 ports ?
22:53.17AgiNamuchfn analog?
22:53.19Pinhole$11k is cheaper than a programmer for a year.  If it save enough time, it is justified.
22:53.21chfnAgiNamu, yes
22:53.24elriahIf you want to plug in gcc, just get C#.NET which comes with visual studio ide for $89.00.
22:53.26AgiNamupinhole, damn straight
22:53.33AgiNamuelriah, but C# doesnt allow gcc
22:53.38AgiNamuyou need VC++
22:53.43elriahAhh..
22:53.46eKo1Just use mono.
22:53.47elriahk, sc
22:53.54AgiNamueKo1, mono isn't an IDE :P
22:54.05chfnAgiNamu, analog.
22:54.10eKo1I was talking about c#.
22:54.10AgiNamuTDM400
22:54.21AgiNamuit's only $350 or so for 4 FXO
22:54.31eKo1Analog sucks. Go digital.
22:54.34Kyrindon't suppose any VoIP providers have irc...?
22:54.48chfnAgiNamu, cnet i could not find prices for TDM400
22:54.54AgiNamuwww.digium.com
22:55.08zilascall parking sucks!
22:55.17chfnAgiNamu, thanks!
22:55.29eKo1Say AgiNamu, do they have ISDN in Guatemala?
22:55.51chfnAgiNamu, hmm asterisk and digium! fine!! :)
22:56.05eKo1Or are the people still stuck with pulse tone phones.
22:56.23AgiNamueko, yea
22:56.29eKo1How much?
22:56.34AgiNamuI can get an E1 too
22:56.36eKo1For a BRI and a PRI.
22:56.37AgiNamufor like $400 a month
22:56.48chfnAgiNamu, love you! :) i will buy soon. probably in the next semester i will buy some.
22:56.49eKo1How much for an E1?
22:56.52AgiNamugood luck
22:56.55AgiNamulike $400 a month
22:56.59AgiNamuup to $900
22:57.00AgiNamudepending
22:57.08AgiNamuincluding like 50K local minutes (@ 2cents US a minute)
22:57.19AgiNamui almost bought one to do bypass
22:57.20eKo1Say I wanted PRI over E1?
22:57.25AgiNamuright
22:57.29AgiNamua few hundred bucks a month
22:57.31chfnWhat type of computer should i have to use more VoIP and less analog lines? but i think i probably will use 10 VoIP lines.
22:57.41AgiNamuchfn, anything really
22:57.43eKo1Interesting.
22:57.50AgiNamueven a 1.5GHz machine can handle 10 lines.
22:58.03eKo1I figured it would be mad expensive in that underdeveloped banana country.
22:58.11AgiNamulol, yea, well, PRI isn't that bad.
22:58.14AgiNamuGetting internet access is
22:58.23AgiNamu2MBps line was like $1000 a month
22:58.26chfnAgiNamu, fuck :( i need to travel to visit a company of a friend that is impl. VoIP in theier ISP.
22:58.28AgiNamu512K ADSL is $230
22:58.37AgiNamuchfn, its not hard
22:58.39chfnAgiNamu, i dont know. Can i handle how much calls with a VoIP line
22:58.42AgiNamuchfn, i didnt know shit a few months ago
22:58.52AgiNamuWith a VoIP line? i dont understand.
22:59.16chfnAgiNamu, with a VoIP phone. Can I handle a call and anwser another. How much can i do this ?
22:59.27AgiNamuthat depends on the phone
22:59.27chfnAgiNamu, it depends of the phone?
22:59.30AgiNamuyep
22:59.36AgiNamusome phones can handle multiple voice channels
22:59.37AgiNamusome cant.
22:59.42chfnAgiNamu, i already saw cisco and tested cisco VoIP phones
22:59.44AgiNamusome can conference 2 calls together.
22:59.49AgiNamuCisco phones are very nice from what I hear.
22:59.54AgiNamuI only use PA168, cause I want IAX2 support.
22:59.59chfnAgiNamu, what country do you live ?
23:00.05AgiNamuguatemala
23:00.09chfnAgiNamu, brasil :D
23:00.26chfnAgiNamu, these FSF projects is getting a lot of users from these kind of countries.
23:00.35AgiNamubrasil.... didnt the presidente just mandate that ALL companies that get government funding must use free software?
23:00.45AgiNamuSounds like a realllly dumb thing to say.... I Can't wait for the integration nightmares.
23:00.59eKo1chfn: 'thse kind of countries'¿
23:01.00AgiNamuchfn, yea, they are getting a lot of support, but for the wrong reasons.
23:01.09chfneKo1, i mean LDEC
23:01.16chfneKo1, i know ?
23:01.32AgiNamuhere, people sya "OH! OpenOffice is free! screw MS"
23:01.34captrbdoes anybody have any tips on troubleshooting a pri line?
23:01.40chfnAgiNamu, for wrong reasons but... i like FSF and its getting a lot of users... so its good
23:01.49AgiNamuand then they realise that oh, OpenOffice ain't even NEAR the quality of Office.
23:01.50chfnAgiNamu, i know your thought
23:02.03AgiNamuthey just dont think past the up front licensing fee.
23:02.05captrbmy telco says that they can't detect signal from digium/asterisk
23:02.11eKo1Yeah, but most users don't even use half the features of Office so...
23:02.19AgiNamueKo1, that's a faulty argument
23:02.29AgiNamumost users might not use many features. But the features they DO use is different for each user.
23:02.30chfnAgiNamu, do you have msn? icq? i would like to test voip with you. lol! :D to talk
23:02.41captrbso they can't loop up the line
23:02.50AgiNamuand the quality of the feature is huge. For instance, CJK support
23:03.05AgiNamuOn XP, Word 2003... CJK support kicks fucking ass. On Linux, Gnome.... OUCH
23:03.09AgiNamueven the IME is a piece of shit
23:03.27AgiNamubut in Office... it detects if i forget to change modes (say, Hangul > Roman) and changes for me.
23:03.35chfnAgiNamu, do you have msn? icq? i would like to test voip with you. lol! :D to talk
23:03.42AgiNamuthat's a SMALL FEATURE.... but they have tons of small features like that.
23:03.47AgiNamuchgfn, mgg@atrevido.net
23:03.50AgiNamumsn
23:04.00chfnAgiNamu, some kind of topic is not for this channel. :| be carefull the 'masters' of this channel probably does not like
23:04.10chfnAgiNamu, thanks. i will add now
23:04.21AgiNamuwell, the only time I've been kicked is when I discussed cracking certain stuff.
23:04.38AgiNamuFSF doesn't extend to reverse engineering :)
23:06.39eKo1I need Visio on Linux.
23:06.46fugitivoeKo1: use kivio
23:07.08KyrinAnyone here use VoIP in Canada and can recommend a provider to use with Asterisk?
23:07.08eKo1Kivio and Dia don't come close to Visio.
23:07.28fugitivowhat else do you need to make a diagram?
23:07.38AgiNamuKyrin: If you want someone in canada, ask Sivana
23:08.19Kyrinsivana: Pls /msg me whtn you get time, I would like a recommendation as to which VoIP provider I should look at
23:08.41eKo1fugitivo: Eh, ERD diagrams for example.
23:08.50KyrinAgiNamu: Thx, I was looking at Link2VoIP 'cause it supports IAX, but I dunno any others
23:09.01shido6use xetricom
23:09.06AgiNamuthere's a lot. goto the wiki
23:09.13shido6xetricom.net's page sucks but they offer voip in canada
23:09.17AgiNamuI do know that sivana has a local PRI in ontario
23:09.21shido6416
23:09.23shido6647
23:09.26mishehubah.
23:09.27shido6and 8xx
23:09.47shido6where do you need the PRI?
23:09.49*** join/#asterisk IronHelix (~irc@ool-182c3fe9.dyn.optonline.net)
23:10.22*** join/#asterisk harryvv (~leonardo@S010600055d210201.vs.shawcable.net)
23:11.19DrelCan anyone recommend a firewall/router with QoS that works well with Asterisk/VoIP in general?  It would also need to support PPTP/VPN passthrough.
23:11.25AgiNamuiptables
23:11.33fugitivopf
23:11.48DrelI'm currently using netgear fvs-318 but it lacks QoS.
23:11.50hardwirewhats the diff inbetween PRI And DSS?
23:11.53dmccollumIPCOP
23:11.58hardwireyou lie!
23:12.35lokocisco 831
23:12.48*** join/#asterisk bjohnson (~bjohnson@ip202-181.tor.istop.com)
23:13.33DrelAgiNamu: Are you suggesting iptables + tc?
23:13.38AgiNamutc?
23:14.24blitzragetc is for QoS I believe.
23:14.30RaYmAn-Bxtraffic control or similar. From linux iproute2
23:14.32Dreltc/lartc: http://lartc.org/
23:14.41blitzrageahhhh yes. Traffic Control.
23:14.50DrelAs far as I know, iptables has no support for QoS.
23:14.56blitzrageiptables rocks, but the FreeBSD filter system looks nicer
23:15.04RaYmAn-BxDrel: you can approximate it though :P
23:15.31blitzrageDrel: it doesn't, you're right (afaik as well). But there are tricks you can do to kind of simulate it. tc is much nicer though.
23:15.34RaYmAn-Bxi.e. you can limit certain connections by dropping packets when they exceed a certain speed
23:15.56DrelWell, I need a firewall/router with QoS that works well with Asterisk, supports PPTP/VPN passthrough, and doesn't require much maintenance.  IE, no patching of kernel to get QoS, I don't want to have to recompile everytime a security update comes out.
23:17.58RaYmAn-Bx2.6 has good QoS support builtin..but obviously it has to be enabled in the kernel.
23:18.13DrelFVS-338 looks like it might not be a bad solution, does anyone have any experience with it?  The FVS-318 I'm currently using seems to have bugs.  It stops forwarding traffic every month or two and requires a restart.  So, I'm a little hesitant to go with Netgear again, but... I like the idea of low-maintenance hardware where I might have to update the firmware once in a while.
23:18.22DrelRaYmAn-Bx: Do any distributions ship with that enabled?
23:18.38RaYmAn-BxI always compile my own kernels so i don't know
23:19.07tzangerDrel: just get a sokeris or WARP platform and throw on a CF card.  same lack of maintenance
23:19.32Dreltzanger: I'm using a sokeris platform for wireless here right now, actually,... Hmm.
23:19.35hardwireWRAP
23:20.27tzangerer yes WRAP I always screw that up
23:20.27hardwireheh
23:20.27hardwireI use WRAP + *
23:20.27hardwire* two tri-mode cards
23:20.27hardwireerr
23:20.27hardwire+
23:20.27tzangerwhat's the geode like for codec conversions
23:20.27hardwireI always screw that up :)
23:20.27hardwiretzanger: I don't even bother.
23:20.28hardwireGSM all the way.
23:20.28Drelhardwire: Got a link for that?
23:20.33tzangerhardwire: :-)
23:20.41tzangerI would love to get a miniPCI ADSL modem for those things
23:20.44Drelhardwire: WRAP that is.
23:20.53hardwiremini-box.com
23:21.06hardwireI talk with the pcengines guy that makes them all the time
23:21.20tzangerhardwire: nice.  get a MiniPCI ADSL modem then
23:21.22hardwireI am so happy there is a quick-n-dirty way of controlling the three leds on the front of it now
23:21.28tzangeryup
23:21.30hardwiretzafrir: that would be nice
23:21.36hardwireI have never seen a mini-pci ADSL card
23:21.49tzangerI use the Sangoma S518 ADSL cards...  I can flood my uplink and maintain call quality
23:21.57hardwireI think the FCC won't like it either
23:22.08tzangerhardwire: why not?
23:22.12hardwirehmm
23:22.15hardwirenm
23:22.18hardwireSangoma eh
23:22.20tzangerthere are tiny tiny modems (my IBM T30 has one)
23:22.21hardwireworks well w/ linux?
23:22.27tzangerhardwire: works *excellent* with linux
23:22.50tzangerit's just a Globespan chipset OEM'd by Sangoma
23:22.50tzangerbut their drivers
23:22.50tzangerit just works
23:24.48Beirdogah
23:24.53Beirdoglobespan
23:24.59tzangerthe funny part is
23:25.20tzangertheir T1 card and their ADSL card don't coexist happily.  Digium's T100P and the S518 work just fine
23:25.47TechDawgAnyone have any experience with the Intel IA92 cards for FXOs?
23:26.57*** join/#asterisk RoyK (~roy@143.80-202-166.nextgentel.com)
23:28.15TechDawgGuess that would be no.
23:29.04Supaplexmmm sangoma gimmie
23:29.20*** part/#asterisk moy (~kvirc@201.137.229.111)
23:29.46hardwirehmm
23:29.54hardwireanybody doing DATA + Voice over PRI?
23:30.05hardwirethats just syncppp on one span right?
23:31.23*** join/#asterisk implicit (~implicit@lgb-cust-66.18.140.106.mpowercom.net)
23:31.44captrbhardwire: trying, but not succeeding
23:31.53hardwirecaptrb: heh!
23:32.12hardwireI know some companies offer dynamic voice line allocation for PRI lines
23:32.21captrbactually, data works, but the telco can't loop up the pri
23:32.21hardwirejust wondering how that can be done w/ two t100p
23:32.31captrbdynamic?
23:32.37hardwireyeh
23:32.50hardwireICG in Colorado offers a data/voice combo and their own router
23:32.51captrbi saw an example of dynamic pri somewhere...
23:33.06hardwirekinda neat
23:33.08RoyK~seen coppice
23:33.11jbotcoppice <~chatzilla@227.166.17.210.dyn.pacific.net.hk> was last seen on IRC in channel #asterisk, 7d 14h 25m 29s ago, saying: 'hanoi is the place for the most delicious food in the world. Gwei Lin is probably the place for the hottest'.
23:33.38captrbwe have fixed channels, but basically that.  from eschelon (craptastic service)
23:33.43hardwireis that all over the D channel how that works?
23:33.57hardwireand how does the PPP session grow
23:34.01hardwireunless its multilink ppp
23:34.03hardwireone per channel
23:34.09zilasplease, can somebody help me with a simple thing for call parking
23:34.19hardwirezilas: I am lost w/ call parking
23:35.53zilasI cant make it to work, I am giving up already... can that a phone doesn't support it?
23:37.10tzangerzilas: what phone
23:39.45*** join/#asterisk tzafrir_laptop (~tzafrir@bzq-179-40-134.cust.bezeqint.net) [NETSPLIT VICTIM]
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23:41.12*** join/#asterisk CoolAcid (~jk@216.99.98.39)
23:41.58Supaplextzanger: you know. that one phone. ;)
23:42.03tzangerheh
23:42.34*** join/#asterisk jason357 (~m00@67.159.26.120)
23:43.57*** part/#asterisk DougNaka (~Doug@207.225.223.187)
23:44.01*** join/#asterisk DougNaka (~Doug@207.225.223.187)
23:45.01zilastzanger: Siemens HiNet LP 5200
23:45.11tzangerregular analog phone or what
23:45.16zilassorry 5200
23:45.20zilas5100
23:45.23*** join/#asterisk shmaltz (~chatzilla@69.28.255.210)
23:45.25zilassip
23:46.17*** join/#asterisk brycec (~brycec@dsl093-157-131.phx1.dsl.speakeasy.net)
23:46.34shmaltzhas anybody tried applying this patch?
23:46.35shmaltzhttp://bugs.digium.com/bug_view_page.php?bug_id=0002905
23:46.59jason357I want a system where people can dial my 800 number, enter an extension and hear a recording with tracking based on extension dialed, most entension will point to a single announcment message
23:46.59shmaltzit fails on 1.0.4, 1.0.5, 1.0.6, 1.0.7
23:47.22jason357my question is, without running my own * server for this, who provides such a thing for a pre minute or monthly fee?
23:47.22tzangershmaltz: probably because it's for HEAD
23:47.42shmaltztzanger, but I want it on stable, no way to that?
23:49.36tzangershmaltz: sure, figur eout what it does and rework it :-)
23:49.46shmaltz;0
23:49.54shmaltz;(
23:50.56brycecbrc_, you alive?
23:50.56jason357is there a directory of asterisk users selling services on their host?
23:51.03brc_yessir
23:52.14brc_brycec, ?
23:54.45reallost1Anyone here use BackgroundDetect?
23:56.20*** part/#asterisk Zipper_32 (~none@s207-6-25-182.bc.hsia.telus.net)
23:57.26zilasI don't believe I was so stupid, wow
23:57.36zilasfigured out...

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